AudioFlinger.cpp revision 25f4395b932fa9859a6e91ba77c5d20d009da64a
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 if (thread == NULL) { 311 LOGE("unknown output thread"); 312 lStatus = BAD_VALUE; 313 goto Exit; 314 } 315 316 wclient = mClients.valueFor(pid); 317 318 if (wclient != NULL) { 319 client = wclient.promote(); 320 } else { 321 client = new Client(this, pid); 322 mClients.add(pid, client); 323 } 324 325 LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 326 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 327 // prevent same audio session on different output threads 328 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 329 if (mPlaybackThreads.keyAt(i) != output && 330 mPlaybackThreads.valueAt(i)->hasAudioSession(*sessionId)) { 331 lStatus = BAD_VALUE; 332 goto Exit; 333 } 334 } 335 lSessionId = *sessionId; 336 } else { 337 // if no audio session id is provided, create one here 338 lSessionId = nextUniqueId(); 339 if (sessionId != NULL) { 340 *sessionId = lSessionId; 341 } 342 } 343 LOGV("createTrack() lSessionId: %d", lSessionId); 344 345 track = thread->createTrack_l(client, streamType, sampleRate, format, 346 channelCount, frameCount, sharedBuffer, lSessionId, &lStatus); 347 } 348 if (lStatus == NO_ERROR) { 349 trackHandle = new TrackHandle(track); 350 } else { 351 // remove local strong reference to Client before deleting the Track so that the Client 352 // destructor is called by the TrackBase destructor with mLock held 353 client.clear(); 354 track.clear(); 355 } 356 357Exit: 358 if(status) { 359 *status = lStatus; 360 } 361 return trackHandle; 362} 363 364uint32_t AudioFlinger::sampleRate(int output) const 365{ 366 Mutex::Autolock _l(mLock); 367 PlaybackThread *thread = checkPlaybackThread_l(output); 368 if (thread == NULL) { 369 LOGW("sampleRate() unknown thread %d", output); 370 return 0; 371 } 372 return thread->sampleRate(); 373} 374 375int AudioFlinger::channelCount(int output) const 376{ 377 Mutex::Autolock _l(mLock); 378 PlaybackThread *thread = checkPlaybackThread_l(output); 379 if (thread == NULL) { 380 LOGW("channelCount() unknown thread %d", output); 381 return 0; 382 } 383 return thread->channelCount(); 384} 385 386int AudioFlinger::format(int output) const 387{ 388 Mutex::Autolock _l(mLock); 389 PlaybackThread *thread = checkPlaybackThread_l(output); 390 if (thread == NULL) { 391 LOGW("format() unknown thread %d", output); 392 return 0; 393 } 394 return thread->format(); 395} 396 397size_t AudioFlinger::frameCount(int output) const 398{ 399 Mutex::Autolock _l(mLock); 400 PlaybackThread *thread = checkPlaybackThread_l(output); 401 if (thread == NULL) { 402 LOGW("frameCount() unknown thread %d", output); 403 return 0; 404 } 405 return thread->frameCount(); 406} 407 408uint32_t AudioFlinger::latency(int output) const 409{ 410 Mutex::Autolock _l(mLock); 411 PlaybackThread *thread = checkPlaybackThread_l(output); 412 if (thread == NULL) { 413 LOGW("latency() unknown thread %d", output); 414 return 0; 415 } 416 return thread->latency(); 417} 418 419status_t AudioFlinger::setMasterVolume(float value) 420{ 421 // check calling permissions 422 if (!settingsAllowed()) { 423 return PERMISSION_DENIED; 424 } 425 426 // when hw supports master volume, don't scale in sw mixer 427 AutoMutex lock(mHardwareLock); 428 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 429 if (mAudioHardware->setMasterVolume(value) == NO_ERROR) { 430 value = 1.0f; 431 } 432 mHardwareStatus = AUDIO_HW_IDLE; 433 434 mMasterVolume = value; 435 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 436 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 437 438 return NO_ERROR; 439} 440 441status_t AudioFlinger::setMode(int mode) 442{ 443 status_t ret; 444 445 // check calling permissions 446 if (!settingsAllowed()) { 447 return PERMISSION_DENIED; 448 } 449 if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) { 450 LOGW("Illegal value: setMode(%d)", mode); 451 return BAD_VALUE; 452 } 453 454 { // scope for the lock 455 AutoMutex lock(mHardwareLock); 456 mHardwareStatus = AUDIO_HW_SET_MODE; 457 ret = mAudioHardware->setMode(mode); 458 mHardwareStatus = AUDIO_HW_IDLE; 459 } 460 461 if (NO_ERROR == ret) { 462 Mutex::Autolock _l(mLock); 463 mMode = mode; 464 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 465 mPlaybackThreads.valueAt(i)->setMode(mode); 466#ifdef LVMX 467 LifeVibes::setMode(mode); 468#endif 469 } 470 471 return ret; 472} 473 474status_t AudioFlinger::setMicMute(bool state) 475{ 476 // check calling permissions 477 if (!settingsAllowed()) { 478 return PERMISSION_DENIED; 479 } 480 481 AutoMutex lock(mHardwareLock); 482 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 483 status_t ret = mAudioHardware->setMicMute(state); 484 mHardwareStatus = AUDIO_HW_IDLE; 485 return ret; 486} 487 488bool AudioFlinger::getMicMute() const 489{ 490 bool state = AudioSystem::MODE_INVALID; 491 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 492 mAudioHardware->getMicMute(&state); 493 mHardwareStatus = AUDIO_HW_IDLE; 494 return state; 495} 496 497status_t AudioFlinger::setMasterMute(bool muted) 498{ 499 // check calling permissions 500 if (!settingsAllowed()) { 501 return PERMISSION_DENIED; 502 } 503 504 mMasterMute = muted; 505 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 506 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 507 508 return NO_ERROR; 509} 510 511float AudioFlinger::masterVolume() const 512{ 513 return mMasterVolume; 514} 515 516bool AudioFlinger::masterMute() const 517{ 518 return mMasterMute; 519} 520 521status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 522{ 523 // check calling permissions 524 if (!settingsAllowed()) { 525 return PERMISSION_DENIED; 526 } 527 528 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { 529 return BAD_VALUE; 530 } 531 532 AutoMutex lock(mLock); 533 PlaybackThread *thread = NULL; 534 if (output) { 535 thread = checkPlaybackThread_l(output); 536 if (thread == NULL) { 537 return BAD_VALUE; 538 } 539 } 540 541 mStreamTypes[stream].volume = value; 542 543 if (thread == NULL) { 544 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 545 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 546 } 547 } else { 548 thread->setStreamVolume(stream, value); 549 } 550 551 return NO_ERROR; 552} 553 554status_t AudioFlinger::setStreamMute(int stream, bool muted) 555{ 556 // check calling permissions 557 if (!settingsAllowed()) { 558 return PERMISSION_DENIED; 559 } 560 561 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || 562 uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { 563 return BAD_VALUE; 564 } 565 566 mStreamTypes[stream].mute = muted; 567 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 568 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 569 570 return NO_ERROR; 571} 572 573float AudioFlinger::streamVolume(int stream, int output) const 574{ 575 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { 576 return 0.0f; 577 } 578 579 AutoMutex lock(mLock); 580 float volume; 581 if (output) { 582 PlaybackThread *thread = checkPlaybackThread_l(output); 583 if (thread == NULL) { 584 return 0.0f; 585 } 586 volume = thread->streamVolume(stream); 587 } else { 588 volume = mStreamTypes[stream].volume; 589 } 590 591 return volume; 592} 593 594bool AudioFlinger::streamMute(int stream) const 595{ 596 if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) { 597 return true; 598 } 599 600 return mStreamTypes[stream].mute; 601} 602 603bool AudioFlinger::isStreamActive(int stream) const 604{ 605 Mutex::Autolock _l(mLock); 606 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 607 if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) { 608 return true; 609 } 610 } 611 return false; 612} 613 614status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 615{ 616 status_t result; 617 618 LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 619 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 620 // check calling permissions 621 if (!settingsAllowed()) { 622 return PERMISSION_DENIED; 623 } 624 625#ifdef LVMX 626 AudioParameter param = AudioParameter(keyValuePairs); 627 LifeVibes::setParameters(ioHandle,keyValuePairs); 628 String8 key = String8(AudioParameter::keyRouting); 629 int device; 630 if (NO_ERROR != param.getInt(key, device)) { 631 device = -1; 632 } 633 634 key = String8(LifevibesTag); 635 String8 value; 636 int musicEnabled = -1; 637 if (NO_ERROR == param.get(key, value)) { 638 if (value == LifevibesEnable) { 639 mLifeVibesClientPid = IPCThreadState::self()->getCallingPid(); 640 musicEnabled = 1; 641 } else if (value == LifevibesDisable) { 642 mLifeVibesClientPid = -1; 643 musicEnabled = 0; 644 } 645 } 646#endif 647 648 // ioHandle == 0 means the parameters are global to the audio hardware interface 649 if (ioHandle == 0) { 650 AutoMutex lock(mHardwareLock); 651 mHardwareStatus = AUDIO_SET_PARAMETER; 652 result = mAudioHardware->setParameters(keyValuePairs); 653#ifdef LVMX 654 if (musicEnabled != -1) { 655 LifeVibes::enableMusic((bool) musicEnabled); 656 } 657#endif 658 mHardwareStatus = AUDIO_HW_IDLE; 659 return result; 660 } 661 662 // hold a strong ref on thread in case closeOutput() or closeInput() is called 663 // and the thread is exited once the lock is released 664 sp<ThreadBase> thread; 665 { 666 Mutex::Autolock _l(mLock); 667 thread = checkPlaybackThread_l(ioHandle); 668 if (thread == NULL) { 669 thread = checkRecordThread_l(ioHandle); 670 } 671 } 672 if (thread != NULL) { 673 result = thread->setParameters(keyValuePairs); 674#ifdef LVMX 675 if ((NO_ERROR == result) && (device != -1)) { 676 LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device); 677 } 678#endif 679 return result; 680 } 681 return BAD_VALUE; 682} 683 684String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 685{ 686// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 687// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 688 689 if (ioHandle == 0) { 690 return mAudioHardware->getParameters(keys); 691 } 692 693 Mutex::Autolock _l(mLock); 694 695 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 696 if (playbackThread != NULL) { 697 return playbackThread->getParameters(keys); 698 } 699 RecordThread *recordThread = checkRecordThread_l(ioHandle); 700 if (recordThread != NULL) { 701 return recordThread->getParameters(keys); 702 } 703 return String8(""); 704} 705 706size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 707{ 708 return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); 709} 710 711unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 712{ 713 if (ioHandle == 0) { 714 return 0; 715 } 716 717 Mutex::Autolock _l(mLock); 718 719 RecordThread *recordThread = checkRecordThread_l(ioHandle); 720 if (recordThread != NULL) { 721 return recordThread->getInputFramesLost(); 722 } 723 return 0; 724} 725 726status_t AudioFlinger::setVoiceVolume(float value) 727{ 728 // check calling permissions 729 if (!settingsAllowed()) { 730 return PERMISSION_DENIED; 731 } 732 733 AutoMutex lock(mHardwareLock); 734 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 735 status_t ret = mAudioHardware->setVoiceVolume(value); 736 mHardwareStatus = AUDIO_HW_IDLE; 737 738 return ret; 739} 740 741status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 742{ 743 status_t status; 744 745 Mutex::Autolock _l(mLock); 746 747 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 748 if (playbackThread != NULL) { 749 return playbackThread->getRenderPosition(halFrames, dspFrames); 750 } 751 752 return BAD_VALUE; 753} 754 755void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 756{ 757 758 Mutex::Autolock _l(mLock); 759 760 int pid = IPCThreadState::self()->getCallingPid(); 761 if (mNotificationClients.indexOfKey(pid) < 0) { 762 sp<NotificationClient> notificationClient = new NotificationClient(this, 763 client, 764 pid); 765 LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 766 767 mNotificationClients.add(pid, notificationClient); 768 769 sp<IBinder> binder = client->asBinder(); 770 binder->linkToDeath(notificationClient); 771 772 // the config change is always sent from playback or record threads to avoid deadlock 773 // with AudioSystem::gLock 774 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 775 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 776 } 777 778 for (size_t i = 0; i < mRecordThreads.size(); i++) { 779 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 780 } 781 } 782} 783 784void AudioFlinger::removeNotificationClient(pid_t pid) 785{ 786 Mutex::Autolock _l(mLock); 787 788 int index = mNotificationClients.indexOfKey(pid); 789 if (index >= 0) { 790 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 791 LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 792#ifdef LVMX 793 if (pid == mLifeVibesClientPid) { 794 LOGV("Disabling lifevibes"); 795 LifeVibes::enableMusic(false); 796 mLifeVibesClientPid = -1; 797 } 798#endif 799 mNotificationClients.removeItem(pid); 800 } 801} 802 803// audioConfigChanged_l() must be called with AudioFlinger::mLock held 804void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 805{ 806 size_t size = mNotificationClients.size(); 807 for (size_t i = 0; i < size; i++) { 808 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 809 } 810} 811 812// removeClient_l() must be called with AudioFlinger::mLock held 813void AudioFlinger::removeClient_l(pid_t pid) 814{ 815 LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 816 mClients.removeItem(pid); 817} 818 819 820// ---------------------------------------------------------------------------- 821 822AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id) 823 : Thread(false), 824 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 825 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false) 826{ 827} 828 829AudioFlinger::ThreadBase::~ThreadBase() 830{ 831 mParamCond.broadcast(); 832 mNewParameters.clear(); 833} 834 835void AudioFlinger::ThreadBase::exit() 836{ 837 // keep a strong ref on ourself so that we wont get 838 // destroyed in the middle of requestExitAndWait() 839 sp <ThreadBase> strongMe = this; 840 841 LOGV("ThreadBase::exit"); 842 { 843 AutoMutex lock(&mLock); 844 mExiting = true; 845 requestExit(); 846 mWaitWorkCV.signal(); 847 } 848 requestExitAndWait(); 849} 850 851uint32_t AudioFlinger::ThreadBase::sampleRate() const 852{ 853 return mSampleRate; 854} 855 856int AudioFlinger::ThreadBase::channelCount() const 857{ 858 return (int)mChannelCount; 859} 860 861int AudioFlinger::ThreadBase::format() const 862{ 863 return mFormat; 864} 865 866size_t AudioFlinger::ThreadBase::frameCount() const 867{ 868 return mFrameCount; 869} 870 871status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 872{ 873 status_t status; 874 875 LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 876 Mutex::Autolock _l(mLock); 877 878 mNewParameters.add(keyValuePairs); 879 mWaitWorkCV.signal(); 880 // wait condition with timeout in case the thread loop has exited 881 // before the request could be processed 882 if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { 883 status = mParamStatus; 884 mWaitWorkCV.signal(); 885 } else { 886 status = TIMED_OUT; 887 } 888 return status; 889} 890 891void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 892{ 893 Mutex::Autolock _l(mLock); 894 sendConfigEvent_l(event, param); 895} 896 897// sendConfigEvent_l() must be called with ThreadBase::mLock held 898void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 899{ 900 ConfigEvent *configEvent = new ConfigEvent(); 901 configEvent->mEvent = event; 902 configEvent->mParam = param; 903 mConfigEvents.add(configEvent); 904 LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 905 mWaitWorkCV.signal(); 906} 907 908void AudioFlinger::ThreadBase::processConfigEvents() 909{ 910 mLock.lock(); 911 while(!mConfigEvents.isEmpty()) { 912 LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 913 ConfigEvent *configEvent = mConfigEvents[0]; 914 mConfigEvents.removeAt(0); 915 // release mLock before locking AudioFlinger mLock: lock order is always 916 // AudioFlinger then ThreadBase to avoid cross deadlock 917 mLock.unlock(); 918 mAudioFlinger->mLock.lock(); 919 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 920 mAudioFlinger->mLock.unlock(); 921 delete configEvent; 922 mLock.lock(); 923 } 924 mLock.unlock(); 925} 926 927status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 928{ 929 const size_t SIZE = 256; 930 char buffer[SIZE]; 931 String8 result; 932 933 bool locked = tryLock(mLock); 934 if (!locked) { 935 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 936 write(fd, buffer, strlen(buffer)); 937 } 938 939 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 940 result.append(buffer); 941 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 942 result.append(buffer); 943 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 944 result.append(buffer); 945 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 946 result.append(buffer); 947 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 948 result.append(buffer); 949 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 950 result.append(buffer); 951 952 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 953 result.append(buffer); 954 result.append(" Index Command"); 955 for (size_t i = 0; i < mNewParameters.size(); ++i) { 956 snprintf(buffer, SIZE, "\n %02d ", i); 957 result.append(buffer); 958 result.append(mNewParameters[i]); 959 } 960 961 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 962 result.append(buffer); 963 snprintf(buffer, SIZE, " Index event param\n"); 964 result.append(buffer); 965 for (size_t i = 0; i < mConfigEvents.size(); i++) { 966 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 967 result.append(buffer); 968 } 969 result.append("\n"); 970 971 write(fd, result.string(), result.size()); 972 973 if (locked) { 974 mLock.unlock(); 975 } 976 return NO_ERROR; 977} 978 979 980// ---------------------------------------------------------------------------- 981 982AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 983 : ThreadBase(audioFlinger, id), 984 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 985 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 986 mDevice(device) 987{ 988 readOutputParameters(); 989 990 mMasterVolume = mAudioFlinger->masterVolume(); 991 mMasterMute = mAudioFlinger->masterMute(); 992 993 for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { 994 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 995 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 996 } 997} 998 999AudioFlinger::PlaybackThread::~PlaybackThread() 1000{ 1001 delete [] mMixBuffer; 1002} 1003 1004status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1005{ 1006 dumpInternals(fd, args); 1007 dumpTracks(fd, args); 1008 dumpEffectChains(fd, args); 1009 return NO_ERROR; 1010} 1011 1012status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1013{ 1014 const size_t SIZE = 256; 1015 char buffer[SIZE]; 1016 String8 result; 1017 1018 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1019 result.append(buffer); 1020 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1021 for (size_t i = 0; i < mTracks.size(); ++i) { 1022 sp<Track> track = mTracks[i]; 1023 if (track != 0) { 1024 track->dump(buffer, SIZE); 1025 result.append(buffer); 1026 } 1027 } 1028 1029 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1030 result.append(buffer); 1031 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1032 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1033 wp<Track> wTrack = mActiveTracks[i]; 1034 if (wTrack != 0) { 1035 sp<Track> track = wTrack.promote(); 1036 if (track != 0) { 1037 track->dump(buffer, SIZE); 1038 result.append(buffer); 1039 } 1040 } 1041 } 1042 write(fd, result.string(), result.size()); 1043 return NO_ERROR; 1044} 1045 1046status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args) 1047{ 1048 const size_t SIZE = 256; 1049 char buffer[SIZE]; 1050 String8 result; 1051 1052 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1053 write(fd, buffer, strlen(buffer)); 1054 1055 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1056 sp<EffectChain> chain = mEffectChains[i]; 1057 if (chain != 0) { 1058 chain->dump(fd, args); 1059 } 1060 } 1061 return NO_ERROR; 1062} 1063 1064status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1065{ 1066 const size_t SIZE = 256; 1067 char buffer[SIZE]; 1068 String8 result; 1069 1070 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1071 result.append(buffer); 1072 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1073 result.append(buffer); 1074 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1075 result.append(buffer); 1076 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1077 result.append(buffer); 1078 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1079 result.append(buffer); 1080 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1081 result.append(buffer); 1082 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1083 result.append(buffer); 1084 write(fd, result.string(), result.size()); 1085 1086 dumpBase(fd, args); 1087 1088 return NO_ERROR; 1089} 1090 1091// Thread virtuals 1092status_t AudioFlinger::PlaybackThread::readyToRun() 1093{ 1094 if (mSampleRate == 0) { 1095 LOGE("No working audio driver found."); 1096 return NO_INIT; 1097 } 1098 LOGI("AudioFlinger's thread %p ready to run", this); 1099 return NO_ERROR; 1100} 1101 1102void AudioFlinger::PlaybackThread::onFirstRef() 1103{ 1104 const size_t SIZE = 256; 1105 char buffer[SIZE]; 1106 1107 snprintf(buffer, SIZE, "Playback Thread %p", this); 1108 1109 run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); 1110} 1111 1112// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1113sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1114 const sp<AudioFlinger::Client>& client, 1115 int streamType, 1116 uint32_t sampleRate, 1117 int format, 1118 int channelCount, 1119 int frameCount, 1120 const sp<IMemory>& sharedBuffer, 1121 int sessionId, 1122 status_t *status) 1123{ 1124 sp<Track> track; 1125 status_t lStatus; 1126 1127 if (mType == DIRECT) { 1128 if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) { 1129 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p", 1130 sampleRate, format, channelCount, mOutput); 1131 lStatus = BAD_VALUE; 1132 goto Exit; 1133 } 1134 } else { 1135 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1136 if (sampleRate > mSampleRate*2) { 1137 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1138 lStatus = BAD_VALUE; 1139 goto Exit; 1140 } 1141 } 1142 1143 if (mOutput == 0) { 1144 LOGE("Audio driver not initialized."); 1145 lStatus = NO_INIT; 1146 goto Exit; 1147 } 1148 1149 { // scope for mLock 1150 Mutex::Autolock _l(mLock); 1151 1152 // all tracks in same audio session must share the same routing strategy otherwise 1153 // conflicts will happen when tracks are moved from one output to another by audio policy 1154 // manager 1155 uint32_t strategy = 1156 AudioSystem::getStrategyForStream((AudioSystem::stream_type)streamType); 1157 for (size_t i = 0; i < mTracks.size(); ++i) { 1158 sp<Track> t = mTracks[i]; 1159 if (t != 0) { 1160 if (sessionId == t->sessionId() && 1161 strategy != AudioSystem::getStrategyForStream((AudioSystem::stream_type)t->type())) { 1162 lStatus = BAD_VALUE; 1163 goto Exit; 1164 } 1165 } 1166 } 1167 1168 track = new Track(this, client, streamType, sampleRate, format, 1169 channelCount, frameCount, sharedBuffer, sessionId); 1170 if (track->getCblk() == NULL || track->name() < 0) { 1171 lStatus = NO_MEMORY; 1172 goto Exit; 1173 } 1174 mTracks.add(track); 1175 1176 sp<EffectChain> chain = getEffectChain_l(sessionId); 1177 if (chain != 0) { 1178 LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1179 track->setMainBuffer(chain->inBuffer()); 1180 chain->setStrategy(AudioSystem::getStrategyForStream((AudioSystem::stream_type)track->type())); 1181 } 1182 } 1183 lStatus = NO_ERROR; 1184 1185Exit: 1186 if(status) { 1187 *status = lStatus; 1188 } 1189 return track; 1190} 1191 1192uint32_t AudioFlinger::PlaybackThread::latency() const 1193{ 1194 if (mOutput) { 1195 return mOutput->latency(); 1196 } 1197 else { 1198 return 0; 1199 } 1200} 1201 1202status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1203{ 1204#ifdef LVMX 1205 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1206 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1207 LifeVibes::setMasterVolume(audioOutputType, value); 1208 } 1209#endif 1210 mMasterVolume = value; 1211 return NO_ERROR; 1212} 1213 1214status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1215{ 1216#ifdef LVMX 1217 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1218 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1219 LifeVibes::setMasterMute(audioOutputType, muted); 1220 } 1221#endif 1222 mMasterMute = muted; 1223 return NO_ERROR; 1224} 1225 1226float AudioFlinger::PlaybackThread::masterVolume() const 1227{ 1228 return mMasterVolume; 1229} 1230 1231bool AudioFlinger::PlaybackThread::masterMute() const 1232{ 1233 return mMasterMute; 1234} 1235 1236status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1237{ 1238#ifdef LVMX 1239 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1240 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1241 LifeVibes::setStreamVolume(audioOutputType, stream, value); 1242 } 1243#endif 1244 mStreamTypes[stream].volume = value; 1245 return NO_ERROR; 1246} 1247 1248status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1249{ 1250#ifdef LVMX 1251 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1252 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1253 LifeVibes::setStreamMute(audioOutputType, stream, muted); 1254 } 1255#endif 1256 mStreamTypes[stream].mute = muted; 1257 return NO_ERROR; 1258} 1259 1260float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1261{ 1262 return mStreamTypes[stream].volume; 1263} 1264 1265bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1266{ 1267 return mStreamTypes[stream].mute; 1268} 1269 1270bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const 1271{ 1272 Mutex::Autolock _l(mLock); 1273 size_t count = mActiveTracks.size(); 1274 for (size_t i = 0 ; i < count ; ++i) { 1275 sp<Track> t = mActiveTracks[i].promote(); 1276 if (t == 0) continue; 1277 Track* const track = t.get(); 1278 if (t->type() == stream) 1279 return true; 1280 } 1281 return false; 1282} 1283 1284// addTrack_l() must be called with ThreadBase::mLock held 1285status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1286{ 1287 status_t status = ALREADY_EXISTS; 1288 1289 // set retry count for buffer fill 1290 track->mRetryCount = kMaxTrackStartupRetries; 1291 if (mActiveTracks.indexOf(track) < 0) { 1292 // the track is newly added, make sure it fills up all its 1293 // buffers before playing. This is to ensure the client will 1294 // effectively get the latency it requested. 1295 track->mFillingUpStatus = Track::FS_FILLING; 1296 track->mResetDone = false; 1297 mActiveTracks.add(track); 1298 if (track->mainBuffer() != mMixBuffer) { 1299 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1300 if (chain != 0) { 1301 LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1302 chain->startTrack(); 1303 } 1304 } 1305 1306 status = NO_ERROR; 1307 } 1308 1309 LOGV("mWaitWorkCV.broadcast"); 1310 mWaitWorkCV.broadcast(); 1311 1312 return status; 1313} 1314 1315// destroyTrack_l() must be called with ThreadBase::mLock held 1316void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1317{ 1318 track->mState = TrackBase::TERMINATED; 1319 if (mActiveTracks.indexOf(track) < 0) { 1320 mTracks.remove(track); 1321 deleteTrackName_l(track->name()); 1322 } 1323} 1324 1325String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1326{ 1327 return mOutput->getParameters(keys); 1328} 1329 1330// destroyTrack_l() must be called with AudioFlinger::mLock held 1331void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1332 AudioSystem::OutputDescriptor desc; 1333 void *param2 = 0; 1334 1335 LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1336 1337 switch (event) { 1338 case AudioSystem::OUTPUT_OPENED: 1339 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1340 desc.channels = mChannels; 1341 desc.samplingRate = mSampleRate; 1342 desc.format = mFormat; 1343 desc.frameCount = mFrameCount; 1344 desc.latency = latency(); 1345 param2 = &desc; 1346 break; 1347 1348 case AudioSystem::STREAM_CONFIG_CHANGED: 1349 param2 = ¶m; 1350 case AudioSystem::OUTPUT_CLOSED: 1351 default: 1352 break; 1353 } 1354 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1355} 1356 1357void AudioFlinger::PlaybackThread::readOutputParameters() 1358{ 1359 mSampleRate = mOutput->sampleRate(); 1360 mChannels = mOutput->channels(); 1361 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 1362 mFormat = mOutput->format(); 1363 mFrameSize = (uint16_t)mOutput->frameSize(); 1364 mFrameCount = mOutput->bufferSize() / mFrameSize; 1365 1366 // FIXME - Current mixer implementation only supports stereo output: Always 1367 // Allocate a stereo buffer even if HW output is mono. 1368 if (mMixBuffer != NULL) delete[] mMixBuffer; 1369 mMixBuffer = new int16_t[mFrameCount * 2]; 1370 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1371 1372 // force reconfiguration of effect chains and engines to take new buffer size and audio 1373 // parameters into account 1374 // Note that mLock is not held when readOutputParameters() is called from the constructor 1375 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1376 // matter. 1377 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1378 Vector< sp<EffectChain> > effectChains = mEffectChains; 1379 for (size_t i = 0; i < effectChains.size(); i ++) { 1380 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this); 1381 } 1382} 1383 1384status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1385{ 1386 if (halFrames == 0 || dspFrames == 0) { 1387 return BAD_VALUE; 1388 } 1389 if (mOutput == 0) { 1390 return INVALID_OPERATION; 1391 } 1392 *halFrames = mBytesWritten/mOutput->frameSize(); 1393 1394 return mOutput->getRenderPosition(dspFrames); 1395} 1396 1397bool AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1398{ 1399 Mutex::Autolock _l(mLock); 1400 if (getEffectChain_l(sessionId) != 0) { 1401 return true; 1402 } 1403 1404 for (size_t i = 0; i < mTracks.size(); ++i) { 1405 sp<Track> track = mTracks[i]; 1406 if (sessionId == track->sessionId() && 1407 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1408 return true; 1409 } 1410 } 1411 1412 return false; 1413} 1414 1415uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1416{ 1417 // session AudioSystem::SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1418 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1419 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 1420 return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 1421 } 1422 for (size_t i = 0; i < mTracks.size(); i++) { 1423 sp<Track> track = mTracks[i]; 1424 if (sessionId == track->sessionId() && 1425 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1426 return AudioSystem::getStrategyForStream((AudioSystem::stream_type) track->type()); 1427 } 1428 } 1429 return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 1430} 1431 1432sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId) 1433{ 1434 Mutex::Autolock _l(mLock); 1435 return getEffectChain_l(sessionId); 1436} 1437 1438sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId) 1439{ 1440 sp<EffectChain> chain; 1441 1442 size_t size = mEffectChains.size(); 1443 for (size_t i = 0; i < size; i++) { 1444 if (mEffectChains[i]->sessionId() == sessionId) { 1445 chain = mEffectChains[i]; 1446 break; 1447 } 1448 } 1449 return chain; 1450} 1451 1452void AudioFlinger::PlaybackThread::setMode(uint32_t mode) 1453{ 1454 Mutex::Autolock _l(mLock); 1455 size_t size = mEffectChains.size(); 1456 for (size_t i = 0; i < size; i++) { 1457 mEffectChains[i]->setMode_l(mode); 1458 } 1459} 1460 1461// ---------------------------------------------------------------------------- 1462 1463AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1464 : PlaybackThread(audioFlinger, output, id, device), 1465 mAudioMixer(0) 1466{ 1467 mType = PlaybackThread::MIXER; 1468 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1469 1470 // FIXME - Current mixer implementation only supports stereo output 1471 if (mChannelCount == 1) { 1472 LOGE("Invalid audio hardware channel count"); 1473 } 1474} 1475 1476AudioFlinger::MixerThread::~MixerThread() 1477{ 1478 delete mAudioMixer; 1479} 1480 1481bool AudioFlinger::MixerThread::threadLoop() 1482{ 1483 Vector< sp<Track> > tracksToRemove; 1484 uint32_t mixerStatus = MIXER_IDLE; 1485 nsecs_t standbyTime = systemTime(); 1486 size_t mixBufferSize = mFrameCount * mFrameSize; 1487 // FIXME: Relaxed timing because of a certain device that can't meet latency 1488 // Should be reduced to 2x after the vendor fixes the driver issue 1489 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1490 nsecs_t lastWarning = 0; 1491 bool longStandbyExit = false; 1492 uint32_t activeSleepTime = activeSleepTimeUs(); 1493 uint32_t idleSleepTime = idleSleepTimeUs(); 1494 uint32_t sleepTime = idleSleepTime; 1495 Vector< sp<EffectChain> > effectChains; 1496 1497 while (!exitPending()) 1498 { 1499 processConfigEvents(); 1500 1501 mixerStatus = MIXER_IDLE; 1502 { // scope for mLock 1503 1504 Mutex::Autolock _l(mLock); 1505 1506 if (checkForNewParameters_l()) { 1507 mixBufferSize = mFrameCount * mFrameSize; 1508 // FIXME: Relaxed timing because of a certain device that can't meet latency 1509 // Should be reduced to 2x after the vendor fixes the driver issue 1510 maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1511 activeSleepTime = activeSleepTimeUs(); 1512 idleSleepTime = idleSleepTimeUs(); 1513 } 1514 1515 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1516 1517 // put audio hardware into standby after short delay 1518 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1519 mSuspended) { 1520 if (!mStandby) { 1521 LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1522 mOutput->standby(); 1523 mStandby = true; 1524 mBytesWritten = 0; 1525 } 1526 1527 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1528 // we're about to wait, flush the binder command buffer 1529 IPCThreadState::self()->flushCommands(); 1530 1531 if (exitPending()) break; 1532 1533 // wait until we have something to do... 1534 LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1535 mWaitWorkCV.wait(mLock); 1536 LOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1537 1538 if (mMasterMute == false) { 1539 char value[PROPERTY_VALUE_MAX]; 1540 property_get("ro.audio.silent", value, "0"); 1541 if (atoi(value)) { 1542 LOGD("Silence is golden"); 1543 setMasterMute(true); 1544 } 1545 } 1546 1547 standbyTime = systemTime() + kStandbyTimeInNsecs; 1548 sleepTime = idleSleepTime; 1549 continue; 1550 } 1551 } 1552 1553 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1554 1555 // prevent any changes in effect chain list and in each effect chain 1556 // during mixing and effect process as the audio buffers could be deleted 1557 // or modified if an effect is created or deleted 1558 lockEffectChains_l(effectChains); 1559 } 1560 1561 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1562 // mix buffers... 1563 mAudioMixer->process(); 1564 sleepTime = 0; 1565 standbyTime = systemTime() + kStandbyTimeInNsecs; 1566 //TODO: delay standby when effects have a tail 1567 } else { 1568 // If no tracks are ready, sleep once for the duration of an output 1569 // buffer size, then write 0s to the output 1570 if (sleepTime == 0) { 1571 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1572 sleepTime = activeSleepTime; 1573 } else { 1574 sleepTime = idleSleepTime; 1575 } 1576 } else if (mBytesWritten != 0 || 1577 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 1578 memset (mMixBuffer, 0, mixBufferSize); 1579 sleepTime = 0; 1580 LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 1581 } 1582 // TODO add standby time extension fct of effect tail 1583 } 1584 1585 if (mSuspended) { 1586 sleepTime = idleSleepTime; 1587 } 1588 // sleepTime == 0 means we must write to audio hardware 1589 if (sleepTime == 0) { 1590 for (size_t i = 0; i < effectChains.size(); i ++) { 1591 effectChains[i]->process_l(); 1592 } 1593 // enable changes in effect chain 1594 unlockEffectChains(effectChains); 1595#ifdef LVMX 1596 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1597 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1598 LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize); 1599 } 1600#endif 1601 mLastWriteTime = systemTime(); 1602 mInWrite = true; 1603 mBytesWritten += mixBufferSize; 1604 1605 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 1606 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 1607 mNumWrites++; 1608 mInWrite = false; 1609 nsecs_t now = systemTime(); 1610 nsecs_t delta = now - mLastWriteTime; 1611 if (delta > maxPeriod) { 1612 mNumDelayedWrites++; 1613 if ((now - lastWarning) > kWarningThrottle) { 1614 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 1615 ns2ms(delta), mNumDelayedWrites, this); 1616 lastWarning = now; 1617 } 1618 if (mStandby) { 1619 longStandbyExit = true; 1620 } 1621 } 1622 mStandby = false; 1623 } else { 1624 // enable changes in effect chain 1625 unlockEffectChains(effectChains); 1626 usleep(sleepTime); 1627 } 1628 1629 // finally let go of all our tracks, without the lock held 1630 // since we can't guarantee the destructors won't acquire that 1631 // same lock. 1632 tracksToRemove.clear(); 1633 1634 // Effect chains will be actually deleted here if they were removed from 1635 // mEffectChains list during mixing or effects processing 1636 effectChains.clear(); 1637 } 1638 1639 if (!mStandby) { 1640 mOutput->standby(); 1641 } 1642 1643 LOGV("MixerThread %p exiting", this); 1644 return false; 1645} 1646 1647// prepareTracks_l() must be called with ThreadBase::mLock held 1648uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 1649{ 1650 1651 uint32_t mixerStatus = MIXER_IDLE; 1652 // find out which tracks need to be processed 1653 size_t count = activeTracks.size(); 1654 size_t mixedTracks = 0; 1655 size_t tracksWithEffect = 0; 1656 1657 float masterVolume = mMasterVolume; 1658 bool masterMute = mMasterMute; 1659 1660#ifdef LVMX 1661 bool tracksConnectedChanged = false; 1662 bool stateChanged = false; 1663 1664 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1665 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1666 { 1667 int activeTypes = 0; 1668 for (size_t i=0 ; i<count ; i++) { 1669 sp<Track> t = activeTracks[i].promote(); 1670 if (t == 0) continue; 1671 Track* const track = t.get(); 1672 int iTracktype=track->type(); 1673 activeTypes |= 1<<track->type(); 1674 } 1675 LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute); 1676 } 1677#endif 1678 // Delegate master volume control to effect in output mix effect chain if needed 1679 sp<EffectChain> chain = getEffectChain_l(AudioSystem::SESSION_OUTPUT_MIX); 1680 if (chain != 0) { 1681 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 1682 chain->setVolume_l(&v, &v); 1683 masterVolume = (float)((v + (1 << 23)) >> 24); 1684 chain.clear(); 1685 } 1686 1687 for (size_t i=0 ; i<count ; i++) { 1688 sp<Track> t = activeTracks[i].promote(); 1689 if (t == 0) continue; 1690 1691 Track* const track = t.get(); 1692 audio_track_cblk_t* cblk = track->cblk(); 1693 1694 // The first time a track is added we wait 1695 // for all its buffers to be filled before processing it 1696 mAudioMixer->setActiveTrack(track->name()); 1697 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 1698 !track->isPaused() && !track->isTerminated()) 1699 { 1700 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 1701 1702 mixedTracks++; 1703 1704 // track->mainBuffer() != mMixBuffer means there is an effect chain 1705 // connected to the track 1706 chain.clear(); 1707 if (track->mainBuffer() != mMixBuffer) { 1708 chain = getEffectChain_l(track->sessionId()); 1709 // Delegate volume control to effect in track effect chain if needed 1710 if (chain != 0) { 1711 tracksWithEffect++; 1712 } else { 1713 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 1714 track->name(), track->sessionId()); 1715 } 1716 } 1717 1718 1719 int param = AudioMixer::VOLUME; 1720 if (track->mFillingUpStatus == Track::FS_FILLED) { 1721 // no ramp for the first volume setting 1722 track->mFillingUpStatus = Track::FS_ACTIVE; 1723 if (track->mState == TrackBase::RESUMING) { 1724 track->mState = TrackBase::ACTIVE; 1725 param = AudioMixer::RAMP_VOLUME; 1726 } 1727 } else if (cblk->server != 0) { 1728 // If the track is stopped before the first frame was mixed, 1729 // do not apply ramp 1730 param = AudioMixer::RAMP_VOLUME; 1731 } 1732 1733 // compute volume for this track 1734 int16_t left, right, aux; 1735 if (track->isMuted() || masterMute || track->isPausing() || 1736 mStreamTypes[track->type()].mute) { 1737 left = right = aux = 0; 1738 if (track->isPausing()) { 1739 track->setPaused(); 1740 } 1741 } else { 1742 // read original volumes with volume control 1743 float typeVolume = mStreamTypes[track->type()].volume; 1744#ifdef LVMX 1745 bool streamMute=false; 1746 // read the volume from the LivesVibes audio engine. 1747 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1748 { 1749 LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); 1750 if (streamMute) { 1751 typeVolume = 0; 1752 } 1753 } 1754#endif 1755 float v = masterVolume * typeVolume; 1756 uint32_t vl = (uint32_t)(v * cblk->volume[0]) << 12; 1757 uint32_t vr = (uint32_t)(v * cblk->volume[1]) << 12; 1758 1759 // Delegate volume control to effect in track effect chain if needed 1760 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1761 // Do not ramp volume is volume is controlled by effect 1762 param = AudioMixer::VOLUME; 1763 } 1764 1765 // Convert volumes from 8.24 to 4.12 format 1766 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1767 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1768 left = int16_t(v_clamped); 1769 v_clamped = (vr + (1 << 11)) >> 12; 1770 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1771 right = int16_t(v_clamped); 1772 1773 v_clamped = (uint32_t)(v * cblk->sendLevel); 1774 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1775 aux = int16_t(v_clamped); 1776 } 1777 1778#ifdef LVMX 1779 if ( tracksConnectedChanged || stateChanged ) 1780 { 1781 // only do the ramp when the volume is changed by the user / application 1782 param = AudioMixer::VOLUME; 1783 } 1784#endif 1785 1786 // XXX: these things DON'T need to be done each time 1787 mAudioMixer->setBufferProvider(track); 1788 mAudioMixer->enable(AudioMixer::MIXING); 1789 1790 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1791 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1792 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1793 mAudioMixer->setParameter( 1794 AudioMixer::TRACK, 1795 AudioMixer::FORMAT, (void *)track->format()); 1796 mAudioMixer->setParameter( 1797 AudioMixer::TRACK, 1798 AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); 1799 mAudioMixer->setParameter( 1800 AudioMixer::RESAMPLE, 1801 AudioMixer::SAMPLE_RATE, 1802 (void *)(cblk->sampleRate)); 1803 mAudioMixer->setParameter( 1804 AudioMixer::TRACK, 1805 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1806 mAudioMixer->setParameter( 1807 AudioMixer::TRACK, 1808 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1809 1810 // reset retry count 1811 track->mRetryCount = kMaxTrackRetries; 1812 mixerStatus = MIXER_TRACKS_READY; 1813 } else { 1814 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1815 if (track->isStopped()) { 1816 track->reset(); 1817 } 1818 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1819 // We have consumed all the buffers of this track. 1820 // Remove it from the list of active tracks. 1821 tracksToRemove->add(track); 1822 } else { 1823 // No buffers for this track. Give it a few chances to 1824 // fill a buffer, then remove it from active list. 1825 if (--(track->mRetryCount) <= 0) { 1826 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1827 tracksToRemove->add(track); 1828 } else if (mixerStatus != MIXER_TRACKS_READY) { 1829 mixerStatus = MIXER_TRACKS_ENABLED; 1830 } 1831 } 1832 mAudioMixer->disable(AudioMixer::MIXING); 1833 } 1834 } 1835 1836 // remove all the tracks that need to be... 1837 count = tracksToRemove->size(); 1838 if (UNLIKELY(count)) { 1839 for (size_t i=0 ; i<count ; i++) { 1840 const sp<Track>& track = tracksToRemove->itemAt(i); 1841 mActiveTracks.remove(track); 1842 if (track->mainBuffer() != mMixBuffer) { 1843 chain = getEffectChain_l(track->sessionId()); 1844 if (chain != 0) { 1845 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1846 chain->stopTrack(); 1847 } 1848 } 1849 if (track->isTerminated()) { 1850 mTracks.remove(track); 1851 deleteTrackName_l(track->mName); 1852 } 1853 } 1854 } 1855 1856 // mix buffer must be cleared if all tracks are connected to an 1857 // effect chain as in this case the mixer will not write to 1858 // mix buffer and track effects will accumulate into it 1859 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1860 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1861 } 1862 1863 return mixerStatus; 1864} 1865 1866void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1867{ 1868 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1869 this, streamType, mTracks.size()); 1870 Mutex::Autolock _l(mLock); 1871 1872 size_t size = mTracks.size(); 1873 for (size_t i = 0; i < size; i++) { 1874 sp<Track> t = mTracks[i]; 1875 if (t->type() == streamType) { 1876 t->mCblk->lock.lock(); 1877 t->mCblk->flags |= CBLK_INVALID_ON; 1878 t->mCblk->cv.signal(); 1879 t->mCblk->lock.unlock(); 1880 } 1881 } 1882} 1883 1884 1885// getTrackName_l() must be called with ThreadBase::mLock held 1886int AudioFlinger::MixerThread::getTrackName_l() 1887{ 1888 return mAudioMixer->getTrackName(); 1889} 1890 1891// deleteTrackName_l() must be called with ThreadBase::mLock held 1892void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1893{ 1894 LOGV("remove track (%d) and delete from mixer", name); 1895 mAudioMixer->deleteTrackName(name); 1896} 1897 1898// checkForNewParameters_l() must be called with ThreadBase::mLock held 1899bool AudioFlinger::MixerThread::checkForNewParameters_l() 1900{ 1901 bool reconfig = false; 1902 1903 while (!mNewParameters.isEmpty()) { 1904 status_t status = NO_ERROR; 1905 String8 keyValuePair = mNewParameters[0]; 1906 AudioParameter param = AudioParameter(keyValuePair); 1907 int value; 1908 1909 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1910 reconfig = true; 1911 } 1912 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1913 if (value != AudioSystem::PCM_16_BIT) { 1914 status = BAD_VALUE; 1915 } else { 1916 reconfig = true; 1917 } 1918 } 1919 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1920 if (value != AudioSystem::CHANNEL_OUT_STEREO) { 1921 status = BAD_VALUE; 1922 } else { 1923 reconfig = true; 1924 } 1925 } 1926 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1927 // do not accept frame count changes if tracks are open as the track buffer 1928 // size depends on frame count and correct behavior would not be garantied 1929 // if frame count is changed after track creation 1930 if (!mTracks.isEmpty()) { 1931 status = INVALID_OPERATION; 1932 } else { 1933 reconfig = true; 1934 } 1935 } 1936 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1937 // forward device change to effects that have requested to be 1938 // aware of attached audio device. 1939 mDevice = (uint32_t)value; 1940 for (size_t i = 0; i < mEffectChains.size(); i++) { 1941 mEffectChains[i]->setDevice_l(mDevice); 1942 } 1943 } 1944 1945 if (status == NO_ERROR) { 1946 status = mOutput->setParameters(keyValuePair); 1947 if (!mStandby && status == INVALID_OPERATION) { 1948 mOutput->standby(); 1949 mStandby = true; 1950 mBytesWritten = 0; 1951 status = mOutput->setParameters(keyValuePair); 1952 } 1953 if (status == NO_ERROR && reconfig) { 1954 delete mAudioMixer; 1955 readOutputParameters(); 1956 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1957 for (size_t i = 0; i < mTracks.size() ; i++) { 1958 int name = getTrackName_l(); 1959 if (name < 0) break; 1960 mTracks[i]->mName = name; 1961 // limit track sample rate to 2 x new output sample rate 1962 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 1963 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 1964 } 1965 } 1966 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 1967 } 1968 } 1969 1970 mNewParameters.removeAt(0); 1971 1972 mParamStatus = status; 1973 mParamCond.signal(); 1974 mWaitWorkCV.wait(mLock); 1975 } 1976 return reconfig; 1977} 1978 1979status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 1980{ 1981 const size_t SIZE = 256; 1982 char buffer[SIZE]; 1983 String8 result; 1984 1985 PlaybackThread::dumpInternals(fd, args); 1986 1987 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 1988 result.append(buffer); 1989 write(fd, result.string(), result.size()); 1990 return NO_ERROR; 1991} 1992 1993uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 1994{ 1995 return (uint32_t)(mOutput->latency() * 1000) / 2; 1996} 1997 1998uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 1999{ 2000 return (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000; 2001} 2002 2003// ---------------------------------------------------------------------------- 2004AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2005 : PlaybackThread(audioFlinger, output, id, device) 2006{ 2007 mType = PlaybackThread::DIRECT; 2008} 2009 2010AudioFlinger::DirectOutputThread::~DirectOutputThread() 2011{ 2012} 2013 2014 2015static inline int16_t clamp16(int32_t sample) 2016{ 2017 if ((sample>>15) ^ (sample>>31)) 2018 sample = 0x7FFF ^ (sample>>31); 2019 return sample; 2020} 2021 2022static inline 2023int32_t mul(int16_t in, int16_t v) 2024{ 2025#if defined(__arm__) && !defined(__thumb__) 2026 int32_t out; 2027 asm( "smulbb %[out], %[in], %[v] \n" 2028 : [out]"=r"(out) 2029 : [in]"%r"(in), [v]"r"(v) 2030 : ); 2031 return out; 2032#else 2033 return in * int32_t(v); 2034#endif 2035} 2036 2037void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2038{ 2039 // Do not apply volume on compressed audio 2040 if (!AudioSystem::isLinearPCM(mFormat)) { 2041 return; 2042 } 2043 2044 // convert to signed 16 bit before volume calculation 2045 if (mFormat == AudioSystem::PCM_8_BIT) { 2046 size_t count = mFrameCount * mChannelCount; 2047 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2048 int16_t *dst = mMixBuffer + count-1; 2049 while(count--) { 2050 *dst-- = (int16_t)(*src--^0x80) << 8; 2051 } 2052 } 2053 2054 size_t frameCount = mFrameCount; 2055 int16_t *out = mMixBuffer; 2056 if (ramp) { 2057 if (mChannelCount == 1) { 2058 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2059 int32_t vlInc = d / (int32_t)frameCount; 2060 int32_t vl = ((int32_t)mLeftVolShort << 16); 2061 do { 2062 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2063 out++; 2064 vl += vlInc; 2065 } while (--frameCount); 2066 2067 } else { 2068 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2069 int32_t vlInc = d / (int32_t)frameCount; 2070 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2071 int32_t vrInc = d / (int32_t)frameCount; 2072 int32_t vl = ((int32_t)mLeftVolShort << 16); 2073 int32_t vr = ((int32_t)mRightVolShort << 16); 2074 do { 2075 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2076 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2077 out += 2; 2078 vl += vlInc; 2079 vr += vrInc; 2080 } while (--frameCount); 2081 } 2082 } else { 2083 if (mChannelCount == 1) { 2084 do { 2085 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2086 out++; 2087 } while (--frameCount); 2088 } else { 2089 do { 2090 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2091 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2092 out += 2; 2093 } while (--frameCount); 2094 } 2095 } 2096 2097 // convert back to unsigned 8 bit after volume calculation 2098 if (mFormat == AudioSystem::PCM_8_BIT) { 2099 size_t count = mFrameCount * mChannelCount; 2100 int16_t *src = mMixBuffer; 2101 uint8_t *dst = (uint8_t *)mMixBuffer; 2102 while(count--) { 2103 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2104 } 2105 } 2106 2107 mLeftVolShort = leftVol; 2108 mRightVolShort = rightVol; 2109} 2110 2111bool AudioFlinger::DirectOutputThread::threadLoop() 2112{ 2113 uint32_t mixerStatus = MIXER_IDLE; 2114 sp<Track> trackToRemove; 2115 sp<Track> activeTrack; 2116 nsecs_t standbyTime = systemTime(); 2117 int8_t *curBuf; 2118 size_t mixBufferSize = mFrameCount*mFrameSize; 2119 uint32_t activeSleepTime = activeSleepTimeUs(); 2120 uint32_t idleSleepTime = idleSleepTimeUs(); 2121 uint32_t sleepTime = idleSleepTime; 2122 // use shorter standby delay as on normal output to release 2123 // hardware resources as soon as possible 2124 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2125 2126 while (!exitPending()) 2127 { 2128 bool rampVolume; 2129 uint16_t leftVol; 2130 uint16_t rightVol; 2131 Vector< sp<EffectChain> > effectChains; 2132 2133 processConfigEvents(); 2134 2135 mixerStatus = MIXER_IDLE; 2136 2137 { // scope for the mLock 2138 2139 Mutex::Autolock _l(mLock); 2140 2141 if (checkForNewParameters_l()) { 2142 mixBufferSize = mFrameCount*mFrameSize; 2143 activeSleepTime = activeSleepTimeUs(); 2144 idleSleepTime = idleSleepTimeUs(); 2145 standbyDelay = microseconds(activeSleepTime*2); 2146 } 2147 2148 // put audio hardware into standby after short delay 2149 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2150 mSuspended) { 2151 // wait until we have something to do... 2152 if (!mStandby) { 2153 LOGV("Audio hardware entering standby, mixer %p\n", this); 2154 mOutput->standby(); 2155 mStandby = true; 2156 mBytesWritten = 0; 2157 } 2158 2159 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2160 // we're about to wait, flush the binder command buffer 2161 IPCThreadState::self()->flushCommands(); 2162 2163 if (exitPending()) break; 2164 2165 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2166 mWaitWorkCV.wait(mLock); 2167 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2168 2169 if (mMasterMute == false) { 2170 char value[PROPERTY_VALUE_MAX]; 2171 property_get("ro.audio.silent", value, "0"); 2172 if (atoi(value)) { 2173 LOGD("Silence is golden"); 2174 setMasterMute(true); 2175 } 2176 } 2177 2178 standbyTime = systemTime() + standbyDelay; 2179 sleepTime = idleSleepTime; 2180 continue; 2181 } 2182 } 2183 2184 effectChains = mEffectChains; 2185 2186 // find out which tracks need to be processed 2187 if (mActiveTracks.size() != 0) { 2188 sp<Track> t = mActiveTracks[0].promote(); 2189 if (t == 0) continue; 2190 2191 Track* const track = t.get(); 2192 audio_track_cblk_t* cblk = track->cblk(); 2193 2194 // The first time a track is added we wait 2195 // for all its buffers to be filled before processing it 2196 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 2197 !track->isPaused() && !track->isTerminated()) 2198 { 2199 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2200 2201 if (track->mFillingUpStatus == Track::FS_FILLED) { 2202 track->mFillingUpStatus = Track::FS_ACTIVE; 2203 mLeftVolFloat = mRightVolFloat = 0; 2204 mLeftVolShort = mRightVolShort = 0; 2205 if (track->mState == TrackBase::RESUMING) { 2206 track->mState = TrackBase::ACTIVE; 2207 rampVolume = true; 2208 } 2209 } else if (cblk->server != 0) { 2210 // If the track is stopped before the first frame was mixed, 2211 // do not apply ramp 2212 rampVolume = true; 2213 } 2214 // compute volume for this track 2215 float left, right; 2216 if (track->isMuted() || mMasterMute || track->isPausing() || 2217 mStreamTypes[track->type()].mute) { 2218 left = right = 0; 2219 if (track->isPausing()) { 2220 track->setPaused(); 2221 } 2222 } else { 2223 float typeVolume = mStreamTypes[track->type()].volume; 2224 float v = mMasterVolume * typeVolume; 2225 float v_clamped = v * cblk->volume[0]; 2226 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2227 left = v_clamped/MAX_GAIN; 2228 v_clamped = v * cblk->volume[1]; 2229 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2230 right = v_clamped/MAX_GAIN; 2231 } 2232 2233 if (left != mLeftVolFloat || right != mRightVolFloat) { 2234 mLeftVolFloat = left; 2235 mRightVolFloat = right; 2236 2237 // If audio HAL implements volume control, 2238 // force software volume to nominal value 2239 if (mOutput->setVolume(left, right) == NO_ERROR) { 2240 left = 1.0f; 2241 right = 1.0f; 2242 } 2243 2244 // Convert volumes from float to 8.24 2245 uint32_t vl = (uint32_t)(left * (1 << 24)); 2246 uint32_t vr = (uint32_t)(right * (1 << 24)); 2247 2248 // Delegate volume control to effect in track effect chain if needed 2249 // only one effect chain can be present on DirectOutputThread, so if 2250 // there is one, the track is connected to it 2251 if (!effectChains.isEmpty()) { 2252 // Do not ramp volume is volume is controlled by effect 2253 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2254 rampVolume = false; 2255 } 2256 } 2257 2258 // Convert volumes from 8.24 to 4.12 format 2259 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2260 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2261 leftVol = (uint16_t)v_clamped; 2262 v_clamped = (vr + (1 << 11)) >> 12; 2263 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2264 rightVol = (uint16_t)v_clamped; 2265 } else { 2266 leftVol = mLeftVolShort; 2267 rightVol = mRightVolShort; 2268 rampVolume = false; 2269 } 2270 2271 // reset retry count 2272 track->mRetryCount = kMaxTrackRetriesDirect; 2273 activeTrack = t; 2274 mixerStatus = MIXER_TRACKS_READY; 2275 } else { 2276 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2277 if (track->isStopped()) { 2278 track->reset(); 2279 } 2280 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2281 // We have consumed all the buffers of this track. 2282 // Remove it from the list of active tracks. 2283 trackToRemove = track; 2284 } else { 2285 // No buffers for this track. Give it a few chances to 2286 // fill a buffer, then remove it from active list. 2287 if (--(track->mRetryCount) <= 0) { 2288 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2289 trackToRemove = track; 2290 } else { 2291 mixerStatus = MIXER_TRACKS_ENABLED; 2292 } 2293 } 2294 } 2295 } 2296 2297 // remove all the tracks that need to be... 2298 if (UNLIKELY(trackToRemove != 0)) { 2299 mActiveTracks.remove(trackToRemove); 2300 if (!effectChains.isEmpty()) { 2301 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2302 trackToRemove->sessionId()); 2303 effectChains[0]->stopTrack(); 2304 } 2305 if (trackToRemove->isTerminated()) { 2306 mTracks.remove(trackToRemove); 2307 deleteTrackName_l(trackToRemove->mName); 2308 } 2309 } 2310 2311 lockEffectChains_l(effectChains); 2312 } 2313 2314 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2315 AudioBufferProvider::Buffer buffer; 2316 size_t frameCount = mFrameCount; 2317 curBuf = (int8_t *)mMixBuffer; 2318 // output audio to hardware 2319 while (frameCount) { 2320 buffer.frameCount = frameCount; 2321 activeTrack->getNextBuffer(&buffer); 2322 if (UNLIKELY(buffer.raw == 0)) { 2323 memset(curBuf, 0, frameCount * mFrameSize); 2324 break; 2325 } 2326 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2327 frameCount -= buffer.frameCount; 2328 curBuf += buffer.frameCount * mFrameSize; 2329 activeTrack->releaseBuffer(&buffer); 2330 } 2331 sleepTime = 0; 2332 standbyTime = systemTime() + standbyDelay; 2333 } else { 2334 if (sleepTime == 0) { 2335 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2336 sleepTime = activeSleepTime; 2337 } else { 2338 sleepTime = idleSleepTime; 2339 } 2340 } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { 2341 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2342 sleepTime = 0; 2343 } 2344 } 2345 2346 if (mSuspended) { 2347 sleepTime = idleSleepTime; 2348 } 2349 // sleepTime == 0 means we must write to audio hardware 2350 if (sleepTime == 0) { 2351 if (mixerStatus == MIXER_TRACKS_READY) { 2352 applyVolume(leftVol, rightVol, rampVolume); 2353 } 2354 for (size_t i = 0; i < effectChains.size(); i ++) { 2355 effectChains[i]->process_l(); 2356 } 2357 unlockEffectChains(effectChains); 2358 2359 mLastWriteTime = systemTime(); 2360 mInWrite = true; 2361 mBytesWritten += mixBufferSize; 2362 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 2363 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2364 mNumWrites++; 2365 mInWrite = false; 2366 mStandby = false; 2367 } else { 2368 unlockEffectChains(effectChains); 2369 usleep(sleepTime); 2370 } 2371 2372 // finally let go of removed track, without the lock held 2373 // since we can't guarantee the destructors won't acquire that 2374 // same lock. 2375 trackToRemove.clear(); 2376 activeTrack.clear(); 2377 2378 // Effect chains will be actually deleted here if they were removed from 2379 // mEffectChains list during mixing or effects processing 2380 effectChains.clear(); 2381 } 2382 2383 if (!mStandby) { 2384 mOutput->standby(); 2385 } 2386 2387 LOGV("DirectOutputThread %p exiting", this); 2388 return false; 2389} 2390 2391// getTrackName_l() must be called with ThreadBase::mLock held 2392int AudioFlinger::DirectOutputThread::getTrackName_l() 2393{ 2394 return 0; 2395} 2396 2397// deleteTrackName_l() must be called with ThreadBase::mLock held 2398void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2399{ 2400} 2401 2402// checkForNewParameters_l() must be called with ThreadBase::mLock held 2403bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2404{ 2405 bool reconfig = false; 2406 2407 while (!mNewParameters.isEmpty()) { 2408 status_t status = NO_ERROR; 2409 String8 keyValuePair = mNewParameters[0]; 2410 AudioParameter param = AudioParameter(keyValuePair); 2411 int value; 2412 2413 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2414 // do not accept frame count changes if tracks are open as the track buffer 2415 // size depends on frame count and correct behavior would not be garantied 2416 // if frame count is changed after track creation 2417 if (!mTracks.isEmpty()) { 2418 status = INVALID_OPERATION; 2419 } else { 2420 reconfig = true; 2421 } 2422 } 2423 if (status == NO_ERROR) { 2424 status = mOutput->setParameters(keyValuePair); 2425 if (!mStandby && status == INVALID_OPERATION) { 2426 mOutput->standby(); 2427 mStandby = true; 2428 mBytesWritten = 0; 2429 status = mOutput->setParameters(keyValuePair); 2430 } 2431 if (status == NO_ERROR && reconfig) { 2432 readOutputParameters(); 2433 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2434 } 2435 } 2436 2437 mNewParameters.removeAt(0); 2438 2439 mParamStatus = status; 2440 mParamCond.signal(); 2441 mWaitWorkCV.wait(mLock); 2442 } 2443 return reconfig; 2444} 2445 2446uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2447{ 2448 uint32_t time; 2449 if (AudioSystem::isLinearPCM(mFormat)) { 2450 time = (uint32_t)(mOutput->latency() * 1000) / 2; 2451 } else { 2452 time = 10000; 2453 } 2454 return time; 2455} 2456 2457uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2458{ 2459 uint32_t time; 2460 if (AudioSystem::isLinearPCM(mFormat)) { 2461 time = (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000; 2462 } else { 2463 time = 10000; 2464 } 2465 return time; 2466} 2467 2468// ---------------------------------------------------------------------------- 2469 2470AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2471 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2472{ 2473 mType = PlaybackThread::DUPLICATING; 2474 addOutputTrack(mainThread); 2475} 2476 2477AudioFlinger::DuplicatingThread::~DuplicatingThread() 2478{ 2479 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2480 mOutputTracks[i]->destroy(); 2481 } 2482 mOutputTracks.clear(); 2483} 2484 2485bool AudioFlinger::DuplicatingThread::threadLoop() 2486{ 2487 Vector< sp<Track> > tracksToRemove; 2488 uint32_t mixerStatus = MIXER_IDLE; 2489 nsecs_t standbyTime = systemTime(); 2490 size_t mixBufferSize = mFrameCount*mFrameSize; 2491 SortedVector< sp<OutputTrack> > outputTracks; 2492 uint32_t writeFrames = 0; 2493 uint32_t activeSleepTime = activeSleepTimeUs(); 2494 uint32_t idleSleepTime = idleSleepTimeUs(); 2495 uint32_t sleepTime = idleSleepTime; 2496 Vector< sp<EffectChain> > effectChains; 2497 2498 while (!exitPending()) 2499 { 2500 processConfigEvents(); 2501 2502 mixerStatus = MIXER_IDLE; 2503 { // scope for the mLock 2504 2505 Mutex::Autolock _l(mLock); 2506 2507 if (checkForNewParameters_l()) { 2508 mixBufferSize = mFrameCount*mFrameSize; 2509 updateWaitTime(); 2510 activeSleepTime = activeSleepTimeUs(); 2511 idleSleepTime = idleSleepTimeUs(); 2512 } 2513 2514 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2515 2516 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2517 outputTracks.add(mOutputTracks[i]); 2518 } 2519 2520 // put audio hardware into standby after short delay 2521 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2522 mSuspended) { 2523 if (!mStandby) { 2524 for (size_t i = 0; i < outputTracks.size(); i++) { 2525 outputTracks[i]->stop(); 2526 } 2527 mStandby = true; 2528 mBytesWritten = 0; 2529 } 2530 2531 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2532 // we're about to wait, flush the binder command buffer 2533 IPCThreadState::self()->flushCommands(); 2534 outputTracks.clear(); 2535 2536 if (exitPending()) break; 2537 2538 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2539 mWaitWorkCV.wait(mLock); 2540 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2541 if (mMasterMute == false) { 2542 char value[PROPERTY_VALUE_MAX]; 2543 property_get("ro.audio.silent", value, "0"); 2544 if (atoi(value)) { 2545 LOGD("Silence is golden"); 2546 setMasterMute(true); 2547 } 2548 } 2549 2550 standbyTime = systemTime() + kStandbyTimeInNsecs; 2551 sleepTime = idleSleepTime; 2552 continue; 2553 } 2554 } 2555 2556 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2557 2558 // prevent any changes in effect chain list and in each effect chain 2559 // during mixing and effect process as the audio buffers could be deleted 2560 // or modified if an effect is created or deleted 2561 lockEffectChains_l(effectChains); 2562 } 2563 2564 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2565 // mix buffers... 2566 if (outputsReady(outputTracks)) { 2567 mAudioMixer->process(); 2568 } else { 2569 memset(mMixBuffer, 0, mixBufferSize); 2570 } 2571 sleepTime = 0; 2572 writeFrames = mFrameCount; 2573 } else { 2574 if (sleepTime == 0) { 2575 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2576 sleepTime = activeSleepTime; 2577 } else { 2578 sleepTime = idleSleepTime; 2579 } 2580 } else if (mBytesWritten != 0) { 2581 // flush remaining overflow buffers in output tracks 2582 for (size_t i = 0; i < outputTracks.size(); i++) { 2583 if (outputTracks[i]->isActive()) { 2584 sleepTime = 0; 2585 writeFrames = 0; 2586 memset(mMixBuffer, 0, mixBufferSize); 2587 break; 2588 } 2589 } 2590 } 2591 } 2592 2593 if (mSuspended) { 2594 sleepTime = idleSleepTime; 2595 } 2596 // sleepTime == 0 means we must write to audio hardware 2597 if (sleepTime == 0) { 2598 for (size_t i = 0; i < effectChains.size(); i ++) { 2599 effectChains[i]->process_l(); 2600 } 2601 // enable changes in effect chain 2602 unlockEffectChains(effectChains); 2603 2604 standbyTime = systemTime() + kStandbyTimeInNsecs; 2605 for (size_t i = 0; i < outputTracks.size(); i++) { 2606 outputTracks[i]->write(mMixBuffer, writeFrames); 2607 } 2608 mStandby = false; 2609 mBytesWritten += mixBufferSize; 2610 } else { 2611 // enable changes in effect chain 2612 unlockEffectChains(effectChains); 2613 usleep(sleepTime); 2614 } 2615 2616 // finally let go of all our tracks, without the lock held 2617 // since we can't guarantee the destructors won't acquire that 2618 // same lock. 2619 tracksToRemove.clear(); 2620 outputTracks.clear(); 2621 2622 // Effect chains will be actually deleted here if they were removed from 2623 // mEffectChains list during mixing or effects processing 2624 effectChains.clear(); 2625 } 2626 2627 return false; 2628} 2629 2630void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2631{ 2632 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2633 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2634 this, 2635 mSampleRate, 2636 mFormat, 2637 mChannelCount, 2638 frameCount); 2639 if (outputTrack->cblk() != NULL) { 2640 thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); 2641 mOutputTracks.add(outputTrack); 2642 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2643 updateWaitTime(); 2644 } 2645} 2646 2647void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2648{ 2649 Mutex::Autolock _l(mLock); 2650 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2651 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2652 mOutputTracks[i]->destroy(); 2653 mOutputTracks.removeAt(i); 2654 updateWaitTime(); 2655 return; 2656 } 2657 } 2658 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2659} 2660 2661void AudioFlinger::DuplicatingThread::updateWaitTime() 2662{ 2663 mWaitTimeMs = UINT_MAX; 2664 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2665 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2666 if (strong != NULL) { 2667 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2668 if (waitTimeMs < mWaitTimeMs) { 2669 mWaitTimeMs = waitTimeMs; 2670 } 2671 } 2672 } 2673} 2674 2675 2676bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2677{ 2678 for (size_t i = 0; i < outputTracks.size(); i++) { 2679 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2680 if (thread == 0) { 2681 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2682 return false; 2683 } 2684 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2685 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2686 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2687 return false; 2688 } 2689 } 2690 return true; 2691} 2692 2693uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2694{ 2695 return (mWaitTimeMs * 1000) / 2; 2696} 2697 2698// ---------------------------------------------------------------------------- 2699 2700// TrackBase constructor must be called with AudioFlinger::mLock held 2701AudioFlinger::ThreadBase::TrackBase::TrackBase( 2702 const wp<ThreadBase>& thread, 2703 const sp<Client>& client, 2704 uint32_t sampleRate, 2705 int format, 2706 int channelCount, 2707 int frameCount, 2708 uint32_t flags, 2709 const sp<IMemory>& sharedBuffer, 2710 int sessionId) 2711 : RefBase(), 2712 mThread(thread), 2713 mClient(client), 2714 mCblk(0), 2715 mFrameCount(0), 2716 mState(IDLE), 2717 mClientTid(-1), 2718 mFormat(format), 2719 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2720 mSessionId(sessionId) 2721{ 2722 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2723 2724 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2725 size_t size = sizeof(audio_track_cblk_t); 2726 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2727 if (sharedBuffer == 0) { 2728 size += bufferSize; 2729 } 2730 2731 if (client != NULL) { 2732 mCblkMemory = client->heap()->allocate(size); 2733 if (mCblkMemory != 0) { 2734 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2735 if (mCblk) { // construct the shared structure in-place. 2736 new(mCblk) audio_track_cblk_t(); 2737 // clear all buffers 2738 mCblk->frameCount = frameCount; 2739 mCblk->sampleRate = sampleRate; 2740 mCblk->channelCount = (uint8_t)channelCount; 2741 if (sharedBuffer == 0) { 2742 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2743 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2744 // Force underrun condition to avoid false underrun callback until first data is 2745 // written to buffer 2746 mCblk->flags = CBLK_UNDERRUN_ON; 2747 } else { 2748 mBuffer = sharedBuffer->pointer(); 2749 } 2750 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2751 } 2752 } else { 2753 LOGE("not enough memory for AudioTrack size=%u", size); 2754 client->heap()->dump("AudioTrack"); 2755 return; 2756 } 2757 } else { 2758 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2759 if (mCblk) { // construct the shared structure in-place. 2760 new(mCblk) audio_track_cblk_t(); 2761 // clear all buffers 2762 mCblk->frameCount = frameCount; 2763 mCblk->sampleRate = sampleRate; 2764 mCblk->channelCount = (uint8_t)channelCount; 2765 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2766 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2767 // Force underrun condition to avoid false underrun callback until first data is 2768 // written to buffer 2769 mCblk->flags = CBLK_UNDERRUN_ON; 2770 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2771 } 2772 } 2773} 2774 2775AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2776{ 2777 if (mCblk) { 2778 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2779 if (mClient == NULL) { 2780 delete mCblk; 2781 } 2782 } 2783 mCblkMemory.clear(); // and free the shared memory 2784 if (mClient != NULL) { 2785 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2786 mClient.clear(); 2787 } 2788} 2789 2790void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2791{ 2792 buffer->raw = 0; 2793 mFrameCount = buffer->frameCount; 2794 step(); 2795 buffer->frameCount = 0; 2796} 2797 2798bool AudioFlinger::ThreadBase::TrackBase::step() { 2799 bool result; 2800 audio_track_cblk_t* cblk = this->cblk(); 2801 2802 result = cblk->stepServer(mFrameCount); 2803 if (!result) { 2804 LOGV("stepServer failed acquiring cblk mutex"); 2805 mFlags |= STEPSERVER_FAILED; 2806 } 2807 return result; 2808} 2809 2810void AudioFlinger::ThreadBase::TrackBase::reset() { 2811 audio_track_cblk_t* cblk = this->cblk(); 2812 2813 cblk->user = 0; 2814 cblk->server = 0; 2815 cblk->userBase = 0; 2816 cblk->serverBase = 0; 2817 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2818 LOGV("TrackBase::reset"); 2819} 2820 2821sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2822{ 2823 return mCblkMemory; 2824} 2825 2826int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2827 return (int)mCblk->sampleRate; 2828} 2829 2830int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2831 return (int)mCblk->channelCount; 2832} 2833 2834void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2835 audio_track_cblk_t* cblk = this->cblk(); 2836 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2837 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2838 2839 // Check validity of returned pointer in case the track control block would have been corrupted. 2840 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2841 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2842 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2843 server %d, serverBase %d, user %d, userBase %d, channelCount %d", 2844 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2845 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); 2846 return 0; 2847 } 2848 2849 return bufferStart; 2850} 2851 2852// ---------------------------------------------------------------------------- 2853 2854// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2855AudioFlinger::PlaybackThread::Track::Track( 2856 const wp<ThreadBase>& thread, 2857 const sp<Client>& client, 2858 int streamType, 2859 uint32_t sampleRate, 2860 int format, 2861 int channelCount, 2862 int frameCount, 2863 const sp<IMemory>& sharedBuffer, 2864 int sessionId) 2865 : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), 2866 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), mAuxEffectId(0) 2867{ 2868 if (mCblk != NULL) { 2869 sp<ThreadBase> baseThread = thread.promote(); 2870 if (baseThread != 0) { 2871 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2872 mName = playbackThread->getTrackName_l(); 2873 mMainBuffer = playbackThread->mixBuffer(); 2874 } 2875 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2876 if (mName < 0) { 2877 LOGE("no more track names available"); 2878 } 2879 mVolume[0] = 1.0f; 2880 mVolume[1] = 1.0f; 2881 mStreamType = streamType; 2882 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2883 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2884 mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); 2885 } 2886} 2887 2888AudioFlinger::PlaybackThread::Track::~Track() 2889{ 2890 LOGV("PlaybackThread::Track destructor"); 2891 sp<ThreadBase> thread = mThread.promote(); 2892 if (thread != 0) { 2893 Mutex::Autolock _l(thread->mLock); 2894 mState = TERMINATED; 2895 } 2896} 2897 2898void AudioFlinger::PlaybackThread::Track::destroy() 2899{ 2900 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2901 // by removing it from mTracks vector, so there is a risk that this Tracks's 2902 // desctructor is called. As the destructor needs to lock mLock, 2903 // we must acquire a strong reference on this Track before locking mLock 2904 // here so that the destructor is called only when exiting this function. 2905 // On the other hand, as long as Track::destroy() is only called by 2906 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2907 // this Track with its member mTrack. 2908 sp<Track> keep(this); 2909 { // scope for mLock 2910 sp<ThreadBase> thread = mThread.promote(); 2911 if (thread != 0) { 2912 if (!isOutputTrack()) { 2913 if (mState == ACTIVE || mState == RESUMING) { 2914 AudioSystem::stopOutput(thread->id(), 2915 (AudioSystem::stream_type)mStreamType, 2916 mSessionId); 2917 } 2918 AudioSystem::releaseOutput(thread->id()); 2919 } 2920 Mutex::Autolock _l(thread->mLock); 2921 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2922 playbackThread->destroyTrack_l(this); 2923 } 2924 } 2925} 2926 2927void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 2928{ 2929 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", 2930 mName - AudioMixer::TRACK0, 2931 (mClient == NULL) ? getpid() : mClient->pid(), 2932 mStreamType, 2933 mFormat, 2934 mCblk->channelCount, 2935 mSessionId, 2936 mFrameCount, 2937 mState, 2938 mMute, 2939 mFillingUpStatus, 2940 mCblk->sampleRate, 2941 mCblk->volume[0], 2942 mCblk->volume[1], 2943 mCblk->server, 2944 mCblk->user, 2945 (int)mMainBuffer, 2946 (int)mAuxBuffer); 2947} 2948 2949status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 2950{ 2951 audio_track_cblk_t* cblk = this->cblk(); 2952 uint32_t framesReady; 2953 uint32_t framesReq = buffer->frameCount; 2954 2955 // Check if last stepServer failed, try to step now 2956 if (mFlags & TrackBase::STEPSERVER_FAILED) { 2957 if (!step()) goto getNextBuffer_exit; 2958 LOGV("stepServer recovered"); 2959 mFlags &= ~TrackBase::STEPSERVER_FAILED; 2960 } 2961 2962 framesReady = cblk->framesReady(); 2963 2964 if (LIKELY(framesReady)) { 2965 uint32_t s = cblk->server; 2966 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 2967 2968 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 2969 if (framesReq > framesReady) { 2970 framesReq = framesReady; 2971 } 2972 if (s + framesReq > bufferEnd) { 2973 framesReq = bufferEnd - s; 2974 } 2975 2976 buffer->raw = getBuffer(s, framesReq); 2977 if (buffer->raw == 0) goto getNextBuffer_exit; 2978 2979 buffer->frameCount = framesReq; 2980 return NO_ERROR; 2981 } 2982 2983getNextBuffer_exit: 2984 buffer->raw = 0; 2985 buffer->frameCount = 0; 2986 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 2987 return NOT_ENOUGH_DATA; 2988} 2989 2990bool AudioFlinger::PlaybackThread::Track::isReady() const { 2991 if (mFillingUpStatus != FS_FILLING) return true; 2992 2993 if (mCblk->framesReady() >= mCblk->frameCount || 2994 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 2995 mFillingUpStatus = FS_FILLED; 2996 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 2997 return true; 2998 } 2999 return false; 3000} 3001 3002status_t AudioFlinger::PlaybackThread::Track::start() 3003{ 3004 status_t status = NO_ERROR; 3005 LOGV("start(%d), calling thread %d session %d", 3006 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3007 sp<ThreadBase> thread = mThread.promote(); 3008 if (thread != 0) { 3009 Mutex::Autolock _l(thread->mLock); 3010 int state = mState; 3011 // here the track could be either new, or restarted 3012 // in both cases "unstop" the track 3013 if (mState == PAUSED) { 3014 mState = TrackBase::RESUMING; 3015 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3016 } else { 3017 mState = TrackBase::ACTIVE; 3018 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3019 } 3020 3021 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3022 thread->mLock.unlock(); 3023 status = AudioSystem::startOutput(thread->id(), 3024 (AudioSystem::stream_type)mStreamType, 3025 mSessionId); 3026 thread->mLock.lock(); 3027 } 3028 if (status == NO_ERROR) { 3029 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3030 playbackThread->addTrack_l(this); 3031 } else { 3032 mState = state; 3033 } 3034 } else { 3035 status = BAD_VALUE; 3036 } 3037 return status; 3038} 3039 3040void AudioFlinger::PlaybackThread::Track::stop() 3041{ 3042 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3043 sp<ThreadBase> thread = mThread.promote(); 3044 if (thread != 0) { 3045 Mutex::Autolock _l(thread->mLock); 3046 int state = mState; 3047 if (mState > STOPPED) { 3048 mState = STOPPED; 3049 // If the track is not active (PAUSED and buffers full), flush buffers 3050 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3051 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3052 reset(); 3053 } 3054 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3055 } 3056 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3057 thread->mLock.unlock(); 3058 AudioSystem::stopOutput(thread->id(), 3059 (AudioSystem::stream_type)mStreamType, 3060 mSessionId); 3061 thread->mLock.lock(); 3062 } 3063 } 3064} 3065 3066void AudioFlinger::PlaybackThread::Track::pause() 3067{ 3068 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3069 sp<ThreadBase> thread = mThread.promote(); 3070 if (thread != 0) { 3071 Mutex::Autolock _l(thread->mLock); 3072 if (mState == ACTIVE || mState == RESUMING) { 3073 mState = PAUSING; 3074 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3075 if (!isOutputTrack()) { 3076 thread->mLock.unlock(); 3077 AudioSystem::stopOutput(thread->id(), 3078 (AudioSystem::stream_type)mStreamType, 3079 mSessionId); 3080 thread->mLock.lock(); 3081 } 3082 } 3083 } 3084} 3085 3086void AudioFlinger::PlaybackThread::Track::flush() 3087{ 3088 LOGV("flush(%d)", mName); 3089 sp<ThreadBase> thread = mThread.promote(); 3090 if (thread != 0) { 3091 Mutex::Autolock _l(thread->mLock); 3092 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3093 return; 3094 } 3095 // No point remaining in PAUSED state after a flush => go to 3096 // STOPPED state 3097 mState = STOPPED; 3098 3099 mCblk->lock.lock(); 3100 // NOTE: reset() will reset cblk->user and cblk->server with 3101 // the risk that at the same time, the AudioMixer is trying to read 3102 // data. In this case, getNextBuffer() would return a NULL pointer 3103 // as audio buffer => the AudioMixer code MUST always test that pointer 3104 // returned by getNextBuffer() is not NULL! 3105 reset(); 3106 mCblk->lock.unlock(); 3107 } 3108} 3109 3110void AudioFlinger::PlaybackThread::Track::reset() 3111{ 3112 // Do not reset twice to avoid discarding data written just after a flush and before 3113 // the audioflinger thread detects the track is stopped. 3114 if (!mResetDone) { 3115 TrackBase::reset(); 3116 // Force underrun condition to avoid false underrun callback until first data is 3117 // written to buffer 3118 mCblk->flags |= CBLK_UNDERRUN_ON; 3119 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3120 mFillingUpStatus = FS_FILLING; 3121 mResetDone = true; 3122 } 3123} 3124 3125void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3126{ 3127 mMute = muted; 3128} 3129 3130void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3131{ 3132 mVolume[0] = left; 3133 mVolume[1] = right; 3134} 3135 3136status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3137{ 3138 status_t status = DEAD_OBJECT; 3139 sp<ThreadBase> thread = mThread.promote(); 3140 if (thread != 0) { 3141 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3142 status = playbackThread->attachAuxEffect(this, EffectId); 3143 } 3144 return status; 3145} 3146 3147void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3148{ 3149 mAuxEffectId = EffectId; 3150 mAuxBuffer = buffer; 3151} 3152 3153// ---------------------------------------------------------------------------- 3154 3155// RecordTrack constructor must be called with AudioFlinger::mLock held 3156AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3157 const wp<ThreadBase>& thread, 3158 const sp<Client>& client, 3159 uint32_t sampleRate, 3160 int format, 3161 int channelCount, 3162 int frameCount, 3163 uint32_t flags, 3164 int sessionId) 3165 : TrackBase(thread, client, sampleRate, format, 3166 channelCount, frameCount, flags, 0, sessionId), 3167 mOverflow(false) 3168{ 3169 if (mCblk != NULL) { 3170 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3171 if (format == AudioSystem::PCM_16_BIT) { 3172 mCblk->frameSize = channelCount * sizeof(int16_t); 3173 } else if (format == AudioSystem::PCM_8_BIT) { 3174 mCblk->frameSize = channelCount * sizeof(int8_t); 3175 } else { 3176 mCblk->frameSize = sizeof(int8_t); 3177 } 3178 } 3179} 3180 3181AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3182{ 3183 sp<ThreadBase> thread = mThread.promote(); 3184 if (thread != 0) { 3185 AudioSystem::releaseInput(thread->id()); 3186 } 3187} 3188 3189status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3190{ 3191 audio_track_cblk_t* cblk = this->cblk(); 3192 uint32_t framesAvail; 3193 uint32_t framesReq = buffer->frameCount; 3194 3195 // Check if last stepServer failed, try to step now 3196 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3197 if (!step()) goto getNextBuffer_exit; 3198 LOGV("stepServer recovered"); 3199 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3200 } 3201 3202 framesAvail = cblk->framesAvailable_l(); 3203 3204 if (LIKELY(framesAvail)) { 3205 uint32_t s = cblk->server; 3206 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3207 3208 if (framesReq > framesAvail) { 3209 framesReq = framesAvail; 3210 } 3211 if (s + framesReq > bufferEnd) { 3212 framesReq = bufferEnd - s; 3213 } 3214 3215 buffer->raw = getBuffer(s, framesReq); 3216 if (buffer->raw == 0) goto getNextBuffer_exit; 3217 3218 buffer->frameCount = framesReq; 3219 return NO_ERROR; 3220 } 3221 3222getNextBuffer_exit: 3223 buffer->raw = 0; 3224 buffer->frameCount = 0; 3225 return NOT_ENOUGH_DATA; 3226} 3227 3228status_t AudioFlinger::RecordThread::RecordTrack::start() 3229{ 3230 sp<ThreadBase> thread = mThread.promote(); 3231 if (thread != 0) { 3232 RecordThread *recordThread = (RecordThread *)thread.get(); 3233 return recordThread->start(this); 3234 } else { 3235 return BAD_VALUE; 3236 } 3237} 3238 3239void AudioFlinger::RecordThread::RecordTrack::stop() 3240{ 3241 sp<ThreadBase> thread = mThread.promote(); 3242 if (thread != 0) { 3243 RecordThread *recordThread = (RecordThread *)thread.get(); 3244 recordThread->stop(this); 3245 TrackBase::reset(); 3246 // Force overerrun condition to avoid false overrun callback until first data is 3247 // read from buffer 3248 mCblk->flags |= CBLK_UNDERRUN_ON; 3249 } 3250} 3251 3252void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3253{ 3254 snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", 3255 (mClient == NULL) ? getpid() : mClient->pid(), 3256 mFormat, 3257 mCblk->channelCount, 3258 mSessionId, 3259 mFrameCount, 3260 mState, 3261 mCblk->sampleRate, 3262 mCblk->server, 3263 mCblk->user); 3264} 3265 3266 3267// ---------------------------------------------------------------------------- 3268 3269AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3270 const wp<ThreadBase>& thread, 3271 DuplicatingThread *sourceThread, 3272 uint32_t sampleRate, 3273 int format, 3274 int channelCount, 3275 int frameCount) 3276 : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), 3277 mActive(false), mSourceThread(sourceThread) 3278{ 3279 3280 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3281 if (mCblk != NULL) { 3282 mCblk->flags |= CBLK_DIRECTION_OUT; 3283 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3284 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3285 mOutBuffer.frameCount = 0; 3286 playbackThread->mTracks.add(this); 3287 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", 3288 mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); 3289 } else { 3290 LOGW("Error creating output track on thread %p", playbackThread); 3291 } 3292} 3293 3294AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3295{ 3296 clearBufferQueue(); 3297} 3298 3299status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3300{ 3301 status_t status = Track::start(); 3302 if (status != NO_ERROR) { 3303 return status; 3304 } 3305 3306 mActive = true; 3307 mRetryCount = 127; 3308 return status; 3309} 3310 3311void AudioFlinger::PlaybackThread::OutputTrack::stop() 3312{ 3313 Track::stop(); 3314 clearBufferQueue(); 3315 mOutBuffer.frameCount = 0; 3316 mActive = false; 3317} 3318 3319bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3320{ 3321 Buffer *pInBuffer; 3322 Buffer inBuffer; 3323 uint32_t channelCount = mCblk->channelCount; 3324 bool outputBufferFull = false; 3325 inBuffer.frameCount = frames; 3326 inBuffer.i16 = data; 3327 3328 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3329 3330 if (!mActive && frames != 0) { 3331 start(); 3332 sp<ThreadBase> thread = mThread.promote(); 3333 if (thread != 0) { 3334 MixerThread *mixerThread = (MixerThread *)thread.get(); 3335 if (mCblk->frameCount > frames){ 3336 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3337 uint32_t startFrames = (mCblk->frameCount - frames); 3338 pInBuffer = new Buffer; 3339 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3340 pInBuffer->frameCount = startFrames; 3341 pInBuffer->i16 = pInBuffer->mBuffer; 3342 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3343 mBufferQueue.add(pInBuffer); 3344 } else { 3345 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3346 } 3347 } 3348 } 3349 } 3350 3351 while (waitTimeLeftMs) { 3352 // First write pending buffers, then new data 3353 if (mBufferQueue.size()) { 3354 pInBuffer = mBufferQueue.itemAt(0); 3355 } else { 3356 pInBuffer = &inBuffer; 3357 } 3358 3359 if (pInBuffer->frameCount == 0) { 3360 break; 3361 } 3362 3363 if (mOutBuffer.frameCount == 0) { 3364 mOutBuffer.frameCount = pInBuffer->frameCount; 3365 nsecs_t startTime = systemTime(); 3366 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3367 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3368 outputBufferFull = true; 3369 break; 3370 } 3371 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3372 if (waitTimeLeftMs >= waitTimeMs) { 3373 waitTimeLeftMs -= waitTimeMs; 3374 } else { 3375 waitTimeLeftMs = 0; 3376 } 3377 } 3378 3379 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3380 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3381 mCblk->stepUser(outFrames); 3382 pInBuffer->frameCount -= outFrames; 3383 pInBuffer->i16 += outFrames * channelCount; 3384 mOutBuffer.frameCount -= outFrames; 3385 mOutBuffer.i16 += outFrames * channelCount; 3386 3387 if (pInBuffer->frameCount == 0) { 3388 if (mBufferQueue.size()) { 3389 mBufferQueue.removeAt(0); 3390 delete [] pInBuffer->mBuffer; 3391 delete pInBuffer; 3392 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3393 } else { 3394 break; 3395 } 3396 } 3397 } 3398 3399 // If we could not write all frames, allocate a buffer and queue it for next time. 3400 if (inBuffer.frameCount) { 3401 sp<ThreadBase> thread = mThread.promote(); 3402 if (thread != 0 && !thread->standby()) { 3403 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3404 pInBuffer = new Buffer; 3405 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3406 pInBuffer->frameCount = inBuffer.frameCount; 3407 pInBuffer->i16 = pInBuffer->mBuffer; 3408 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3409 mBufferQueue.add(pInBuffer); 3410 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3411 } else { 3412 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3413 } 3414 } 3415 } 3416 3417 // Calling write() with a 0 length buffer, means that no more data will be written: 3418 // If no more buffers are pending, fill output track buffer to make sure it is started 3419 // by output mixer. 3420 if (frames == 0 && mBufferQueue.size() == 0) { 3421 if (mCblk->user < mCblk->frameCount) { 3422 frames = mCblk->frameCount - mCblk->user; 3423 pInBuffer = new Buffer; 3424 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3425 pInBuffer->frameCount = frames; 3426 pInBuffer->i16 = pInBuffer->mBuffer; 3427 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3428 mBufferQueue.add(pInBuffer); 3429 } else if (mActive) { 3430 stop(); 3431 } 3432 } 3433 3434 return outputBufferFull; 3435} 3436 3437status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3438{ 3439 int active; 3440 status_t result; 3441 audio_track_cblk_t* cblk = mCblk; 3442 uint32_t framesReq = buffer->frameCount; 3443 3444// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3445 buffer->frameCount = 0; 3446 3447 uint32_t framesAvail = cblk->framesAvailable(); 3448 3449 3450 if (framesAvail == 0) { 3451 Mutex::Autolock _l(cblk->lock); 3452 goto start_loop_here; 3453 while (framesAvail == 0) { 3454 active = mActive; 3455 if (UNLIKELY(!active)) { 3456 LOGV("Not active and NO_MORE_BUFFERS"); 3457 return AudioTrack::NO_MORE_BUFFERS; 3458 } 3459 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3460 if (result != NO_ERROR) { 3461 return AudioTrack::NO_MORE_BUFFERS; 3462 } 3463 // read the server count again 3464 start_loop_here: 3465 framesAvail = cblk->framesAvailable_l(); 3466 } 3467 } 3468 3469// if (framesAvail < framesReq) { 3470// return AudioTrack::NO_MORE_BUFFERS; 3471// } 3472 3473 if (framesReq > framesAvail) { 3474 framesReq = framesAvail; 3475 } 3476 3477 uint32_t u = cblk->user; 3478 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3479 3480 if (u + framesReq > bufferEnd) { 3481 framesReq = bufferEnd - u; 3482 } 3483 3484 buffer->frameCount = framesReq; 3485 buffer->raw = (void *)cblk->buffer(u); 3486 return NO_ERROR; 3487} 3488 3489 3490void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3491{ 3492 size_t size = mBufferQueue.size(); 3493 Buffer *pBuffer; 3494 3495 for (size_t i = 0; i < size; i++) { 3496 pBuffer = mBufferQueue.itemAt(i); 3497 delete [] pBuffer->mBuffer; 3498 delete pBuffer; 3499 } 3500 mBufferQueue.clear(); 3501} 3502 3503// ---------------------------------------------------------------------------- 3504 3505AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3506 : RefBase(), 3507 mAudioFlinger(audioFlinger), 3508 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3509 mPid(pid) 3510{ 3511 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3512} 3513 3514// Client destructor must be called with AudioFlinger::mLock held 3515AudioFlinger::Client::~Client() 3516{ 3517 mAudioFlinger->removeClient_l(mPid); 3518} 3519 3520const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3521{ 3522 return mMemoryDealer; 3523} 3524 3525// ---------------------------------------------------------------------------- 3526 3527AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3528 const sp<IAudioFlingerClient>& client, 3529 pid_t pid) 3530 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3531{ 3532} 3533 3534AudioFlinger::NotificationClient::~NotificationClient() 3535{ 3536 mClient.clear(); 3537} 3538 3539void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3540{ 3541 sp<NotificationClient> keep(this); 3542 { 3543 mAudioFlinger->removeNotificationClient(mPid); 3544 } 3545} 3546 3547// ---------------------------------------------------------------------------- 3548 3549AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3550 : BnAudioTrack(), 3551 mTrack(track) 3552{ 3553} 3554 3555AudioFlinger::TrackHandle::~TrackHandle() { 3556 // just stop the track on deletion, associated resources 3557 // will be freed from the main thread once all pending buffers have 3558 // been played. Unless it's not in the active track list, in which 3559 // case we free everything now... 3560 mTrack->destroy(); 3561} 3562 3563status_t AudioFlinger::TrackHandle::start() { 3564 return mTrack->start(); 3565} 3566 3567void AudioFlinger::TrackHandle::stop() { 3568 mTrack->stop(); 3569} 3570 3571void AudioFlinger::TrackHandle::flush() { 3572 mTrack->flush(); 3573} 3574 3575void AudioFlinger::TrackHandle::mute(bool e) { 3576 mTrack->mute(e); 3577} 3578 3579void AudioFlinger::TrackHandle::pause() { 3580 mTrack->pause(); 3581} 3582 3583void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3584 mTrack->setVolume(left, right); 3585} 3586 3587sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3588 return mTrack->getCblk(); 3589} 3590 3591status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3592{ 3593 return mTrack->attachAuxEffect(EffectId); 3594} 3595 3596status_t AudioFlinger::TrackHandle::onTransact( 3597 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3598{ 3599 return BnAudioTrack::onTransact(code, data, reply, flags); 3600} 3601 3602// ---------------------------------------------------------------------------- 3603 3604sp<IAudioRecord> AudioFlinger::openRecord( 3605 pid_t pid, 3606 int input, 3607 uint32_t sampleRate, 3608 int format, 3609 int channelCount, 3610 int frameCount, 3611 uint32_t flags, 3612 int *sessionId, 3613 status_t *status) 3614{ 3615 sp<RecordThread::RecordTrack> recordTrack; 3616 sp<RecordHandle> recordHandle; 3617 sp<Client> client; 3618 wp<Client> wclient; 3619 status_t lStatus; 3620 RecordThread *thread; 3621 size_t inFrameCount; 3622 int lSessionId; 3623 3624 // check calling permissions 3625 if (!recordingAllowed()) { 3626 lStatus = PERMISSION_DENIED; 3627 goto Exit; 3628 } 3629 3630 // add client to list 3631 { // scope for mLock 3632 Mutex::Autolock _l(mLock); 3633 thread = checkRecordThread_l(input); 3634 if (thread == NULL) { 3635 lStatus = BAD_VALUE; 3636 goto Exit; 3637 } 3638 3639 wclient = mClients.valueFor(pid); 3640 if (wclient != NULL) { 3641 client = wclient.promote(); 3642 } else { 3643 client = new Client(this, pid); 3644 mClients.add(pid, client); 3645 } 3646 3647 // If no audio session id is provided, create one here 3648 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 3649 lSessionId = *sessionId; 3650 } else { 3651 lSessionId = nextUniqueId(); 3652 if (sessionId != NULL) { 3653 *sessionId = lSessionId; 3654 } 3655 } 3656 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3657 recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, 3658 format, channelCount, frameCount, flags, lSessionId); 3659 } 3660 if (recordTrack->getCblk() == NULL) { 3661 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3662 // destructor is called by the TrackBase destructor with mLock held 3663 client.clear(); 3664 recordTrack.clear(); 3665 lStatus = NO_MEMORY; 3666 goto Exit; 3667 } 3668 3669 // return to handle to client 3670 recordHandle = new RecordHandle(recordTrack); 3671 lStatus = NO_ERROR; 3672 3673Exit: 3674 if (status) { 3675 *status = lStatus; 3676 } 3677 return recordHandle; 3678} 3679 3680// ---------------------------------------------------------------------------- 3681 3682AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3683 : BnAudioRecord(), 3684 mRecordTrack(recordTrack) 3685{ 3686} 3687 3688AudioFlinger::RecordHandle::~RecordHandle() { 3689 stop(); 3690} 3691 3692status_t AudioFlinger::RecordHandle::start() { 3693 LOGV("RecordHandle::start()"); 3694 return mRecordTrack->start(); 3695} 3696 3697void AudioFlinger::RecordHandle::stop() { 3698 LOGV("RecordHandle::stop()"); 3699 mRecordTrack->stop(); 3700} 3701 3702sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3703 return mRecordTrack->getCblk(); 3704} 3705 3706status_t AudioFlinger::RecordHandle::onTransact( 3707 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3708{ 3709 return BnAudioRecord::onTransact(code, data, reply, flags); 3710} 3711 3712// ---------------------------------------------------------------------------- 3713 3714AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : 3715 ThreadBase(audioFlinger, id), 3716 mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3717{ 3718 mReqChannelCount = AudioSystem::popCount(channels); 3719 mReqSampleRate = sampleRate; 3720 readInputParameters(); 3721} 3722 3723 3724AudioFlinger::RecordThread::~RecordThread() 3725{ 3726 delete[] mRsmpInBuffer; 3727 if (mResampler != 0) { 3728 delete mResampler; 3729 delete[] mRsmpOutBuffer; 3730 } 3731} 3732 3733void AudioFlinger::RecordThread::onFirstRef() 3734{ 3735 const size_t SIZE = 256; 3736 char buffer[SIZE]; 3737 3738 snprintf(buffer, SIZE, "Record Thread %p", this); 3739 3740 run(buffer, PRIORITY_URGENT_AUDIO); 3741} 3742 3743bool AudioFlinger::RecordThread::threadLoop() 3744{ 3745 AudioBufferProvider::Buffer buffer; 3746 sp<RecordTrack> activeTrack; 3747 3748 // start recording 3749 while (!exitPending()) { 3750 3751 processConfigEvents(); 3752 3753 { // scope for mLock 3754 Mutex::Autolock _l(mLock); 3755 checkForNewParameters_l(); 3756 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3757 if (!mStandby) { 3758 mInput->standby(); 3759 mStandby = true; 3760 } 3761 3762 if (exitPending()) break; 3763 3764 LOGV("RecordThread: loop stopping"); 3765 // go to sleep 3766 mWaitWorkCV.wait(mLock); 3767 LOGV("RecordThread: loop starting"); 3768 continue; 3769 } 3770 if (mActiveTrack != 0) { 3771 if (mActiveTrack->mState == TrackBase::PAUSING) { 3772 if (!mStandby) { 3773 mInput->standby(); 3774 mStandby = true; 3775 } 3776 mActiveTrack.clear(); 3777 mStartStopCond.broadcast(); 3778 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3779 if (mReqChannelCount != mActiveTrack->channelCount()) { 3780 mActiveTrack.clear(); 3781 mStartStopCond.broadcast(); 3782 } else if (mBytesRead != 0) { 3783 // record start succeeds only if first read from audio input 3784 // succeeds 3785 if (mBytesRead > 0) { 3786 mActiveTrack->mState = TrackBase::ACTIVE; 3787 } else { 3788 mActiveTrack.clear(); 3789 } 3790 mStartStopCond.broadcast(); 3791 } 3792 mStandby = false; 3793 } 3794 } 3795 } 3796 3797 if (mActiveTrack != 0) { 3798 if (mActiveTrack->mState != TrackBase::ACTIVE && 3799 mActiveTrack->mState != TrackBase::RESUMING) { 3800 usleep(5000); 3801 continue; 3802 } 3803 buffer.frameCount = mFrameCount; 3804 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3805 size_t framesOut = buffer.frameCount; 3806 if (mResampler == 0) { 3807 // no resampling 3808 while (framesOut) { 3809 size_t framesIn = mFrameCount - mRsmpInIndex; 3810 if (framesIn) { 3811 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3812 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3813 if (framesIn > framesOut) 3814 framesIn = framesOut; 3815 mRsmpInIndex += framesIn; 3816 framesOut -= framesIn; 3817 if ((int)mChannelCount == mReqChannelCount || 3818 mFormat != AudioSystem::PCM_16_BIT) { 3819 memcpy(dst, src, framesIn * mFrameSize); 3820 } else { 3821 int16_t *src16 = (int16_t *)src; 3822 int16_t *dst16 = (int16_t *)dst; 3823 if (mChannelCount == 1) { 3824 while (framesIn--) { 3825 *dst16++ = *src16; 3826 *dst16++ = *src16++; 3827 } 3828 } else { 3829 while (framesIn--) { 3830 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3831 src16 += 2; 3832 } 3833 } 3834 } 3835 } 3836 if (framesOut && mFrameCount == mRsmpInIndex) { 3837 if (framesOut == mFrameCount && 3838 ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { 3839 mBytesRead = mInput->read(buffer.raw, mInputBytes); 3840 framesOut = 0; 3841 } else { 3842 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 3843 mRsmpInIndex = 0; 3844 } 3845 if (mBytesRead < 0) { 3846 LOGE("Error reading audio input"); 3847 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3848 // Force input into standby so that it tries to 3849 // recover at next read attempt 3850 mInput->standby(); 3851 usleep(5000); 3852 } 3853 mRsmpInIndex = mFrameCount; 3854 framesOut = 0; 3855 buffer.frameCount = 0; 3856 } 3857 } 3858 } 3859 } else { 3860 // resampling 3861 3862 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3863 // alter output frame count as if we were expecting stereo samples 3864 if (mChannelCount == 1 && mReqChannelCount == 1) { 3865 framesOut >>= 1; 3866 } 3867 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3868 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3869 // are 32 bit aligned which should be always true. 3870 if (mChannelCount == 2 && mReqChannelCount == 1) { 3871 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3872 // the resampler always outputs stereo samples: do post stereo to mono conversion 3873 int16_t *src = (int16_t *)mRsmpOutBuffer; 3874 int16_t *dst = buffer.i16; 3875 while (framesOut--) { 3876 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3877 src += 2; 3878 } 3879 } else { 3880 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3881 } 3882 3883 } 3884 mActiveTrack->releaseBuffer(&buffer); 3885 mActiveTrack->overflow(); 3886 } 3887 // client isn't retrieving buffers fast enough 3888 else { 3889 if (!mActiveTrack->setOverflow()) 3890 LOGW("RecordThread: buffer overflow"); 3891 // Release the processor for a while before asking for a new buffer. 3892 // This will give the application more chance to read from the buffer and 3893 // clear the overflow. 3894 usleep(5000); 3895 } 3896 } 3897 } 3898 3899 if (!mStandby) { 3900 mInput->standby(); 3901 } 3902 mActiveTrack.clear(); 3903 3904 mStartStopCond.broadcast(); 3905 3906 LOGV("RecordThread %p exiting", this); 3907 return false; 3908} 3909 3910status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 3911{ 3912 LOGV("RecordThread::start"); 3913 sp <ThreadBase> strongMe = this; 3914 status_t status = NO_ERROR; 3915 { 3916 AutoMutex lock(&mLock); 3917 if (mActiveTrack != 0) { 3918 if (recordTrack != mActiveTrack.get()) { 3919 status = -EBUSY; 3920 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3921 mActiveTrack->mState = TrackBase::ACTIVE; 3922 } 3923 return status; 3924 } 3925 3926 recordTrack->mState = TrackBase::IDLE; 3927 mActiveTrack = recordTrack; 3928 mLock.unlock(); 3929 status_t status = AudioSystem::startInput(mId); 3930 mLock.lock(); 3931 if (status != NO_ERROR) { 3932 mActiveTrack.clear(); 3933 return status; 3934 } 3935 mActiveTrack->mState = TrackBase::RESUMING; 3936 mRsmpInIndex = mFrameCount; 3937 mBytesRead = 0; 3938 // signal thread to start 3939 LOGV("Signal record thread"); 3940 mWaitWorkCV.signal(); 3941 // do not wait for mStartStopCond if exiting 3942 if (mExiting) { 3943 mActiveTrack.clear(); 3944 status = INVALID_OPERATION; 3945 goto startError; 3946 } 3947 mStartStopCond.wait(mLock); 3948 if (mActiveTrack == 0) { 3949 LOGV("Record failed to start"); 3950 status = BAD_VALUE; 3951 goto startError; 3952 } 3953 LOGV("Record started OK"); 3954 return status; 3955 } 3956startError: 3957 AudioSystem::stopInput(mId); 3958 return status; 3959} 3960 3961void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 3962 LOGV("RecordThread::stop"); 3963 sp <ThreadBase> strongMe = this; 3964 { 3965 AutoMutex lock(&mLock); 3966 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 3967 mActiveTrack->mState = TrackBase::PAUSING; 3968 // do not wait for mStartStopCond if exiting 3969 if (mExiting) { 3970 return; 3971 } 3972 mStartStopCond.wait(mLock); 3973 // if we have been restarted, recordTrack == mActiveTrack.get() here 3974 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 3975 mLock.unlock(); 3976 AudioSystem::stopInput(mId); 3977 mLock.lock(); 3978 LOGV("Record stopped OK"); 3979 } 3980 } 3981 } 3982} 3983 3984status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 3985{ 3986 const size_t SIZE = 256; 3987 char buffer[SIZE]; 3988 String8 result; 3989 pid_t pid = 0; 3990 3991 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 3992 result.append(buffer); 3993 3994 if (mActiveTrack != 0) { 3995 result.append("Active Track:\n"); 3996 result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); 3997 mActiveTrack->dump(buffer, SIZE); 3998 result.append(buffer); 3999 4000 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4001 result.append(buffer); 4002 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4003 result.append(buffer); 4004 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4005 result.append(buffer); 4006 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4007 result.append(buffer); 4008 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4009 result.append(buffer); 4010 4011 4012 } else { 4013 result.append("No record client\n"); 4014 } 4015 write(fd, result.string(), result.size()); 4016 4017 dumpBase(fd, args); 4018 4019 return NO_ERROR; 4020} 4021 4022status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4023{ 4024 size_t framesReq = buffer->frameCount; 4025 size_t framesReady = mFrameCount - mRsmpInIndex; 4026 int channelCount; 4027 4028 if (framesReady == 0) { 4029 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 4030 if (mBytesRead < 0) { 4031 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4032 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4033 // Force input into standby so that it tries to 4034 // recover at next read attempt 4035 mInput->standby(); 4036 usleep(5000); 4037 } 4038 buffer->raw = 0; 4039 buffer->frameCount = 0; 4040 return NOT_ENOUGH_DATA; 4041 } 4042 mRsmpInIndex = 0; 4043 framesReady = mFrameCount; 4044 } 4045 4046 if (framesReq > framesReady) { 4047 framesReq = framesReady; 4048 } 4049 4050 if (mChannelCount == 1 && mReqChannelCount == 2) { 4051 channelCount = 1; 4052 } else { 4053 channelCount = 2; 4054 } 4055 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4056 buffer->frameCount = framesReq; 4057 return NO_ERROR; 4058} 4059 4060void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4061{ 4062 mRsmpInIndex += buffer->frameCount; 4063 buffer->frameCount = 0; 4064} 4065 4066bool AudioFlinger::RecordThread::checkForNewParameters_l() 4067{ 4068 bool reconfig = false; 4069 4070 while (!mNewParameters.isEmpty()) { 4071 status_t status = NO_ERROR; 4072 String8 keyValuePair = mNewParameters[0]; 4073 AudioParameter param = AudioParameter(keyValuePair); 4074 int value; 4075 int reqFormat = mFormat; 4076 int reqSamplingRate = mReqSampleRate; 4077 int reqChannelCount = mReqChannelCount; 4078 4079 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4080 reqSamplingRate = value; 4081 reconfig = true; 4082 } 4083 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4084 reqFormat = value; 4085 reconfig = true; 4086 } 4087 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4088 reqChannelCount = AudioSystem::popCount(value); 4089 reconfig = true; 4090 } 4091 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4092 // do not accept frame count changes if tracks are open as the track buffer 4093 // size depends on frame count and correct behavior would not be garantied 4094 // if frame count is changed after track creation 4095 if (mActiveTrack != 0) { 4096 status = INVALID_OPERATION; 4097 } else { 4098 reconfig = true; 4099 } 4100 } 4101 if (status == NO_ERROR) { 4102 status = mInput->setParameters(keyValuePair); 4103 if (status == INVALID_OPERATION) { 4104 mInput->standby(); 4105 status = mInput->setParameters(keyValuePair); 4106 } 4107 if (reconfig) { 4108 if (status == BAD_VALUE && 4109 reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && 4110 ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && 4111 (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { 4112 status = NO_ERROR; 4113 } 4114 if (status == NO_ERROR) { 4115 readInputParameters(); 4116 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4117 } 4118 } 4119 } 4120 4121 mNewParameters.removeAt(0); 4122 4123 mParamStatus = status; 4124 mParamCond.signal(); 4125 mWaitWorkCV.wait(mLock); 4126 } 4127 return reconfig; 4128} 4129 4130String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4131{ 4132 return mInput->getParameters(keys); 4133} 4134 4135void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4136 AudioSystem::OutputDescriptor desc; 4137 void *param2 = 0; 4138 4139 switch (event) { 4140 case AudioSystem::INPUT_OPENED: 4141 case AudioSystem::INPUT_CONFIG_CHANGED: 4142 desc.channels = mChannels; 4143 desc.samplingRate = mSampleRate; 4144 desc.format = mFormat; 4145 desc.frameCount = mFrameCount; 4146 desc.latency = 0; 4147 param2 = &desc; 4148 break; 4149 4150 case AudioSystem::INPUT_CLOSED: 4151 default: 4152 break; 4153 } 4154 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4155} 4156 4157void AudioFlinger::RecordThread::readInputParameters() 4158{ 4159 if (mRsmpInBuffer) delete mRsmpInBuffer; 4160 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4161 if (mResampler) delete mResampler; 4162 mResampler = 0; 4163 4164 mSampleRate = mInput->sampleRate(); 4165 mChannels = mInput->channels(); 4166 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 4167 mFormat = mInput->format(); 4168 mFrameSize = (uint16_t)mInput->frameSize(); 4169 mInputBytes = mInput->bufferSize(); 4170 mFrameCount = mInputBytes / mFrameSize; 4171 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4172 4173 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4174 { 4175 int channelCount; 4176 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4177 // stereo to mono post process as the resampler always outputs stereo. 4178 if (mChannelCount == 1 && mReqChannelCount == 2) { 4179 channelCount = 1; 4180 } else { 4181 channelCount = 2; 4182 } 4183 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4184 mResampler->setSampleRate(mSampleRate); 4185 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4186 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4187 4188 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4189 if (mChannelCount == 1 && mReqChannelCount == 1) { 4190 mFrameCount >>= 1; 4191 } 4192 4193 } 4194 mRsmpInIndex = mFrameCount; 4195} 4196 4197unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4198{ 4199 return mInput->getInputFramesLost(); 4200} 4201 4202// ---------------------------------------------------------------------------- 4203 4204int AudioFlinger::openOutput(uint32_t *pDevices, 4205 uint32_t *pSamplingRate, 4206 uint32_t *pFormat, 4207 uint32_t *pChannels, 4208 uint32_t *pLatencyMs, 4209 uint32_t flags) 4210{ 4211 status_t status; 4212 PlaybackThread *thread = NULL; 4213 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4214 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4215 uint32_t format = pFormat ? *pFormat : 0; 4216 uint32_t channels = pChannels ? *pChannels : 0; 4217 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4218 4219 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4220 pDevices ? *pDevices : 0, 4221 samplingRate, 4222 format, 4223 channels, 4224 flags); 4225 4226 if (pDevices == NULL || *pDevices == 0) { 4227 return 0; 4228 } 4229 Mutex::Autolock _l(mLock); 4230 4231 AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, 4232 (int *)&format, 4233 &channels, 4234 &samplingRate, 4235 &status); 4236 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4237 output, 4238 samplingRate, 4239 format, 4240 channels, 4241 status); 4242 4243 mHardwareStatus = AUDIO_HW_IDLE; 4244 if (output != 0) { 4245 int id = nextUniqueId(); 4246 if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || 4247 (format != AudioSystem::PCM_16_BIT) || 4248 (channels != AudioSystem::CHANNEL_OUT_STEREO)) { 4249 thread = new DirectOutputThread(this, output, id, *pDevices); 4250 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4251 } else { 4252 thread = new MixerThread(this, output, id, *pDevices); 4253 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4254 4255#ifdef LVMX 4256 unsigned bitsPerSample = 4257 (format == AudioSystem::PCM_16_BIT) ? 16 : 4258 ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); 4259 unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; 4260 int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); 4261 4262 LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); 4263 LifeVibes::setDevice(audioOutputType, *pDevices); 4264#endif 4265 4266 } 4267 mPlaybackThreads.add(id, thread); 4268 4269 if (pSamplingRate) *pSamplingRate = samplingRate; 4270 if (pFormat) *pFormat = format; 4271 if (pChannels) *pChannels = channels; 4272 if (pLatencyMs) *pLatencyMs = thread->latency(); 4273 4274 // notify client processes of the new output creation 4275 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4276 return id; 4277 } 4278 4279 return 0; 4280} 4281 4282int AudioFlinger::openDuplicateOutput(int output1, int output2) 4283{ 4284 Mutex::Autolock _l(mLock); 4285 MixerThread *thread1 = checkMixerThread_l(output1); 4286 MixerThread *thread2 = checkMixerThread_l(output2); 4287 4288 if (thread1 == NULL || thread2 == NULL) { 4289 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4290 return 0; 4291 } 4292 4293 int id = nextUniqueId(); 4294 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4295 thread->addOutputTrack(thread2); 4296 mPlaybackThreads.add(id, thread); 4297 // notify client processes of the new output creation 4298 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4299 return id; 4300} 4301 4302status_t AudioFlinger::closeOutput(int output) 4303{ 4304 // keep strong reference on the playback thread so that 4305 // it is not destroyed while exit() is executed 4306 sp <PlaybackThread> thread; 4307 { 4308 Mutex::Autolock _l(mLock); 4309 thread = checkPlaybackThread_l(output); 4310 if (thread == NULL) { 4311 return BAD_VALUE; 4312 } 4313 4314 LOGV("closeOutput() %d", output); 4315 4316 if (thread->type() == PlaybackThread::MIXER) { 4317 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4318 if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { 4319 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4320 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4321 } 4322 } 4323 } 4324 void *param2 = 0; 4325 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4326 mPlaybackThreads.removeItem(output); 4327 } 4328 thread->exit(); 4329 4330 if (thread->type() != PlaybackThread::DUPLICATING) { 4331 mAudioHardware->closeOutputStream(thread->getOutput()); 4332 } 4333 return NO_ERROR; 4334} 4335 4336status_t AudioFlinger::suspendOutput(int output) 4337{ 4338 Mutex::Autolock _l(mLock); 4339 PlaybackThread *thread = checkPlaybackThread_l(output); 4340 4341 if (thread == NULL) { 4342 return BAD_VALUE; 4343 } 4344 4345 LOGV("suspendOutput() %d", output); 4346 thread->suspend(); 4347 4348 return NO_ERROR; 4349} 4350 4351status_t AudioFlinger::restoreOutput(int output) 4352{ 4353 Mutex::Autolock _l(mLock); 4354 PlaybackThread *thread = checkPlaybackThread_l(output); 4355 4356 if (thread == NULL) { 4357 return BAD_VALUE; 4358 } 4359 4360 LOGV("restoreOutput() %d", output); 4361 4362 thread->restore(); 4363 4364 return NO_ERROR; 4365} 4366 4367int AudioFlinger::openInput(uint32_t *pDevices, 4368 uint32_t *pSamplingRate, 4369 uint32_t *pFormat, 4370 uint32_t *pChannels, 4371 uint32_t acoustics) 4372{ 4373 status_t status; 4374 RecordThread *thread = NULL; 4375 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4376 uint32_t format = pFormat ? *pFormat : 0; 4377 uint32_t channels = pChannels ? *pChannels : 0; 4378 uint32_t reqSamplingRate = samplingRate; 4379 uint32_t reqFormat = format; 4380 uint32_t reqChannels = channels; 4381 4382 if (pDevices == NULL || *pDevices == 0) { 4383 return 0; 4384 } 4385 Mutex::Autolock _l(mLock); 4386 4387 AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, 4388 (int *)&format, 4389 &channels, 4390 &samplingRate, 4391 &status, 4392 (AudioSystem::audio_in_acoustics)acoustics); 4393 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4394 input, 4395 samplingRate, 4396 format, 4397 channels, 4398 acoustics, 4399 status); 4400 4401 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4402 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4403 // or stereo to mono conversions on 16 bit PCM inputs. 4404 if (input == 0 && status == BAD_VALUE && 4405 reqFormat == format && format == AudioSystem::PCM_16_BIT && 4406 (samplingRate <= 2 * reqSamplingRate) && 4407 (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) { 4408 LOGV("openInput() reopening with proposed sampling rate and channels"); 4409 input = mAudioHardware->openInputStream(*pDevices, 4410 (int *)&format, 4411 &channels, 4412 &samplingRate, 4413 &status, 4414 (AudioSystem::audio_in_acoustics)acoustics); 4415 } 4416 4417 if (input != 0) { 4418 int id = nextUniqueId(); 4419 // Start record thread 4420 thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); 4421 mRecordThreads.add(id, thread); 4422 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4423 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4424 if (pFormat) *pFormat = format; 4425 if (pChannels) *pChannels = reqChannels; 4426 4427 input->standby(); 4428 4429 // notify client processes of the new input creation 4430 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4431 return id; 4432 } 4433 4434 return 0; 4435} 4436 4437status_t AudioFlinger::closeInput(int input) 4438{ 4439 // keep strong reference on the record thread so that 4440 // it is not destroyed while exit() is executed 4441 sp <RecordThread> thread; 4442 { 4443 Mutex::Autolock _l(mLock); 4444 thread = checkRecordThread_l(input); 4445 if (thread == NULL) { 4446 return BAD_VALUE; 4447 } 4448 4449 LOGV("closeInput() %d", input); 4450 void *param2 = 0; 4451 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4452 mRecordThreads.removeItem(input); 4453 } 4454 thread->exit(); 4455 4456 mAudioHardware->closeInputStream(thread->getInput()); 4457 4458 return NO_ERROR; 4459} 4460 4461status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4462{ 4463 Mutex::Autolock _l(mLock); 4464 MixerThread *dstThread = checkMixerThread_l(output); 4465 if (dstThread == NULL) { 4466 LOGW("setStreamOutput() bad output id %d", output); 4467 return BAD_VALUE; 4468 } 4469 4470 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4471 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4472 4473 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4474 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4475 if (thread != dstThread && 4476 thread->type() != PlaybackThread::DIRECT) { 4477 MixerThread *srcThread = (MixerThread *)thread; 4478 srcThread->invalidateTracks(stream); 4479 } 4480 } 4481 4482 return NO_ERROR; 4483} 4484 4485 4486int AudioFlinger::newAudioSessionId() 4487{ 4488 return nextUniqueId(); 4489} 4490 4491// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4492AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4493{ 4494 PlaybackThread *thread = NULL; 4495 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4496 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4497 } 4498 return thread; 4499} 4500 4501// checkMixerThread_l() must be called with AudioFlinger::mLock held 4502AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4503{ 4504 PlaybackThread *thread = checkPlaybackThread_l(output); 4505 if (thread != NULL) { 4506 if (thread->type() == PlaybackThread::DIRECT) { 4507 thread = NULL; 4508 } 4509 } 4510 return (MixerThread *)thread; 4511} 4512 4513// checkRecordThread_l() must be called with AudioFlinger::mLock held 4514AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4515{ 4516 RecordThread *thread = NULL; 4517 if (mRecordThreads.indexOfKey(input) >= 0) { 4518 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4519 } 4520 return thread; 4521} 4522 4523int AudioFlinger::nextUniqueId() 4524{ 4525 return android_atomic_inc(&mNextUniqueId); 4526} 4527 4528// ---------------------------------------------------------------------------- 4529// Effect management 4530// ---------------------------------------------------------------------------- 4531 4532 4533status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) 4534{ 4535 // check calling permissions 4536 if (!settingsAllowed()) { 4537 return PERMISSION_DENIED; 4538 } 4539 // only allow libraries loaded from /system/lib/soundfx for now 4540 if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) { 4541 return PERMISSION_DENIED; 4542 } 4543 4544 Mutex::Autolock _l(mLock); 4545 return EffectLoadLibrary(libPath, handle); 4546} 4547 4548status_t AudioFlinger::unloadEffectLibrary(int handle) 4549{ 4550 // check calling permissions 4551 if (!settingsAllowed()) { 4552 return PERMISSION_DENIED; 4553 } 4554 4555 Mutex::Autolock _l(mLock); 4556 return EffectUnloadLibrary(handle); 4557} 4558 4559status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4560{ 4561 Mutex::Autolock _l(mLock); 4562 return EffectQueryNumberEffects(numEffects); 4563} 4564 4565status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4566{ 4567 Mutex::Autolock _l(mLock); 4568 return EffectQueryEffect(index, descriptor); 4569} 4570 4571status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4572{ 4573 Mutex::Autolock _l(mLock); 4574 return EffectGetDescriptor(pUuid, descriptor); 4575} 4576 4577 4578// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4579static const effect_uuid_t VISUALIZATION_UUID_ = 4580 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4581 4582sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4583 effect_descriptor_t *pDesc, 4584 const sp<IEffectClient>& effectClient, 4585 int32_t priority, 4586 int output, 4587 int sessionId, 4588 status_t *status, 4589 int *id, 4590 int *enabled) 4591{ 4592 status_t lStatus = NO_ERROR; 4593 sp<EffectHandle> handle; 4594 effect_interface_t itfe; 4595 effect_descriptor_t desc; 4596 sp<Client> client; 4597 wp<Client> wclient; 4598 4599 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", 4600 pid, effectClient.get(), priority, sessionId, output); 4601 4602 if (pDesc == NULL) { 4603 lStatus = BAD_VALUE; 4604 goto Exit; 4605 } 4606 4607 { 4608 Mutex::Autolock _l(mLock); 4609 4610 // check recording permission for visualizer 4611 if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4612 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) { 4613 if (!recordingAllowed()) { 4614 lStatus = PERMISSION_DENIED; 4615 goto Exit; 4616 } 4617 } 4618 4619 if (!EffectIsNullUuid(&pDesc->uuid)) { 4620 // if uuid is specified, request effect descriptor 4621 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4622 if (lStatus < 0) { 4623 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4624 goto Exit; 4625 } 4626 } else { 4627 // if uuid is not specified, look for an available implementation 4628 // of the required type in effect factory 4629 if (EffectIsNullUuid(&pDesc->type)) { 4630 LOGW("createEffect() no effect type"); 4631 lStatus = BAD_VALUE; 4632 goto Exit; 4633 } 4634 uint32_t numEffects = 0; 4635 effect_descriptor_t d; 4636 bool found = false; 4637 4638 lStatus = EffectQueryNumberEffects(&numEffects); 4639 if (lStatus < 0) { 4640 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4641 goto Exit; 4642 } 4643 for (uint32_t i = 0; i < numEffects; i++) { 4644 lStatus = EffectQueryEffect(i, &desc); 4645 if (lStatus < 0) { 4646 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4647 continue; 4648 } 4649 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4650 // If matching type found save effect descriptor. If the session is 4651 // 0 and the effect is not auxiliary, continue enumeration in case 4652 // an auxiliary version of this effect type is available 4653 found = true; 4654 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4655 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX || 4656 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4657 break; 4658 } 4659 } 4660 } 4661 if (!found) { 4662 lStatus = BAD_VALUE; 4663 LOGW("createEffect() effect not found"); 4664 goto Exit; 4665 } 4666 // For same effect type, chose auxiliary version over insert version if 4667 // connect to output mix (Compliance to OpenSL ES) 4668 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && 4669 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4670 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4671 } 4672 } 4673 4674 // Do not allow auxiliary effects on a session different from 0 (output mix) 4675 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX && 4676 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4677 lStatus = INVALID_OPERATION; 4678 goto Exit; 4679 } 4680 4681 // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects 4682 // that can only be created by audio policy manager (running in same process) 4683 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && 4684 getpid() != IPCThreadState::self()->getCallingPid()) { 4685 lStatus = INVALID_OPERATION; 4686 goto Exit; 4687 } 4688 4689 // return effect descriptor 4690 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4691 4692 // If output is not specified try to find a matching audio session ID in one of the 4693 // output threads. 4694 // TODO: allow attachment of effect to inputs 4695 if (output == 0) { 4696 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) { 4697 // output must be specified by AudioPolicyManager when using session 4698 // AudioSystem::SESSION_OUTPUT_STAGE 4699 lStatus = BAD_VALUE; 4700 goto Exit; 4701 } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 4702 output = AudioSystem::getOutputForEffect(&desc); 4703 LOGV("createEffect() got output %d for effect %s", output, desc.name); 4704 } else { 4705 // look for the thread where the specified audio session is present 4706 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4707 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId)) { 4708 output = mPlaybackThreads.keyAt(i); 4709 break; 4710 } 4711 } 4712 } 4713 } 4714 PlaybackThread *thread = checkPlaybackThread_l(output); 4715 if (thread == NULL) { 4716 LOGE("createEffect() unknown output thread"); 4717 lStatus = BAD_VALUE; 4718 goto Exit; 4719 } 4720 4721 wclient = mClients.valueFor(pid); 4722 4723 if (wclient != NULL) { 4724 client = wclient.promote(); 4725 } else { 4726 client = new Client(this, pid); 4727 mClients.add(pid, client); 4728 } 4729 4730 // create effect on selected output trhead 4731 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4732 &desc, enabled, &lStatus); 4733 if (handle != 0 && id != NULL) { 4734 *id = handle->id(); 4735 } 4736 } 4737 4738Exit: 4739 if(status) { 4740 *status = lStatus; 4741 } 4742 return handle; 4743} 4744 4745status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4746{ 4747 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4748 session, srcOutput, dstOutput); 4749 Mutex::Autolock _l(mLock); 4750 if (srcOutput == dstOutput) { 4751 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4752 return NO_ERROR; 4753 } 4754 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4755 if (srcThread == NULL) { 4756 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4757 return BAD_VALUE; 4758 } 4759 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 4760 if (dstThread == NULL) { 4761 LOGW("moveEffects() bad dstOutput %d", dstOutput); 4762 return BAD_VALUE; 4763 } 4764 4765 Mutex::Autolock _dl(dstThread->mLock); 4766 Mutex::Autolock _sl(srcThread->mLock); 4767 moveEffectChain_l(session, srcThread, dstThread); 4768 4769 return NO_ERROR; 4770} 4771 4772// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 4773status_t AudioFlinger::moveEffectChain_l(int session, 4774 AudioFlinger::PlaybackThread *srcThread, 4775 AudioFlinger::PlaybackThread *dstThread) 4776{ 4777 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 4778 session, srcThread, dstThread); 4779 4780 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 4781 if (chain == 0) { 4782 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 4783 session, srcThread); 4784 return INVALID_OPERATION; 4785 } 4786 4787 // remove chain first. This is usefull only if reconfiguring effect chain on same output thread, 4788 // so that a new chain is created with correct parameters when first effect is added. This is 4789 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 4790 // removed. 4791 srcThread->removeEffectChain_l(chain); 4792 4793 // transfer all effects one by one so that new effect chain is created on new thread with 4794 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 4795 sp<EffectModule> effect = chain->getEffectFromId_l(0); 4796 while (effect != 0) { 4797 srcThread->removeEffect_l(effect); 4798 dstThread->addEffect_l(effect); 4799 effect = chain->getEffectFromId_l(0); 4800 } 4801 4802 return NO_ERROR; 4803} 4804 4805// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 4806sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( 4807 const sp<AudioFlinger::Client>& client, 4808 const sp<IEffectClient>& effectClient, 4809 int32_t priority, 4810 int sessionId, 4811 effect_descriptor_t *desc, 4812 int *enabled, 4813 status_t *status 4814 ) 4815{ 4816 sp<EffectModule> effect; 4817 sp<EffectHandle> handle; 4818 status_t lStatus; 4819 sp<Track> track; 4820 sp<EffectChain> chain; 4821 bool chainCreated = false; 4822 bool effectCreated = false; 4823 bool effectRegistered = false; 4824 4825 if (mOutput == 0) { 4826 LOGW("createEffect_l() Audio driver not initialized."); 4827 lStatus = NO_INIT; 4828 goto Exit; 4829 } 4830 4831 // Do not allow auxiliary effect on session other than 0 4832 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && 4833 sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 4834 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4835 desc->name, sessionId); 4836 lStatus = BAD_VALUE; 4837 goto Exit; 4838 } 4839 4840 // Do not allow effects with session ID 0 on direct output or duplicating threads 4841 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 4842 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) { 4843 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4844 desc->name, sessionId); 4845 lStatus = BAD_VALUE; 4846 goto Exit; 4847 } 4848 4849 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 4850 4851 { // scope for mLock 4852 Mutex::Autolock _l(mLock); 4853 4854 // check for existing effect chain with the requested audio session 4855 chain = getEffectChain_l(sessionId); 4856 if (chain == 0) { 4857 // create a new chain for this session 4858 LOGV("createEffect_l() new effect chain for session %d", sessionId); 4859 chain = new EffectChain(this, sessionId); 4860 addEffectChain_l(chain); 4861 chain->setStrategy(getStrategyForSession_l(sessionId)); 4862 chainCreated = true; 4863 } else { 4864 effect = chain->getEffectFromDesc_l(desc); 4865 } 4866 4867 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 4868 4869 if (effect == 0) { 4870 int id = mAudioFlinger->nextUniqueId(); 4871 // Check CPU and memory usage 4872 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 4873 if (lStatus != NO_ERROR) { 4874 goto Exit; 4875 } 4876 effectRegistered = true; 4877 // create a new effect module if none present in the chain 4878 effect = new EffectModule(this, chain, desc, id, sessionId); 4879 lStatus = effect->status(); 4880 if (lStatus != NO_ERROR) { 4881 goto Exit; 4882 } 4883 lStatus = chain->addEffect_l(effect); 4884 if (lStatus != NO_ERROR) { 4885 goto Exit; 4886 } 4887 effectCreated = true; 4888 4889 effect->setDevice(mDevice); 4890 effect->setMode(mAudioFlinger->getMode()); 4891 } 4892 // create effect handle and connect it to effect module 4893 handle = new EffectHandle(effect, client, effectClient, priority); 4894 lStatus = effect->addHandle(handle); 4895 if (enabled) { 4896 *enabled = (int)effect->isEnabled(); 4897 } 4898 } 4899 4900Exit: 4901 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 4902 Mutex::Autolock _l(mLock); 4903 if (effectCreated) { 4904 chain->removeEffect_l(effect); 4905 } 4906 if (effectRegistered) { 4907 AudioSystem::unregisterEffect(effect->id()); 4908 } 4909 if (chainCreated) { 4910 removeEffectChain_l(chain); 4911 } 4912 handle.clear(); 4913 } 4914 4915 if(status) { 4916 *status = lStatus; 4917 } 4918 return handle; 4919} 4920 4921// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 4922// PlaybackThread::mLock held 4923status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect) 4924{ 4925 // check for existing effect chain with the requested audio session 4926 int sessionId = effect->sessionId(); 4927 sp<EffectChain> chain = getEffectChain_l(sessionId); 4928 bool chainCreated = false; 4929 4930 if (chain == 0) { 4931 // create a new chain for this session 4932 LOGV("addEffect_l() new effect chain for session %d", sessionId); 4933 chain = new EffectChain(this, sessionId); 4934 addEffectChain_l(chain); 4935 chain->setStrategy(getStrategyForSession_l(sessionId)); 4936 chainCreated = true; 4937 } 4938 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 4939 4940 if (chain->getEffectFromId_l(effect->id()) != 0) { 4941 LOGW("addEffect_l() %p effect %s already present in chain %p", 4942 this, effect->desc().name, chain.get()); 4943 return BAD_VALUE; 4944 } 4945 4946 status_t status = chain->addEffect_l(effect); 4947 if (status != NO_ERROR) { 4948 if (chainCreated) { 4949 removeEffectChain_l(chain); 4950 } 4951 return status; 4952 } 4953 4954 effect->setDevice(mDevice); 4955 effect->setMode(mAudioFlinger->getMode()); 4956 return NO_ERROR; 4957} 4958 4959void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) { 4960 4961 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 4962 effect_descriptor_t desc = effect->desc(); 4963 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4964 detachAuxEffect_l(effect->id()); 4965 } 4966 4967 sp<EffectChain> chain = effect->chain().promote(); 4968 if (chain != 0) { 4969 // remove effect chain if removing last effect 4970 if (chain->removeEffect_l(effect) == 0) { 4971 removeEffectChain_l(chain); 4972 } 4973 } else { 4974 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 4975 } 4976} 4977 4978void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect, 4979 const wp<EffectHandle>& handle) { 4980 Mutex::Autolock _l(mLock); 4981 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 4982 // delete the effect module if removing last handle on it 4983 if (effect->removeHandle(handle) == 0) { 4984 removeEffect_l(effect); 4985 AudioSystem::unregisterEffect(effect->id()); 4986 } 4987} 4988 4989status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 4990{ 4991 int session = chain->sessionId(); 4992 int16_t *buffer = mMixBuffer; 4993 bool ownsBuffer = false; 4994 4995 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 4996 if (session > 0) { 4997 // Only one effect chain can be present in direct output thread and it uses 4998 // the mix buffer as input 4999 if (mType != DIRECT) { 5000 size_t numSamples = mFrameCount * mChannelCount; 5001 buffer = new int16_t[numSamples]; 5002 memset(buffer, 0, numSamples * sizeof(int16_t)); 5003 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5004 ownsBuffer = true; 5005 } 5006 5007 // Attach all tracks with same session ID to this chain. 5008 for (size_t i = 0; i < mTracks.size(); ++i) { 5009 sp<Track> track = mTracks[i]; 5010 if (session == track->sessionId()) { 5011 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5012 track->setMainBuffer(buffer); 5013 } 5014 } 5015 5016 // indicate all active tracks in the chain 5017 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5018 sp<Track> track = mActiveTracks[i].promote(); 5019 if (track == 0) continue; 5020 if (session == track->sessionId()) { 5021 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5022 chain->startTrack(); 5023 } 5024 } 5025 } 5026 5027 chain->setInBuffer(buffer, ownsBuffer); 5028 chain->setOutBuffer(mMixBuffer); 5029 // Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect 5030 // chains list in order to be processed last as it contains output stage effects 5031 // Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before 5032 // session AudioSystem::SESSION_OUTPUT_STAGE to be processed 5033 // after track specific effects and before output stage 5034 // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and 5035 // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX 5036 // Effect chain for other sessions are inserted at beginning of effect 5037 // chains list to be processed before output mix effects. Relative order between other 5038 // sessions is not important 5039 size_t size = mEffectChains.size(); 5040 size_t i = 0; 5041 for (i = 0; i < size; i++) { 5042 if (mEffectChains[i]->sessionId() < session) break; 5043 } 5044 mEffectChains.insertAt(chain, i); 5045 5046 return NO_ERROR; 5047} 5048 5049size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5050{ 5051 int session = chain->sessionId(); 5052 5053 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5054 5055 for (size_t i = 0; i < mEffectChains.size(); i++) { 5056 if (chain == mEffectChains[i]) { 5057 mEffectChains.removeAt(i); 5058 // detach all tracks with same session ID from this chain 5059 for (size_t i = 0; i < mTracks.size(); ++i) { 5060 sp<Track> track = mTracks[i]; 5061 if (session == track->sessionId()) { 5062 track->setMainBuffer(mMixBuffer); 5063 } 5064 } 5065 break; 5066 } 5067 } 5068 return mEffectChains.size(); 5069} 5070 5071void AudioFlinger::PlaybackThread::lockEffectChains_l( 5072 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5073{ 5074 effectChains = mEffectChains; 5075 for (size_t i = 0; i < mEffectChains.size(); i++) { 5076 mEffectChains[i]->lock(); 5077 } 5078} 5079 5080void AudioFlinger::PlaybackThread::unlockEffectChains( 5081 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5082{ 5083 for (size_t i = 0; i < effectChains.size(); i++) { 5084 effectChains[i]->unlock(); 5085 } 5086} 5087 5088 5089sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) 5090{ 5091 sp<EffectModule> effect; 5092 5093 sp<EffectChain> chain = getEffectChain_l(sessionId); 5094 if (chain != 0) { 5095 effect = chain->getEffectFromId_l(effectId); 5096 } 5097 return effect; 5098} 5099 5100status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5101 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5102{ 5103 Mutex::Autolock _l(mLock); 5104 return attachAuxEffect_l(track, EffectId); 5105} 5106 5107status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5108 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5109{ 5110 status_t status = NO_ERROR; 5111 5112 if (EffectId == 0) { 5113 track->setAuxBuffer(0, NULL); 5114 } else { 5115 // Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX 5116 sp<EffectModule> effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId); 5117 if (effect != 0) { 5118 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5119 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5120 } else { 5121 status = INVALID_OPERATION; 5122 } 5123 } else { 5124 status = BAD_VALUE; 5125 } 5126 } 5127 return status; 5128} 5129 5130void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5131{ 5132 for (size_t i = 0; i < mTracks.size(); ++i) { 5133 sp<Track> track = mTracks[i]; 5134 if (track->auxEffectId() == effectId) { 5135 attachAuxEffect_l(track, 0); 5136 } 5137 } 5138} 5139 5140// ---------------------------------------------------------------------------- 5141// EffectModule implementation 5142// ---------------------------------------------------------------------------- 5143 5144#undef LOG_TAG 5145#define LOG_TAG "AudioFlinger::EffectModule" 5146 5147AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5148 const wp<AudioFlinger::EffectChain>& chain, 5149 effect_descriptor_t *desc, 5150 int id, 5151 int sessionId) 5152 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5153 mStatus(NO_INIT), mState(IDLE) 5154{ 5155 LOGV("Constructor %p", this); 5156 int lStatus; 5157 sp<ThreadBase> thread = mThread.promote(); 5158 if (thread == 0) { 5159 return; 5160 } 5161 PlaybackThread *p = (PlaybackThread *)thread.get(); 5162 5163 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5164 5165 // create effect engine from effect factory 5166 mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); 5167 5168 if (mStatus != NO_ERROR) { 5169 return; 5170 } 5171 lStatus = init(); 5172 if (lStatus < 0) { 5173 mStatus = lStatus; 5174 goto Error; 5175 } 5176 5177 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5178 return; 5179Error: 5180 EffectRelease(mEffectInterface); 5181 mEffectInterface = NULL; 5182 LOGV("Constructor Error %d", mStatus); 5183} 5184 5185AudioFlinger::EffectModule::~EffectModule() 5186{ 5187 LOGV("Destructor %p", this); 5188 if (mEffectInterface != NULL) { 5189 // release effect engine 5190 EffectRelease(mEffectInterface); 5191 } 5192} 5193 5194status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5195{ 5196 status_t status; 5197 5198 Mutex::Autolock _l(mLock); 5199 // First handle in mHandles has highest priority and controls the effect module 5200 int priority = handle->priority(); 5201 size_t size = mHandles.size(); 5202 sp<EffectHandle> h; 5203 size_t i; 5204 for (i = 0; i < size; i++) { 5205 h = mHandles[i].promote(); 5206 if (h == 0) continue; 5207 if (h->priority() <= priority) break; 5208 } 5209 // if inserted in first place, move effect control from previous owner to this handle 5210 if (i == 0) { 5211 if (h != 0) { 5212 h->setControl(false, true); 5213 } 5214 handle->setControl(true, false); 5215 status = NO_ERROR; 5216 } else { 5217 status = ALREADY_EXISTS; 5218 } 5219 mHandles.insertAt(handle, i); 5220 return status; 5221} 5222 5223size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5224{ 5225 Mutex::Autolock _l(mLock); 5226 size_t size = mHandles.size(); 5227 size_t i; 5228 for (i = 0; i < size; i++) { 5229 if (mHandles[i] == handle) break; 5230 } 5231 if (i == size) { 5232 return size; 5233 } 5234 mHandles.removeAt(i); 5235 size = mHandles.size(); 5236 // if removed from first place, move effect control from this handle to next in line 5237 if (i == 0 && size != 0) { 5238 sp<EffectHandle> h = mHandles[0].promote(); 5239 if (h != 0) { 5240 h->setControl(true, true); 5241 } 5242 } 5243 5244 return size; 5245} 5246 5247void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5248{ 5249 // keep a strong reference on this EffectModule to avoid calling the 5250 // destructor before we exit 5251 sp<EffectModule> keep(this); 5252 { 5253 sp<ThreadBase> thread = mThread.promote(); 5254 if (thread != 0) { 5255 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5256 playbackThread->disconnectEffect(keep, handle); 5257 } 5258 } 5259} 5260 5261void AudioFlinger::EffectModule::updateState() { 5262 Mutex::Autolock _l(mLock); 5263 5264 switch (mState) { 5265 case RESTART: 5266 reset_l(); 5267 // FALL THROUGH 5268 5269 case STARTING: 5270 // clear auxiliary effect input buffer for next accumulation 5271 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5272 memset(mConfig.inputCfg.buffer.raw, 5273 0, 5274 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5275 } 5276 start_l(); 5277 mState = ACTIVE; 5278 break; 5279 case STOPPING: 5280 stop_l(); 5281 mDisableWaitCnt = mMaxDisableWaitCnt; 5282 mState = STOPPED; 5283 break; 5284 case STOPPED: 5285 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5286 // turn off sequence. 5287 if (--mDisableWaitCnt == 0) { 5288 reset_l(); 5289 mState = IDLE; 5290 } 5291 break; 5292 default: //IDLE , ACTIVE 5293 break; 5294 } 5295} 5296 5297void AudioFlinger::EffectModule::process() 5298{ 5299 Mutex::Autolock _l(mLock); 5300 5301 if (mEffectInterface == NULL || 5302 mConfig.inputCfg.buffer.raw == NULL || 5303 mConfig.outputCfg.buffer.raw == NULL) { 5304 return; 5305 } 5306 5307 if (mState == ACTIVE || mState == STOPPING || mState == STOPPED) { 5308 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5309 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5310 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5311 mConfig.inputCfg.buffer.s32, 5312 mConfig.inputCfg.buffer.frameCount/2); 5313 } 5314 5315 // do the actual processing in the effect engine 5316 int ret = (*mEffectInterface)->process(mEffectInterface, 5317 &mConfig.inputCfg.buffer, 5318 &mConfig.outputCfg.buffer); 5319 5320 // force transition to IDLE state when engine is ready 5321 if (mState == STOPPED && ret == -ENODATA) { 5322 mDisableWaitCnt = 1; 5323 } 5324 5325 // clear auxiliary effect input buffer for next accumulation 5326 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5327 memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5328 } 5329 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5330 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){ 5331 // If an insert effect is idle and input buffer is different from output buffer, copy input to 5332 // output 5333 sp<EffectChain> chain = mChain.promote(); 5334 if (chain != 0 && chain->activeTracks() != 0) { 5335 size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t); 5336 if (mConfig.inputCfg.channels == CHANNEL_STEREO) { 5337 size *= 2; 5338 } 5339 memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size); 5340 } 5341 } 5342} 5343 5344void AudioFlinger::EffectModule::reset_l() 5345{ 5346 if (mEffectInterface == NULL) { 5347 return; 5348 } 5349 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5350} 5351 5352status_t AudioFlinger::EffectModule::configure() 5353{ 5354 uint32_t channels; 5355 if (mEffectInterface == NULL) { 5356 return NO_INIT; 5357 } 5358 5359 sp<ThreadBase> thread = mThread.promote(); 5360 if (thread == 0) { 5361 return DEAD_OBJECT; 5362 } 5363 5364 // TODO: handle configuration of effects replacing track process 5365 if (thread->channelCount() == 1) { 5366 channels = CHANNEL_MONO; 5367 } else { 5368 channels = CHANNEL_STEREO; 5369 } 5370 5371 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5372 mConfig.inputCfg.channels = CHANNEL_MONO; 5373 } else { 5374 mConfig.inputCfg.channels = channels; 5375 } 5376 mConfig.outputCfg.channels = channels; 5377 mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; 5378 mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; 5379 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5380 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5381 mConfig.inputCfg.bufferProvider.cookie = NULL; 5382 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5383 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5384 mConfig.outputCfg.bufferProvider.cookie = NULL; 5385 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5386 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5387 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5388 // Insert effect: 5389 // - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE, 5390 // always overwrites output buffer: input buffer == output buffer 5391 // - in other sessions: 5392 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5393 // other effect: overwrites output buffer: input buffer == output buffer 5394 // Auxiliary effect: 5395 // accumulates in output buffer: input buffer != output buffer 5396 // Therefore: accumulate <=> input buffer != output buffer 5397 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5398 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5399 } else { 5400 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5401 } 5402 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5403 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5404 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5405 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5406 5407 LOGV("configure() %p thread %p buffer %p framecount %d", 5408 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5409 5410 status_t cmdStatus; 5411 uint32_t size = sizeof(int); 5412 status_t status = (*mEffectInterface)->command(mEffectInterface, 5413 EFFECT_CMD_CONFIGURE, 5414 sizeof(effect_config_t), 5415 &mConfig, 5416 &size, 5417 &cmdStatus); 5418 if (status == 0) { 5419 status = cmdStatus; 5420 } 5421 5422 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5423 (1000 * mConfig.outputCfg.buffer.frameCount); 5424 5425 return status; 5426} 5427 5428status_t AudioFlinger::EffectModule::init() 5429{ 5430 Mutex::Autolock _l(mLock); 5431 if (mEffectInterface == NULL) { 5432 return NO_INIT; 5433 } 5434 status_t cmdStatus; 5435 uint32_t size = sizeof(status_t); 5436 status_t status = (*mEffectInterface)->command(mEffectInterface, 5437 EFFECT_CMD_INIT, 5438 0, 5439 NULL, 5440 &size, 5441 &cmdStatus); 5442 if (status == 0) { 5443 status = cmdStatus; 5444 } 5445 return status; 5446} 5447 5448status_t AudioFlinger::EffectModule::start_l() 5449{ 5450 if (mEffectInterface == NULL) { 5451 return NO_INIT; 5452 } 5453 status_t cmdStatus; 5454 uint32_t size = sizeof(status_t); 5455 status_t status = (*mEffectInterface)->command(mEffectInterface, 5456 EFFECT_CMD_ENABLE, 5457 0, 5458 NULL, 5459 &size, 5460 &cmdStatus); 5461 if (status == 0) { 5462 status = cmdStatus; 5463 } 5464 return status; 5465} 5466 5467status_t AudioFlinger::EffectModule::stop_l() 5468{ 5469 if (mEffectInterface == NULL) { 5470 return NO_INIT; 5471 } 5472 status_t cmdStatus; 5473 uint32_t size = sizeof(status_t); 5474 status_t status = (*mEffectInterface)->command(mEffectInterface, 5475 EFFECT_CMD_DISABLE, 5476 0, 5477 NULL, 5478 &size, 5479 &cmdStatus); 5480 if (status == 0) { 5481 status = cmdStatus; 5482 } 5483 return status; 5484} 5485 5486status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5487 uint32_t cmdSize, 5488 void *pCmdData, 5489 uint32_t *replySize, 5490 void *pReplyData) 5491{ 5492 Mutex::Autolock _l(mLock); 5493// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5494 5495 if (mEffectInterface == NULL) { 5496 return NO_INIT; 5497 } 5498 status_t status = (*mEffectInterface)->command(mEffectInterface, 5499 cmdCode, 5500 cmdSize, 5501 pCmdData, 5502 replySize, 5503 pReplyData); 5504 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5505 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5506 for (size_t i = 1; i < mHandles.size(); i++) { 5507 sp<EffectHandle> h = mHandles[i].promote(); 5508 if (h != 0) { 5509 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5510 } 5511 } 5512 } 5513 return status; 5514} 5515 5516status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5517{ 5518 Mutex::Autolock _l(mLock); 5519 LOGV("setEnabled %p enabled %d", this, enabled); 5520 5521 if (enabled != isEnabled()) { 5522 switch (mState) { 5523 // going from disabled to enabled 5524 case IDLE: 5525 mState = STARTING; 5526 break; 5527 case STOPPED: 5528 mState = RESTART; 5529 break; 5530 case STOPPING: 5531 mState = ACTIVE; 5532 break; 5533 5534 // going from enabled to disabled 5535 case RESTART: 5536 case STARTING: 5537 mState = IDLE; 5538 break; 5539 case ACTIVE: 5540 mState = STOPPING; 5541 break; 5542 } 5543 for (size_t i = 1; i < mHandles.size(); i++) { 5544 sp<EffectHandle> h = mHandles[i].promote(); 5545 if (h != 0) { 5546 h->setEnabled(enabled); 5547 } 5548 } 5549 } 5550 return NO_ERROR; 5551} 5552 5553bool AudioFlinger::EffectModule::isEnabled() 5554{ 5555 switch (mState) { 5556 case RESTART: 5557 case STARTING: 5558 case ACTIVE: 5559 return true; 5560 case IDLE: 5561 case STOPPING: 5562 case STOPPED: 5563 default: 5564 return false; 5565 } 5566} 5567 5568status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5569{ 5570 Mutex::Autolock _l(mLock); 5571 status_t status = NO_ERROR; 5572 5573 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5574 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5575 if ((mState >= ACTIVE) && 5576 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5577 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5578 status_t cmdStatus; 5579 uint32_t volume[2]; 5580 uint32_t *pVolume = NULL; 5581 uint32_t size = sizeof(volume); 5582 volume[0] = *left; 5583 volume[1] = *right; 5584 if (controller) { 5585 pVolume = volume; 5586 } 5587 status = (*mEffectInterface)->command(mEffectInterface, 5588 EFFECT_CMD_SET_VOLUME, 5589 size, 5590 volume, 5591 &size, 5592 pVolume); 5593 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5594 *left = volume[0]; 5595 *right = volume[1]; 5596 } 5597 } 5598 return status; 5599} 5600 5601status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5602{ 5603 Mutex::Autolock _l(mLock); 5604 status_t status = NO_ERROR; 5605 if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5606 // convert device bit field from AudioSystem to EffectApi format. 5607 device = deviceAudioSystemToEffectApi(device); 5608 if (device == 0) { 5609 return BAD_VALUE; 5610 } 5611 status_t cmdStatus; 5612 uint32_t size = sizeof(status_t); 5613 status = (*mEffectInterface)->command(mEffectInterface, 5614 EFFECT_CMD_SET_DEVICE, 5615 sizeof(uint32_t), 5616 &device, 5617 &size, 5618 &cmdStatus); 5619 if (status == NO_ERROR) { 5620 status = cmdStatus; 5621 } 5622 } 5623 return status; 5624} 5625 5626status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 5627{ 5628 Mutex::Autolock _l(mLock); 5629 status_t status = NO_ERROR; 5630 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 5631 // convert audio mode from AudioSystem to EffectApi format. 5632 int effectMode = modeAudioSystemToEffectApi(mode); 5633 if (effectMode < 0) { 5634 return BAD_VALUE; 5635 } 5636 status_t cmdStatus; 5637 uint32_t size = sizeof(status_t); 5638 status = (*mEffectInterface)->command(mEffectInterface, 5639 EFFECT_CMD_SET_AUDIO_MODE, 5640 sizeof(int), 5641 &effectMode, 5642 &size, 5643 &cmdStatus); 5644 if (status == NO_ERROR) { 5645 status = cmdStatus; 5646 } 5647 } 5648 return status; 5649} 5650 5651// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified 5652const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { 5653 DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE 5654 DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER 5655 DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET 5656 DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE 5657 DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO 5658 DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET 5659 DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT 5660 DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP 5661 DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES 5662 DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER 5663 DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL 5664}; 5665 5666uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) 5667{ 5668 uint32_t deviceOut = 0; 5669 while (device) { 5670 const uint32_t i = 31 - __builtin_clz(device); 5671 device &= ~(1 << i); 5672 if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { 5673 LOGE("device convertion error for AudioSystem device 0x%08x", device); 5674 return 0; 5675 } 5676 deviceOut |= (uint32_t)sDeviceConvTable[i]; 5677 } 5678 return deviceOut; 5679} 5680 5681// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified 5682const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { 5683 AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL 5684 AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE 5685 AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL 5686}; 5687 5688int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) 5689{ 5690 int modeOut = -1; 5691 if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { 5692 modeOut = (int)sModeConvTable[mode]; 5693 } 5694 return modeOut; 5695} 5696 5697status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 5698{ 5699 const size_t SIZE = 256; 5700 char buffer[SIZE]; 5701 String8 result; 5702 5703 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 5704 result.append(buffer); 5705 5706 bool locked = tryLock(mLock); 5707 // failed to lock - AudioFlinger is probably deadlocked 5708 if (!locked) { 5709 result.append("\t\tCould not lock Fx mutex:\n"); 5710 } 5711 5712 result.append("\t\tSession Status State Engine:\n"); 5713 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 5714 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 5715 result.append(buffer); 5716 5717 result.append("\t\tDescriptor:\n"); 5718 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5719 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 5720 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 5721 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 5722 result.append(buffer); 5723 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5724 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 5725 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 5726 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 5727 result.append(buffer); 5728 snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", 5729 mDescriptor.apiVersion, 5730 mDescriptor.flags); 5731 result.append(buffer); 5732 snprintf(buffer, SIZE, "\t\t- name: %s\n", 5733 mDescriptor.name); 5734 result.append(buffer); 5735 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 5736 mDescriptor.implementor); 5737 result.append(buffer); 5738 5739 result.append("\t\t- Input configuration:\n"); 5740 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5741 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5742 (uint32_t)mConfig.inputCfg.buffer.raw, 5743 mConfig.inputCfg.buffer.frameCount, 5744 mConfig.inputCfg.samplingRate, 5745 mConfig.inputCfg.channels, 5746 mConfig.inputCfg.format); 5747 result.append(buffer); 5748 5749 result.append("\t\t- Output configuration:\n"); 5750 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5751 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5752 (uint32_t)mConfig.outputCfg.buffer.raw, 5753 mConfig.outputCfg.buffer.frameCount, 5754 mConfig.outputCfg.samplingRate, 5755 mConfig.outputCfg.channels, 5756 mConfig.outputCfg.format); 5757 result.append(buffer); 5758 5759 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 5760 result.append(buffer); 5761 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 5762 for (size_t i = 0; i < mHandles.size(); ++i) { 5763 sp<EffectHandle> handle = mHandles[i].promote(); 5764 if (handle != 0) { 5765 handle->dump(buffer, SIZE); 5766 result.append(buffer); 5767 } 5768 } 5769 5770 result.append("\n"); 5771 5772 write(fd, result.string(), result.length()); 5773 5774 if (locked) { 5775 mLock.unlock(); 5776 } 5777 5778 return NO_ERROR; 5779} 5780 5781// ---------------------------------------------------------------------------- 5782// EffectHandle implementation 5783// ---------------------------------------------------------------------------- 5784 5785#undef LOG_TAG 5786#define LOG_TAG "AudioFlinger::EffectHandle" 5787 5788AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 5789 const sp<AudioFlinger::Client>& client, 5790 const sp<IEffectClient>& effectClient, 5791 int32_t priority) 5792 : BnEffect(), 5793 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 5794{ 5795 LOGV("constructor %p", this); 5796 5797 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 5798 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 5799 if (mCblkMemory != 0) { 5800 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 5801 5802 if (mCblk) { 5803 new(mCblk) effect_param_cblk_t(); 5804 mBuffer = (uint8_t *)mCblk + bufOffset; 5805 } 5806 } else { 5807 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 5808 return; 5809 } 5810} 5811 5812AudioFlinger::EffectHandle::~EffectHandle() 5813{ 5814 LOGV("Destructor %p", this); 5815 disconnect(); 5816} 5817 5818status_t AudioFlinger::EffectHandle::enable() 5819{ 5820 if (!mHasControl) return INVALID_OPERATION; 5821 if (mEffect == 0) return DEAD_OBJECT; 5822 5823 return mEffect->setEnabled(true); 5824} 5825 5826status_t AudioFlinger::EffectHandle::disable() 5827{ 5828 if (!mHasControl) return INVALID_OPERATION; 5829 if (mEffect == NULL) return DEAD_OBJECT; 5830 5831 return mEffect->setEnabled(false); 5832} 5833 5834void AudioFlinger::EffectHandle::disconnect() 5835{ 5836 if (mEffect == 0) { 5837 return; 5838 } 5839 mEffect->disconnect(this); 5840 // release sp on module => module destructor can be called now 5841 mEffect.clear(); 5842 if (mCblk) { 5843 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 5844 } 5845 mCblkMemory.clear(); // and free the shared memory 5846 if (mClient != 0) { 5847 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 5848 mClient.clear(); 5849 } 5850} 5851 5852status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 5853 uint32_t cmdSize, 5854 void *pCmdData, 5855 uint32_t *replySize, 5856 void *pReplyData) 5857{ 5858// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 5859// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 5860 5861 // only get parameter command is permitted for applications not controlling the effect 5862 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 5863 return INVALID_OPERATION; 5864 } 5865 if (mEffect == 0) return DEAD_OBJECT; 5866 5867 // handle commands that are not forwarded transparently to effect engine 5868 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 5869 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 5870 // no risk to block the whole media server process or mixer threads is we are stuck here 5871 Mutex::Autolock _l(mCblk->lock); 5872 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 5873 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 5874 mCblk->serverIndex = 0; 5875 mCblk->clientIndex = 0; 5876 return BAD_VALUE; 5877 } 5878 status_t status = NO_ERROR; 5879 while (mCblk->serverIndex < mCblk->clientIndex) { 5880 int reply; 5881 uint32_t rsize = sizeof(int); 5882 int *p = (int *)(mBuffer + mCblk->serverIndex); 5883 int size = *p++; 5884 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 5885 LOGW("command(): invalid parameter block size"); 5886 break; 5887 } 5888 effect_param_t *param = (effect_param_t *)p; 5889 if (param->psize == 0 || param->vsize == 0) { 5890 LOGW("command(): null parameter or value size"); 5891 mCblk->serverIndex += size; 5892 continue; 5893 } 5894 uint32_t psize = sizeof(effect_param_t) + 5895 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 5896 param->vsize; 5897 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 5898 psize, 5899 p, 5900 &rsize, 5901 &reply); 5902 if (ret == NO_ERROR) { 5903 if (reply != NO_ERROR) { 5904 status = reply; 5905 } 5906 } else { 5907 status = ret; 5908 } 5909 mCblk->serverIndex += size; 5910 } 5911 mCblk->serverIndex = 0; 5912 mCblk->clientIndex = 0; 5913 return status; 5914 } else if (cmdCode == EFFECT_CMD_ENABLE) { 5915 return enable(); 5916 } else if (cmdCode == EFFECT_CMD_DISABLE) { 5917 return disable(); 5918 } 5919 5920 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 5921} 5922 5923sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 5924 return mCblkMemory; 5925} 5926 5927void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 5928{ 5929 LOGV("setControl %p control %d", this, hasControl); 5930 5931 mHasControl = hasControl; 5932 if (signal && mEffectClient != 0) { 5933 mEffectClient->controlStatusChanged(hasControl); 5934 } 5935} 5936 5937void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 5938 uint32_t cmdSize, 5939 void *pCmdData, 5940 uint32_t replySize, 5941 void *pReplyData) 5942{ 5943 if (mEffectClient != 0) { 5944 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 5945 } 5946} 5947 5948 5949 5950void AudioFlinger::EffectHandle::setEnabled(bool enabled) 5951{ 5952 if (mEffectClient != 0) { 5953 mEffectClient->enableStatusChanged(enabled); 5954 } 5955} 5956 5957status_t AudioFlinger::EffectHandle::onTransact( 5958 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5959{ 5960 return BnEffect::onTransact(code, data, reply, flags); 5961} 5962 5963 5964void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 5965{ 5966 bool locked = tryLock(mCblk->lock); 5967 5968 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 5969 (mClient == NULL) ? getpid() : mClient->pid(), 5970 mPriority, 5971 mHasControl, 5972 !locked, 5973 mCblk->clientIndex, 5974 mCblk->serverIndex 5975 ); 5976 5977 if (locked) { 5978 mCblk->lock.unlock(); 5979 } 5980} 5981 5982#undef LOG_TAG 5983#define LOG_TAG "AudioFlinger::EffectChain" 5984 5985AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 5986 int sessionId) 5987 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false), 5988 mVolumeCtrlIdx(-1), mLeftVolume(0), mRightVolume(0), 5989 mNewLeftVolume(0), mNewRightVolume(0) 5990{ 5991 mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 5992} 5993 5994AudioFlinger::EffectChain::~EffectChain() 5995{ 5996 if (mOwnInBuffer) { 5997 delete mInBuffer; 5998 } 5999 6000} 6001 6002// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6003sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6004{ 6005 sp<EffectModule> effect; 6006 size_t size = mEffects.size(); 6007 6008 for (size_t i = 0; i < size; i++) { 6009 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6010 effect = mEffects[i]; 6011 break; 6012 } 6013 } 6014 return effect; 6015} 6016 6017// getEffectFromId_l() must be called with PlaybackThread::mLock held 6018sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6019{ 6020 sp<EffectModule> effect; 6021 size_t size = mEffects.size(); 6022 6023 for (size_t i = 0; i < size; i++) { 6024 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6025 if (id == 0 || mEffects[i]->id() == id) { 6026 effect = mEffects[i]; 6027 break; 6028 } 6029 } 6030 return effect; 6031} 6032 6033// Must be called with EffectChain::mLock locked 6034void AudioFlinger::EffectChain::process_l() 6035{ 6036 size_t size = mEffects.size(); 6037 for (size_t i = 0; i < size; i++) { 6038 mEffects[i]->process(); 6039 } 6040 for (size_t i = 0; i < size; i++) { 6041 mEffects[i]->updateState(); 6042 } 6043 // if no track is active, input buffer must be cleared here as the mixer process 6044 // will not do it 6045 if (mSessionId > 0 && activeTracks() == 0) { 6046 sp<ThreadBase> thread = mThread.promote(); 6047 if (thread != 0) { 6048 size_t numSamples = thread->frameCount() * thread->channelCount(); 6049 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6050 } 6051 } 6052} 6053 6054// addEffect_l() must be called with PlaybackThread::mLock held 6055status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6056{ 6057 effect_descriptor_t desc = effect->desc(); 6058 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6059 6060 Mutex::Autolock _l(mLock); 6061 effect->setChain(this); 6062 sp<ThreadBase> thread = mThread.promote(); 6063 if (thread == 0) { 6064 return NO_INIT; 6065 } 6066 effect->setThread(thread); 6067 6068 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6069 // Auxiliary effects are inserted at the beginning of mEffects vector as 6070 // they are processed first and accumulated in chain input buffer 6071 mEffects.insertAt(effect, 0); 6072 6073 // the input buffer for auxiliary effect contains mono samples in 6074 // 32 bit format. This is to avoid saturation in AudoMixer 6075 // accumulation stage. Saturation is done in EffectModule::process() before 6076 // calling the process in effect engine 6077 size_t numSamples = thread->frameCount(); 6078 int32_t *buffer = new int32_t[numSamples]; 6079 memset(buffer, 0, numSamples * sizeof(int32_t)); 6080 effect->setInBuffer((int16_t *)buffer); 6081 // auxiliary effects output samples to chain input buffer for further processing 6082 // by insert effects 6083 effect->setOutBuffer(mInBuffer); 6084 } else { 6085 // Insert effects are inserted at the end of mEffects vector as they are processed 6086 // after track and auxiliary effects. 6087 // Insert effect order as a function of indicated preference: 6088 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6089 // another effect is present 6090 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6091 // last effect claiming first position 6092 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6093 // first effect claiming last position 6094 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6095 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6096 // already present 6097 6098 int size = (int)mEffects.size(); 6099 int idx_insert = size; 6100 int idx_insert_first = -1; 6101 int idx_insert_last = -1; 6102 6103 for (int i = 0; i < size; i++) { 6104 effect_descriptor_t d = mEffects[i]->desc(); 6105 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6106 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6107 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6108 // check invalid effect chaining combinations 6109 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6110 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6111 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6112 return INVALID_OPERATION; 6113 } 6114 // remember position of first insert effect and by default 6115 // select this as insert position for new effect 6116 if (idx_insert == size) { 6117 idx_insert = i; 6118 } 6119 // remember position of last insert effect claiming 6120 // first position 6121 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6122 idx_insert_first = i; 6123 } 6124 // remember position of first insert effect claiming 6125 // last position 6126 if (iPref == EFFECT_FLAG_INSERT_LAST && 6127 idx_insert_last == -1) { 6128 idx_insert_last = i; 6129 } 6130 } 6131 } 6132 6133 // modify idx_insert from first position if needed 6134 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6135 if (idx_insert_last != -1) { 6136 idx_insert = idx_insert_last; 6137 } else { 6138 idx_insert = size; 6139 } 6140 } else { 6141 if (idx_insert_first != -1) { 6142 idx_insert = idx_insert_first + 1; 6143 } 6144 } 6145 6146 // always read samples from chain input buffer 6147 effect->setInBuffer(mInBuffer); 6148 6149 // if last effect in the chain, output samples to chain 6150 // output buffer, otherwise to chain input buffer 6151 if (idx_insert == size) { 6152 if (idx_insert != 0) { 6153 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6154 mEffects[idx_insert-1]->configure(); 6155 } 6156 effect->setOutBuffer(mOutBuffer); 6157 } else { 6158 effect->setOutBuffer(mInBuffer); 6159 } 6160 mEffects.insertAt(effect, idx_insert); 6161 6162 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6163 } 6164 effect->configure(); 6165 return NO_ERROR; 6166} 6167 6168// removeEffect_l() must be called with PlaybackThread::mLock held 6169size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6170{ 6171 Mutex::Autolock _l(mLock); 6172 int size = (int)mEffects.size(); 6173 int i; 6174 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6175 6176 for (i = 0; i < size; i++) { 6177 if (effect == mEffects[i]) { 6178 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6179 delete[] effect->inBuffer(); 6180 } else { 6181 if (i == size - 1 && i != 0) { 6182 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6183 mEffects[i - 1]->configure(); 6184 } 6185 } 6186 mEffects.removeAt(i); 6187 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6188 break; 6189 } 6190 } 6191 6192 return mEffects.size(); 6193} 6194 6195// setDevice_l() must be called with PlaybackThread::mLock held 6196void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6197{ 6198 size_t size = mEffects.size(); 6199 for (size_t i = 0; i < size; i++) { 6200 mEffects[i]->setDevice(device); 6201 } 6202} 6203 6204// setMode_l() must be called with PlaybackThread::mLock held 6205void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6206{ 6207 size_t size = mEffects.size(); 6208 for (size_t i = 0; i < size; i++) { 6209 mEffects[i]->setMode(mode); 6210 } 6211} 6212 6213// setVolume_l() must be called with PlaybackThread::mLock held 6214bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6215{ 6216 uint32_t newLeft = *left; 6217 uint32_t newRight = *right; 6218 bool hasControl = false; 6219 int ctrlIdx = -1; 6220 size_t size = mEffects.size(); 6221 6222 // first update volume controller 6223 for (size_t i = size; i > 0; i--) { 6224 if ((mEffects[i - 1]->state() >= EffectModule::ACTIVE) && 6225 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6226 ctrlIdx = i - 1; 6227 hasControl = true; 6228 break; 6229 } 6230 } 6231 6232 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6233 if (hasControl) { 6234 *left = mNewLeftVolume; 6235 *right = mNewRightVolume; 6236 } 6237 return hasControl; 6238 } 6239 6240 if (mVolumeCtrlIdx != -1) { 6241 hasControl = true; 6242 } 6243 mVolumeCtrlIdx = ctrlIdx; 6244 mLeftVolume = newLeft; 6245 mRightVolume = newRight; 6246 6247 // second get volume update from volume controller 6248 if (ctrlIdx >= 0) { 6249 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6250 mNewLeftVolume = newLeft; 6251 mNewRightVolume = newRight; 6252 } 6253 // then indicate volume to all other effects in chain. 6254 // Pass altered volume to effects before volume controller 6255 // and requested volume to effects after controller 6256 uint32_t lVol = newLeft; 6257 uint32_t rVol = newRight; 6258 6259 for (size_t i = 0; i < size; i++) { 6260 if ((int)i == ctrlIdx) continue; 6261 // this also works for ctrlIdx == -1 when there is no volume controller 6262 if ((int)i > ctrlIdx) { 6263 lVol = *left; 6264 rVol = *right; 6265 } 6266 mEffects[i]->setVolume(&lVol, &rVol, false); 6267 } 6268 *left = newLeft; 6269 *right = newRight; 6270 6271 return hasControl; 6272} 6273 6274status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6275{ 6276 const size_t SIZE = 256; 6277 char buffer[SIZE]; 6278 String8 result; 6279 6280 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6281 result.append(buffer); 6282 6283 bool locked = tryLock(mLock); 6284 // failed to lock - AudioFlinger is probably deadlocked 6285 if (!locked) { 6286 result.append("\tCould not lock mutex:\n"); 6287 } 6288 6289 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6290 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6291 mEffects.size(), 6292 (uint32_t)mInBuffer, 6293 (uint32_t)mOutBuffer, 6294 mActiveTrackCnt); 6295 result.append(buffer); 6296 write(fd, result.string(), result.size()); 6297 6298 for (size_t i = 0; i < mEffects.size(); ++i) { 6299 sp<EffectModule> effect = mEffects[i]; 6300 if (effect != 0) { 6301 effect->dump(fd, args); 6302 } 6303 } 6304 6305 if (locked) { 6306 mLock.unlock(); 6307 } 6308 6309 return NO_ERROR; 6310} 6311 6312#undef LOG_TAG 6313#define LOG_TAG "AudioFlinger" 6314 6315// ---------------------------------------------------------------------------- 6316 6317status_t AudioFlinger::onTransact( 6318 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6319{ 6320 return BnAudioFlinger::onTransact(code, data, reply, flags); 6321} 6322 6323}; // namespace android 6324