AudioFlinger.cpp revision e0aed6ddcb4e3c301b80aa26706b6052dab42c41
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IServiceManager.h> 28#include <utils/Log.h> 29#include <binder/Parcel.h> 30#include <binder/IPCThreadState.h> 31#include <utils/String16.h> 32#include <utils/threads.h> 33 34#include <cutils/properties.h> 35 36#include <media/AudioTrack.h> 37#include <media/AudioRecord.h> 38 39#include <private/media/AudioTrackShared.h> 40#include <private/media/AudioEffectShared.h> 41#include <hardware_legacy/AudioHardwareInterface.h> 42 43#include "AudioMixer.h" 44#include "AudioFlinger.h" 45 46#ifdef WITH_A2DP 47#include "A2dpAudioInterface.h" 48#endif 49 50#ifdef LVMX 51#include "lifevibes.h" 52#endif 53 54#include <media/EffectsFactoryApi.h> 55#include <media/EffectVisualizerApi.h> 56 57// ---------------------------------------------------------------------------- 58// the sim build doesn't have gettid 59 60#ifndef HAVE_GETTID 61# define gettid getpid 62#endif 63 64// ---------------------------------------------------------------------------- 65 66extern const char * const gEffectLibPath; 67 68namespace android { 69 70static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; 71static const char* kHardwareLockedString = "Hardware lock is taken\n"; 72 73//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 74static const float MAX_GAIN = 4096.0f; 75static const float MAX_GAIN_INT = 0x1000; 76 77// retry counts for buffer fill timeout 78// 50 * ~20msecs = 1 second 79static const int8_t kMaxTrackRetries = 50; 80static const int8_t kMaxTrackStartupRetries = 50; 81// allow less retry attempts on direct output thread. 82// direct outputs can be a scarce resource in audio hardware and should 83// be released as quickly as possible. 84static const int8_t kMaxTrackRetriesDirect = 2; 85 86static const int kDumpLockRetries = 50; 87static const int kDumpLockSleep = 20000; 88 89static const nsecs_t kWarningThrottle = seconds(5); 90 91 92#define AUDIOFLINGER_SECURITY_ENABLED 1 93 94// ---------------------------------------------------------------------------- 95 96static bool recordingAllowed() { 97#ifndef HAVE_ANDROID_OS 98 return true; 99#endif 100#if AUDIOFLINGER_SECURITY_ENABLED 101 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 102 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 103 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 104 return ok; 105#else 106 if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO"))) 107 LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest"); 108 return true; 109#endif 110} 111 112static bool settingsAllowed() { 113#ifndef HAVE_ANDROID_OS 114 return true; 115#endif 116#if AUDIOFLINGER_SECURITY_ENABLED 117 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 118 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 119 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 120 return ok; 121#else 122 if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"))) 123 LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest"); 124 return true; 125#endif 126} 127 128// ---------------------------------------------------------------------------- 129 130AudioFlinger::AudioFlinger() 131 : BnAudioFlinger(), 132 mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1) 133{ 134 mHardwareStatus = AUDIO_HW_IDLE; 135 136 mAudioHardware = AudioHardwareInterface::create(); 137 138 mHardwareStatus = AUDIO_HW_INIT; 139 if (mAudioHardware->initCheck() == NO_ERROR) { 140 // open 16-bit output stream for s/w mixer 141 mMode = AudioSystem::MODE_NORMAL; 142 setMode(mMode); 143 144 setMasterVolume(1.0f); 145 setMasterMute(false); 146 } else { 147 LOGE("Couldn't even initialize the stubbed audio hardware!"); 148 } 149#ifdef LVMX 150 LifeVibes::init(); 151 mLifeVibesClientPid = -1; 152#endif 153} 154 155AudioFlinger::~AudioFlinger() 156{ 157 while (!mRecordThreads.isEmpty()) { 158 // closeInput() will remove first entry from mRecordThreads 159 closeInput(mRecordThreads.keyAt(0)); 160 } 161 while (!mPlaybackThreads.isEmpty()) { 162 // closeOutput() will remove first entry from mPlaybackThreads 163 closeOutput(mPlaybackThreads.keyAt(0)); 164 } 165 if (mAudioHardware) { 166 delete mAudioHardware; 167 } 168} 169 170 171 172status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 173{ 174 const size_t SIZE = 256; 175 char buffer[SIZE]; 176 String8 result; 177 178 result.append("Clients:\n"); 179 for (size_t i = 0; i < mClients.size(); ++i) { 180 wp<Client> wClient = mClients.valueAt(i); 181 if (wClient != 0) { 182 sp<Client> client = wClient.promote(); 183 if (client != 0) { 184 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 185 result.append(buffer); 186 } 187 } 188 } 189 write(fd, result.string(), result.size()); 190 return NO_ERROR; 191} 192 193 194status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 195{ 196 const size_t SIZE = 256; 197 char buffer[SIZE]; 198 String8 result; 199 int hardwareStatus = mHardwareStatus; 200 201 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 202 result.append(buffer); 203 write(fd, result.string(), result.size()); 204 return NO_ERROR; 205} 206 207status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 208{ 209 const size_t SIZE = 256; 210 char buffer[SIZE]; 211 String8 result; 212 snprintf(buffer, SIZE, "Permission Denial: " 213 "can't dump AudioFlinger from pid=%d, uid=%d\n", 214 IPCThreadState::self()->getCallingPid(), 215 IPCThreadState::self()->getCallingUid()); 216 result.append(buffer); 217 write(fd, result.string(), result.size()); 218 return NO_ERROR; 219} 220 221static bool tryLock(Mutex& mutex) 222{ 223 bool locked = false; 224 for (int i = 0; i < kDumpLockRetries; ++i) { 225 if (mutex.tryLock() == NO_ERROR) { 226 locked = true; 227 break; 228 } 229 usleep(kDumpLockSleep); 230 } 231 return locked; 232} 233 234status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 235{ 236 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 237 dumpPermissionDenial(fd, args); 238 } else { 239 // get state of hardware lock 240 bool hardwareLocked = tryLock(mHardwareLock); 241 if (!hardwareLocked) { 242 String8 result(kHardwareLockedString); 243 write(fd, result.string(), result.size()); 244 } else { 245 mHardwareLock.unlock(); 246 } 247 248 bool locked = tryLock(mLock); 249 250 // failed to lock - AudioFlinger is probably deadlocked 251 if (!locked) { 252 String8 result(kDeadlockedString); 253 write(fd, result.string(), result.size()); 254 } 255 256 dumpClients(fd, args); 257 dumpInternals(fd, args); 258 259 // dump playback threads 260 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 261 mPlaybackThreads.valueAt(i)->dump(fd, args); 262 } 263 264 // dump record threads 265 for (size_t i = 0; i < mRecordThreads.size(); i++) { 266 mRecordThreads.valueAt(i)->dump(fd, args); 267 } 268 269 if (mAudioHardware) { 270 mAudioHardware->dumpState(fd, args); 271 } 272 if (locked) mLock.unlock(); 273 } 274 return NO_ERROR; 275} 276 277 278// IAudioFlinger interface 279 280 281sp<IAudioTrack> AudioFlinger::createTrack( 282 pid_t pid, 283 int streamType, 284 uint32_t sampleRate, 285 int format, 286 int channelCount, 287 int frameCount, 288 uint32_t flags, 289 const sp<IMemory>& sharedBuffer, 290 int output, 291 int *sessionId, 292 status_t *status) 293{ 294 sp<PlaybackThread::Track> track; 295 sp<TrackHandle> trackHandle; 296 sp<Client> client; 297 wp<Client> wclient; 298 status_t lStatus; 299 int lSessionId; 300 301 if (streamType >= AudioSystem::NUM_STREAM_TYPES) { 302 LOGE("invalid stream type"); 303 lStatus = BAD_VALUE; 304 goto Exit; 305 } 306 307 { 308 Mutex::Autolock _l(mLock); 309 PlaybackThread *thread = checkPlaybackThread_l(output); 310 PlaybackThread *effectThread = NULL; 311 if (thread == NULL) { 312 LOGE("unknown output thread"); 313 lStatus = BAD_VALUE; 314 goto Exit; 315 } 316 317 wclient = mClients.valueFor(pid); 318 319 if (wclient != NULL) { 320 client = wclient.promote(); 321 } else { 322 client = new Client(this, pid); 323 mClients.add(pid, client); 324 } 325 326 LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 327 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 328 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 329 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 330 if (mPlaybackThreads.keyAt(i) != output) { 331 // prevent same audio session on different output threads 332 uint32_t sessions = t->hasAudioSession(*sessionId); 333 if (sessions & PlaybackThread::TRACK_SESSION) { 334 lStatus = BAD_VALUE; 335 goto Exit; 336 } 337 // check if an effect with same session ID is waiting for a track to be created 338 if (sessions & PlaybackThread::EFFECT_SESSION) { 339 effectThread = t.get(); 340 } 341 } 342 } 343 lSessionId = *sessionId; 344 } else { 345 // if no audio session id is provided, create one here 346 lSessionId = nextUniqueId(); 347 if (sessionId != NULL) { 348 *sessionId = lSessionId; 349 } 350 } 351 LOGV("createTrack() lSessionId: %d", lSessionId); 352 353 track = thread->createTrack_l(client, streamType, sampleRate, format, 354 channelCount, frameCount, sharedBuffer, lSessionId, &lStatus); 355 356 // move effect chain to this output thread if an effect on same session was waiting 357 // for a track to be created 358 if (lStatus == NO_ERROR && effectThread != NULL) { 359 Mutex::Autolock _dl(thread->mLock); 360 Mutex::Autolock _sl(effectThread->mLock); 361 moveEffectChain_l(lSessionId, effectThread, thread, true); 362 } 363 } 364 if (lStatus == NO_ERROR) { 365 trackHandle = new TrackHandle(track); 366 } else { 367 // remove local strong reference to Client before deleting the Track so that the Client 368 // destructor is called by the TrackBase destructor with mLock held 369 client.clear(); 370 track.clear(); 371 } 372 373Exit: 374 if(status) { 375 *status = lStatus; 376 } 377 return trackHandle; 378} 379 380uint32_t AudioFlinger::sampleRate(int output) const 381{ 382 Mutex::Autolock _l(mLock); 383 PlaybackThread *thread = checkPlaybackThread_l(output); 384 if (thread == NULL) { 385 LOGW("sampleRate() unknown thread %d", output); 386 return 0; 387 } 388 return thread->sampleRate(); 389} 390 391int AudioFlinger::channelCount(int output) const 392{ 393 Mutex::Autolock _l(mLock); 394 PlaybackThread *thread = checkPlaybackThread_l(output); 395 if (thread == NULL) { 396 LOGW("channelCount() unknown thread %d", output); 397 return 0; 398 } 399 return thread->channelCount(); 400} 401 402int AudioFlinger::format(int output) const 403{ 404 Mutex::Autolock _l(mLock); 405 PlaybackThread *thread = checkPlaybackThread_l(output); 406 if (thread == NULL) { 407 LOGW("format() unknown thread %d", output); 408 return 0; 409 } 410 return thread->format(); 411} 412 413size_t AudioFlinger::frameCount(int output) const 414{ 415 Mutex::Autolock _l(mLock); 416 PlaybackThread *thread = checkPlaybackThread_l(output); 417 if (thread == NULL) { 418 LOGW("frameCount() unknown thread %d", output); 419 return 0; 420 } 421 return thread->frameCount(); 422} 423 424uint32_t AudioFlinger::latency(int output) const 425{ 426 Mutex::Autolock _l(mLock); 427 PlaybackThread *thread = checkPlaybackThread_l(output); 428 if (thread == NULL) { 429 LOGW("latency() unknown thread %d", output); 430 return 0; 431 } 432 return thread->latency(); 433} 434 435status_t AudioFlinger::setMasterVolume(float value) 436{ 437 // check calling permissions 438 if (!settingsAllowed()) { 439 return PERMISSION_DENIED; 440 } 441 442 // when hw supports master volume, don't scale in sw mixer 443 AutoMutex lock(mHardwareLock); 444 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 445 if (mAudioHardware->setMasterVolume(value) == NO_ERROR) { 446 value = 1.0f; 447 } 448 mHardwareStatus = AUDIO_HW_IDLE; 449 450 mMasterVolume = value; 451 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 452 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 453 454 return NO_ERROR; 455} 456 457status_t AudioFlinger::setMode(int mode) 458{ 459 status_t ret; 460 461 // check calling permissions 462 if (!settingsAllowed()) { 463 return PERMISSION_DENIED; 464 } 465 if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) { 466 LOGW("Illegal value: setMode(%d)", mode); 467 return BAD_VALUE; 468 } 469 470 { // scope for the lock 471 AutoMutex lock(mHardwareLock); 472 mHardwareStatus = AUDIO_HW_SET_MODE; 473 ret = mAudioHardware->setMode(mode); 474 mHardwareStatus = AUDIO_HW_IDLE; 475 } 476 477 if (NO_ERROR == ret) { 478 Mutex::Autolock _l(mLock); 479 mMode = mode; 480 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 481 mPlaybackThreads.valueAt(i)->setMode(mode); 482#ifdef LVMX 483 LifeVibes::setMode(mode); 484#endif 485 } 486 487 return ret; 488} 489 490status_t AudioFlinger::setMicMute(bool state) 491{ 492 // check calling permissions 493 if (!settingsAllowed()) { 494 return PERMISSION_DENIED; 495 } 496 497 AutoMutex lock(mHardwareLock); 498 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 499 status_t ret = mAudioHardware->setMicMute(state); 500 mHardwareStatus = AUDIO_HW_IDLE; 501 return ret; 502} 503 504bool AudioFlinger::getMicMute() const 505{ 506 bool state = AudioSystem::MODE_INVALID; 507 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 508 mAudioHardware->getMicMute(&state); 509 mHardwareStatus = AUDIO_HW_IDLE; 510 return state; 511} 512 513status_t AudioFlinger::setMasterMute(bool muted) 514{ 515 // check calling permissions 516 if (!settingsAllowed()) { 517 return PERMISSION_DENIED; 518 } 519 520 mMasterMute = muted; 521 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 522 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 523 524 return NO_ERROR; 525} 526 527float AudioFlinger::masterVolume() const 528{ 529 return mMasterVolume; 530} 531 532bool AudioFlinger::masterMute() const 533{ 534 return mMasterMute; 535} 536 537status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 538{ 539 // check calling permissions 540 if (!settingsAllowed()) { 541 return PERMISSION_DENIED; 542 } 543 544 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { 545 return BAD_VALUE; 546 } 547 548 AutoMutex lock(mLock); 549 PlaybackThread *thread = NULL; 550 if (output) { 551 thread = checkPlaybackThread_l(output); 552 if (thread == NULL) { 553 return BAD_VALUE; 554 } 555 } 556 557 mStreamTypes[stream].volume = value; 558 559 if (thread == NULL) { 560 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 561 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 562 } 563 } else { 564 thread->setStreamVolume(stream, value); 565 } 566 567 return NO_ERROR; 568} 569 570status_t AudioFlinger::setStreamMute(int stream, bool muted) 571{ 572 // check calling permissions 573 if (!settingsAllowed()) { 574 return PERMISSION_DENIED; 575 } 576 577 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || 578 uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { 579 return BAD_VALUE; 580 } 581 582 mStreamTypes[stream].mute = muted; 583 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 584 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 585 586 return NO_ERROR; 587} 588 589float AudioFlinger::streamVolume(int stream, int output) const 590{ 591 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { 592 return 0.0f; 593 } 594 595 AutoMutex lock(mLock); 596 float volume; 597 if (output) { 598 PlaybackThread *thread = checkPlaybackThread_l(output); 599 if (thread == NULL) { 600 return 0.0f; 601 } 602 volume = thread->streamVolume(stream); 603 } else { 604 volume = mStreamTypes[stream].volume; 605 } 606 607 return volume; 608} 609 610bool AudioFlinger::streamMute(int stream) const 611{ 612 if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) { 613 return true; 614 } 615 616 return mStreamTypes[stream].mute; 617} 618 619bool AudioFlinger::isStreamActive(int stream) const 620{ 621 Mutex::Autolock _l(mLock); 622 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 623 if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) { 624 return true; 625 } 626 } 627 return false; 628} 629 630status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 631{ 632 status_t result; 633 634 LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 635 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 636 // check calling permissions 637 if (!settingsAllowed()) { 638 return PERMISSION_DENIED; 639 } 640 641#ifdef LVMX 642 AudioParameter param = AudioParameter(keyValuePairs); 643 LifeVibes::setParameters(ioHandle,keyValuePairs); 644 String8 key = String8(AudioParameter::keyRouting); 645 int device; 646 if (NO_ERROR != param.getInt(key, device)) { 647 device = -1; 648 } 649 650 key = String8(LifevibesTag); 651 String8 value; 652 int musicEnabled = -1; 653 if (NO_ERROR == param.get(key, value)) { 654 if (value == LifevibesEnable) { 655 mLifeVibesClientPid = IPCThreadState::self()->getCallingPid(); 656 musicEnabled = 1; 657 } else if (value == LifevibesDisable) { 658 mLifeVibesClientPid = -1; 659 musicEnabled = 0; 660 } 661 } 662#endif 663 664 // ioHandle == 0 means the parameters are global to the audio hardware interface 665 if (ioHandle == 0) { 666 AutoMutex lock(mHardwareLock); 667 mHardwareStatus = AUDIO_SET_PARAMETER; 668 result = mAudioHardware->setParameters(keyValuePairs); 669#ifdef LVMX 670 if (musicEnabled != -1) { 671 LifeVibes::enableMusic((bool) musicEnabled); 672 } 673#endif 674 mHardwareStatus = AUDIO_HW_IDLE; 675 return result; 676 } 677 678 // hold a strong ref on thread in case closeOutput() or closeInput() is called 679 // and the thread is exited once the lock is released 680 sp<ThreadBase> thread; 681 { 682 Mutex::Autolock _l(mLock); 683 thread = checkPlaybackThread_l(ioHandle); 684 if (thread == NULL) { 685 thread = checkRecordThread_l(ioHandle); 686 } 687 } 688 if (thread != NULL) { 689 result = thread->setParameters(keyValuePairs); 690#ifdef LVMX 691 if ((NO_ERROR == result) && (device != -1)) { 692 LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device); 693 } 694#endif 695 return result; 696 } 697 return BAD_VALUE; 698} 699 700String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 701{ 702// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 703// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 704 705 if (ioHandle == 0) { 706 return mAudioHardware->getParameters(keys); 707 } 708 709 Mutex::Autolock _l(mLock); 710 711 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 712 if (playbackThread != NULL) { 713 return playbackThread->getParameters(keys); 714 } 715 RecordThread *recordThread = checkRecordThread_l(ioHandle); 716 if (recordThread != NULL) { 717 return recordThread->getParameters(keys); 718 } 719 return String8(""); 720} 721 722size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 723{ 724 return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); 725} 726 727unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 728{ 729 if (ioHandle == 0) { 730 return 0; 731 } 732 733 Mutex::Autolock _l(mLock); 734 735 RecordThread *recordThread = checkRecordThread_l(ioHandle); 736 if (recordThread != NULL) { 737 return recordThread->getInputFramesLost(); 738 } 739 return 0; 740} 741 742status_t AudioFlinger::setVoiceVolume(float value) 743{ 744 // check calling permissions 745 if (!settingsAllowed()) { 746 return PERMISSION_DENIED; 747 } 748 749 AutoMutex lock(mHardwareLock); 750 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 751 status_t ret = mAudioHardware->setVoiceVolume(value); 752 mHardwareStatus = AUDIO_HW_IDLE; 753 754 return ret; 755} 756 757status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 758{ 759 status_t status; 760 761 Mutex::Autolock _l(mLock); 762 763 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 764 if (playbackThread != NULL) { 765 return playbackThread->getRenderPosition(halFrames, dspFrames); 766 } 767 768 return BAD_VALUE; 769} 770 771void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 772{ 773 774 Mutex::Autolock _l(mLock); 775 776 int pid = IPCThreadState::self()->getCallingPid(); 777 if (mNotificationClients.indexOfKey(pid) < 0) { 778 sp<NotificationClient> notificationClient = new NotificationClient(this, 779 client, 780 pid); 781 LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 782 783 mNotificationClients.add(pid, notificationClient); 784 785 sp<IBinder> binder = client->asBinder(); 786 binder->linkToDeath(notificationClient); 787 788 // the config change is always sent from playback or record threads to avoid deadlock 789 // with AudioSystem::gLock 790 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 791 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 792 } 793 794 for (size_t i = 0; i < mRecordThreads.size(); i++) { 795 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 796 } 797 } 798} 799 800void AudioFlinger::removeNotificationClient(pid_t pid) 801{ 802 Mutex::Autolock _l(mLock); 803 804 int index = mNotificationClients.indexOfKey(pid); 805 if (index >= 0) { 806 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 807 LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 808#ifdef LVMX 809 if (pid == mLifeVibesClientPid) { 810 LOGV("Disabling lifevibes"); 811 LifeVibes::enableMusic(false); 812 mLifeVibesClientPid = -1; 813 } 814#endif 815 mNotificationClients.removeItem(pid); 816 } 817} 818 819// audioConfigChanged_l() must be called with AudioFlinger::mLock held 820void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 821{ 822 size_t size = mNotificationClients.size(); 823 for (size_t i = 0; i < size; i++) { 824 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 825 } 826} 827 828// removeClient_l() must be called with AudioFlinger::mLock held 829void AudioFlinger::removeClient_l(pid_t pid) 830{ 831 LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 832 mClients.removeItem(pid); 833} 834 835 836// ---------------------------------------------------------------------------- 837 838AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id) 839 : Thread(false), 840 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 841 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false) 842{ 843} 844 845AudioFlinger::ThreadBase::~ThreadBase() 846{ 847 mParamCond.broadcast(); 848 mNewParameters.clear(); 849} 850 851void AudioFlinger::ThreadBase::exit() 852{ 853 // keep a strong ref on ourself so that we wont get 854 // destroyed in the middle of requestExitAndWait() 855 sp <ThreadBase> strongMe = this; 856 857 LOGV("ThreadBase::exit"); 858 { 859 AutoMutex lock(&mLock); 860 mExiting = true; 861 requestExit(); 862 mWaitWorkCV.signal(); 863 } 864 requestExitAndWait(); 865} 866 867uint32_t AudioFlinger::ThreadBase::sampleRate() const 868{ 869 return mSampleRate; 870} 871 872int AudioFlinger::ThreadBase::channelCount() const 873{ 874 return (int)mChannelCount; 875} 876 877int AudioFlinger::ThreadBase::format() const 878{ 879 return mFormat; 880} 881 882size_t AudioFlinger::ThreadBase::frameCount() const 883{ 884 return mFrameCount; 885} 886 887status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 888{ 889 status_t status; 890 891 LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 892 Mutex::Autolock _l(mLock); 893 894 mNewParameters.add(keyValuePairs); 895 mWaitWorkCV.signal(); 896 // wait condition with timeout in case the thread loop has exited 897 // before the request could be processed 898 if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { 899 status = mParamStatus; 900 mWaitWorkCV.signal(); 901 } else { 902 status = TIMED_OUT; 903 } 904 return status; 905} 906 907void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 908{ 909 Mutex::Autolock _l(mLock); 910 sendConfigEvent_l(event, param); 911} 912 913// sendConfigEvent_l() must be called with ThreadBase::mLock held 914void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 915{ 916 ConfigEvent *configEvent = new ConfigEvent(); 917 configEvent->mEvent = event; 918 configEvent->mParam = param; 919 mConfigEvents.add(configEvent); 920 LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 921 mWaitWorkCV.signal(); 922} 923 924void AudioFlinger::ThreadBase::processConfigEvents() 925{ 926 mLock.lock(); 927 while(!mConfigEvents.isEmpty()) { 928 LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 929 ConfigEvent *configEvent = mConfigEvents[0]; 930 mConfigEvents.removeAt(0); 931 // release mLock before locking AudioFlinger mLock: lock order is always 932 // AudioFlinger then ThreadBase to avoid cross deadlock 933 mLock.unlock(); 934 mAudioFlinger->mLock.lock(); 935 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 936 mAudioFlinger->mLock.unlock(); 937 delete configEvent; 938 mLock.lock(); 939 } 940 mLock.unlock(); 941} 942 943status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 944{ 945 const size_t SIZE = 256; 946 char buffer[SIZE]; 947 String8 result; 948 949 bool locked = tryLock(mLock); 950 if (!locked) { 951 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 952 write(fd, buffer, strlen(buffer)); 953 } 954 955 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 956 result.append(buffer); 957 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 958 result.append(buffer); 959 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 960 result.append(buffer); 961 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 962 result.append(buffer); 963 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 964 result.append(buffer); 965 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 966 result.append(buffer); 967 968 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 969 result.append(buffer); 970 result.append(" Index Command"); 971 for (size_t i = 0; i < mNewParameters.size(); ++i) { 972 snprintf(buffer, SIZE, "\n %02d ", i); 973 result.append(buffer); 974 result.append(mNewParameters[i]); 975 } 976 977 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 978 result.append(buffer); 979 snprintf(buffer, SIZE, " Index event param\n"); 980 result.append(buffer); 981 for (size_t i = 0; i < mConfigEvents.size(); i++) { 982 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 983 result.append(buffer); 984 } 985 result.append("\n"); 986 987 write(fd, result.string(), result.size()); 988 989 if (locked) { 990 mLock.unlock(); 991 } 992 return NO_ERROR; 993} 994 995 996// ---------------------------------------------------------------------------- 997 998AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 999 : ThreadBase(audioFlinger, id), 1000 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 1001 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1002 mDevice(device) 1003{ 1004 readOutputParameters(); 1005 1006 mMasterVolume = mAudioFlinger->masterVolume(); 1007 mMasterMute = mAudioFlinger->masterMute(); 1008 1009 for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { 1010 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1011 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1012 } 1013} 1014 1015AudioFlinger::PlaybackThread::~PlaybackThread() 1016{ 1017 delete [] mMixBuffer; 1018} 1019 1020status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1021{ 1022 dumpInternals(fd, args); 1023 dumpTracks(fd, args); 1024 dumpEffectChains(fd, args); 1025 return NO_ERROR; 1026} 1027 1028status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1029{ 1030 const size_t SIZE = 256; 1031 char buffer[SIZE]; 1032 String8 result; 1033 1034 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1035 result.append(buffer); 1036 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1037 for (size_t i = 0; i < mTracks.size(); ++i) { 1038 sp<Track> track = mTracks[i]; 1039 if (track != 0) { 1040 track->dump(buffer, SIZE); 1041 result.append(buffer); 1042 } 1043 } 1044 1045 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1046 result.append(buffer); 1047 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1048 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1049 wp<Track> wTrack = mActiveTracks[i]; 1050 if (wTrack != 0) { 1051 sp<Track> track = wTrack.promote(); 1052 if (track != 0) { 1053 track->dump(buffer, SIZE); 1054 result.append(buffer); 1055 } 1056 } 1057 } 1058 write(fd, result.string(), result.size()); 1059 return NO_ERROR; 1060} 1061 1062status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args) 1063{ 1064 const size_t SIZE = 256; 1065 char buffer[SIZE]; 1066 String8 result; 1067 1068 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1069 write(fd, buffer, strlen(buffer)); 1070 1071 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1072 sp<EffectChain> chain = mEffectChains[i]; 1073 if (chain != 0) { 1074 chain->dump(fd, args); 1075 } 1076 } 1077 return NO_ERROR; 1078} 1079 1080status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1081{ 1082 const size_t SIZE = 256; 1083 char buffer[SIZE]; 1084 String8 result; 1085 1086 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1087 result.append(buffer); 1088 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1089 result.append(buffer); 1090 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1091 result.append(buffer); 1092 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1093 result.append(buffer); 1094 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1095 result.append(buffer); 1096 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1097 result.append(buffer); 1098 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1099 result.append(buffer); 1100 write(fd, result.string(), result.size()); 1101 1102 dumpBase(fd, args); 1103 1104 return NO_ERROR; 1105} 1106 1107// Thread virtuals 1108status_t AudioFlinger::PlaybackThread::readyToRun() 1109{ 1110 if (mSampleRate == 0) { 1111 LOGE("No working audio driver found."); 1112 return NO_INIT; 1113 } 1114 LOGI("AudioFlinger's thread %p ready to run", this); 1115 return NO_ERROR; 1116} 1117 1118void AudioFlinger::PlaybackThread::onFirstRef() 1119{ 1120 const size_t SIZE = 256; 1121 char buffer[SIZE]; 1122 1123 snprintf(buffer, SIZE, "Playback Thread %p", this); 1124 1125 run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); 1126} 1127 1128// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1129sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1130 const sp<AudioFlinger::Client>& client, 1131 int streamType, 1132 uint32_t sampleRate, 1133 int format, 1134 int channelCount, 1135 int frameCount, 1136 const sp<IMemory>& sharedBuffer, 1137 int sessionId, 1138 status_t *status) 1139{ 1140 sp<Track> track; 1141 status_t lStatus; 1142 1143 if (mType == DIRECT) { 1144 if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) { 1145 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p", 1146 sampleRate, format, channelCount, mOutput); 1147 lStatus = BAD_VALUE; 1148 goto Exit; 1149 } 1150 } else { 1151 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1152 if (sampleRate > mSampleRate*2) { 1153 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1154 lStatus = BAD_VALUE; 1155 goto Exit; 1156 } 1157 } 1158 1159 if (mOutput == 0) { 1160 LOGE("Audio driver not initialized."); 1161 lStatus = NO_INIT; 1162 goto Exit; 1163 } 1164 1165 { // scope for mLock 1166 Mutex::Autolock _l(mLock); 1167 1168 // all tracks in same audio session must share the same routing strategy otherwise 1169 // conflicts will happen when tracks are moved from one output to another by audio policy 1170 // manager 1171 uint32_t strategy = 1172 AudioSystem::getStrategyForStream((AudioSystem::stream_type)streamType); 1173 for (size_t i = 0; i < mTracks.size(); ++i) { 1174 sp<Track> t = mTracks[i]; 1175 if (t != 0) { 1176 if (sessionId == t->sessionId() && 1177 strategy != AudioSystem::getStrategyForStream((AudioSystem::stream_type)t->type())) { 1178 lStatus = BAD_VALUE; 1179 goto Exit; 1180 } 1181 } 1182 } 1183 1184 track = new Track(this, client, streamType, sampleRate, format, 1185 channelCount, frameCount, sharedBuffer, sessionId); 1186 if (track->getCblk() == NULL || track->name() < 0) { 1187 lStatus = NO_MEMORY; 1188 goto Exit; 1189 } 1190 mTracks.add(track); 1191 1192 sp<EffectChain> chain = getEffectChain_l(sessionId); 1193 if (chain != 0) { 1194 LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1195 track->setMainBuffer(chain->inBuffer()); 1196 chain->setStrategy(AudioSystem::getStrategyForStream((AudioSystem::stream_type)track->type())); 1197 } 1198 } 1199 lStatus = NO_ERROR; 1200 1201Exit: 1202 if(status) { 1203 *status = lStatus; 1204 } 1205 return track; 1206} 1207 1208uint32_t AudioFlinger::PlaybackThread::latency() const 1209{ 1210 if (mOutput) { 1211 return mOutput->latency(); 1212 } 1213 else { 1214 return 0; 1215 } 1216} 1217 1218status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1219{ 1220#ifdef LVMX 1221 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1222 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1223 LifeVibes::setMasterVolume(audioOutputType, value); 1224 } 1225#endif 1226 mMasterVolume = value; 1227 return NO_ERROR; 1228} 1229 1230status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1231{ 1232#ifdef LVMX 1233 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1234 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1235 LifeVibes::setMasterMute(audioOutputType, muted); 1236 } 1237#endif 1238 mMasterMute = muted; 1239 return NO_ERROR; 1240} 1241 1242float AudioFlinger::PlaybackThread::masterVolume() const 1243{ 1244 return mMasterVolume; 1245} 1246 1247bool AudioFlinger::PlaybackThread::masterMute() const 1248{ 1249 return mMasterMute; 1250} 1251 1252status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1253{ 1254#ifdef LVMX 1255 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1256 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1257 LifeVibes::setStreamVolume(audioOutputType, stream, value); 1258 } 1259#endif 1260 mStreamTypes[stream].volume = value; 1261 return NO_ERROR; 1262} 1263 1264status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1265{ 1266#ifdef LVMX 1267 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1268 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1269 LifeVibes::setStreamMute(audioOutputType, stream, muted); 1270 } 1271#endif 1272 mStreamTypes[stream].mute = muted; 1273 return NO_ERROR; 1274} 1275 1276float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1277{ 1278 return mStreamTypes[stream].volume; 1279} 1280 1281bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1282{ 1283 return mStreamTypes[stream].mute; 1284} 1285 1286bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const 1287{ 1288 Mutex::Autolock _l(mLock); 1289 size_t count = mActiveTracks.size(); 1290 for (size_t i = 0 ; i < count ; ++i) { 1291 sp<Track> t = mActiveTracks[i].promote(); 1292 if (t == 0) continue; 1293 Track* const track = t.get(); 1294 if (t->type() == stream) 1295 return true; 1296 } 1297 return false; 1298} 1299 1300// addTrack_l() must be called with ThreadBase::mLock held 1301status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1302{ 1303 status_t status = ALREADY_EXISTS; 1304 1305 // set retry count for buffer fill 1306 track->mRetryCount = kMaxTrackStartupRetries; 1307 if (mActiveTracks.indexOf(track) < 0) { 1308 // the track is newly added, make sure it fills up all its 1309 // buffers before playing. This is to ensure the client will 1310 // effectively get the latency it requested. 1311 track->mFillingUpStatus = Track::FS_FILLING; 1312 track->mResetDone = false; 1313 mActiveTracks.add(track); 1314 if (track->mainBuffer() != mMixBuffer) { 1315 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1316 if (chain != 0) { 1317 LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1318 chain->startTrack(); 1319 } 1320 } 1321 1322 status = NO_ERROR; 1323 } 1324 1325 LOGV("mWaitWorkCV.broadcast"); 1326 mWaitWorkCV.broadcast(); 1327 1328 return status; 1329} 1330 1331// destroyTrack_l() must be called with ThreadBase::mLock held 1332void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1333{ 1334 track->mState = TrackBase::TERMINATED; 1335 if (mActiveTracks.indexOf(track) < 0) { 1336 mTracks.remove(track); 1337 deleteTrackName_l(track->name()); 1338 } 1339} 1340 1341String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1342{ 1343 return mOutput->getParameters(keys); 1344} 1345 1346// destroyTrack_l() must be called with AudioFlinger::mLock held 1347void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1348 AudioSystem::OutputDescriptor desc; 1349 void *param2 = 0; 1350 1351 LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1352 1353 switch (event) { 1354 case AudioSystem::OUTPUT_OPENED: 1355 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1356 desc.channels = mChannels; 1357 desc.samplingRate = mSampleRate; 1358 desc.format = mFormat; 1359 desc.frameCount = mFrameCount; 1360 desc.latency = latency(); 1361 param2 = &desc; 1362 break; 1363 1364 case AudioSystem::STREAM_CONFIG_CHANGED: 1365 param2 = ¶m; 1366 case AudioSystem::OUTPUT_CLOSED: 1367 default: 1368 break; 1369 } 1370 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1371} 1372 1373void AudioFlinger::PlaybackThread::readOutputParameters() 1374{ 1375 mSampleRate = mOutput->sampleRate(); 1376 mChannels = mOutput->channels(); 1377 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 1378 mFormat = mOutput->format(); 1379 mFrameSize = (uint16_t)mOutput->frameSize(); 1380 mFrameCount = mOutput->bufferSize() / mFrameSize; 1381 1382 // FIXME - Current mixer implementation only supports stereo output: Always 1383 // Allocate a stereo buffer even if HW output is mono. 1384 if (mMixBuffer != NULL) delete[] mMixBuffer; 1385 mMixBuffer = new int16_t[mFrameCount * 2]; 1386 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1387 1388 // force reconfiguration of effect chains and engines to take new buffer size and audio 1389 // parameters into account 1390 // Note that mLock is not held when readOutputParameters() is called from the constructor 1391 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1392 // matter. 1393 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1394 Vector< sp<EffectChain> > effectChains = mEffectChains; 1395 for (size_t i = 0; i < effectChains.size(); i ++) { 1396 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1397 } 1398} 1399 1400status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1401{ 1402 if (halFrames == 0 || dspFrames == 0) { 1403 return BAD_VALUE; 1404 } 1405 if (mOutput == 0) { 1406 return INVALID_OPERATION; 1407 } 1408 *halFrames = mBytesWritten/mOutput->frameSize(); 1409 1410 return mOutput->getRenderPosition(dspFrames); 1411} 1412 1413uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1414{ 1415 Mutex::Autolock _l(mLock); 1416 uint32_t result = 0; 1417 if (getEffectChain_l(sessionId) != 0) { 1418 result = EFFECT_SESSION; 1419 } 1420 1421 for (size_t i = 0; i < mTracks.size(); ++i) { 1422 sp<Track> track = mTracks[i]; 1423 if (sessionId == track->sessionId() && 1424 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1425 result |= TRACK_SESSION; 1426 break; 1427 } 1428 } 1429 1430 return result; 1431} 1432 1433uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1434{ 1435 // session AudioSystem::SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1436 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1437 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 1438 return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 1439 } 1440 for (size_t i = 0; i < mTracks.size(); i++) { 1441 sp<Track> track = mTracks[i]; 1442 if (sessionId == track->sessionId() && 1443 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1444 return AudioSystem::getStrategyForStream((AudioSystem::stream_type) track->type()); 1445 } 1446 } 1447 return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 1448} 1449 1450sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId) 1451{ 1452 Mutex::Autolock _l(mLock); 1453 return getEffectChain_l(sessionId); 1454} 1455 1456sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId) 1457{ 1458 sp<EffectChain> chain; 1459 1460 size_t size = mEffectChains.size(); 1461 for (size_t i = 0; i < size; i++) { 1462 if (mEffectChains[i]->sessionId() == sessionId) { 1463 chain = mEffectChains[i]; 1464 break; 1465 } 1466 } 1467 return chain; 1468} 1469 1470void AudioFlinger::PlaybackThread::setMode(uint32_t mode) 1471{ 1472 Mutex::Autolock _l(mLock); 1473 size_t size = mEffectChains.size(); 1474 for (size_t i = 0; i < size; i++) { 1475 mEffectChains[i]->setMode_l(mode); 1476 } 1477} 1478 1479// ---------------------------------------------------------------------------- 1480 1481AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1482 : PlaybackThread(audioFlinger, output, id, device), 1483 mAudioMixer(0) 1484{ 1485 mType = PlaybackThread::MIXER; 1486 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1487 1488 // FIXME - Current mixer implementation only supports stereo output 1489 if (mChannelCount == 1) { 1490 LOGE("Invalid audio hardware channel count"); 1491 } 1492} 1493 1494AudioFlinger::MixerThread::~MixerThread() 1495{ 1496 delete mAudioMixer; 1497} 1498 1499bool AudioFlinger::MixerThread::threadLoop() 1500{ 1501 Vector< sp<Track> > tracksToRemove; 1502 uint32_t mixerStatus = MIXER_IDLE; 1503 nsecs_t standbyTime = systemTime(); 1504 size_t mixBufferSize = mFrameCount * mFrameSize; 1505 // FIXME: Relaxed timing because of a certain device that can't meet latency 1506 // Should be reduced to 2x after the vendor fixes the driver issue 1507 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1508 nsecs_t lastWarning = 0; 1509 bool longStandbyExit = false; 1510 uint32_t activeSleepTime = activeSleepTimeUs(); 1511 uint32_t idleSleepTime = idleSleepTimeUs(); 1512 uint32_t sleepTime = idleSleepTime; 1513 Vector< sp<EffectChain> > effectChains; 1514 1515 while (!exitPending()) 1516 { 1517 processConfigEvents(); 1518 1519 mixerStatus = MIXER_IDLE; 1520 { // scope for mLock 1521 1522 Mutex::Autolock _l(mLock); 1523 1524 if (checkForNewParameters_l()) { 1525 mixBufferSize = mFrameCount * mFrameSize; 1526 // FIXME: Relaxed timing because of a certain device that can't meet latency 1527 // Should be reduced to 2x after the vendor fixes the driver issue 1528 maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1529 activeSleepTime = activeSleepTimeUs(); 1530 idleSleepTime = idleSleepTimeUs(); 1531 } 1532 1533 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1534 1535 // put audio hardware into standby after short delay 1536 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1537 mSuspended) { 1538 if (!mStandby) { 1539 LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1540 mOutput->standby(); 1541 mStandby = true; 1542 mBytesWritten = 0; 1543 } 1544 1545 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1546 // we're about to wait, flush the binder command buffer 1547 IPCThreadState::self()->flushCommands(); 1548 1549 if (exitPending()) break; 1550 1551 // wait until we have something to do... 1552 LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1553 mWaitWorkCV.wait(mLock); 1554 LOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1555 1556 if (mMasterMute == false) { 1557 char value[PROPERTY_VALUE_MAX]; 1558 property_get("ro.audio.silent", value, "0"); 1559 if (atoi(value)) { 1560 LOGD("Silence is golden"); 1561 setMasterMute(true); 1562 } 1563 } 1564 1565 standbyTime = systemTime() + kStandbyTimeInNsecs; 1566 sleepTime = idleSleepTime; 1567 continue; 1568 } 1569 } 1570 1571 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1572 1573 // prevent any changes in effect chain list and in each effect chain 1574 // during mixing and effect process as the audio buffers could be deleted 1575 // or modified if an effect is created or deleted 1576 lockEffectChains_l(effectChains); 1577 } 1578 1579 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1580 // mix buffers... 1581 mAudioMixer->process(); 1582 sleepTime = 0; 1583 standbyTime = systemTime() + kStandbyTimeInNsecs; 1584 //TODO: delay standby when effects have a tail 1585 } else { 1586 // If no tracks are ready, sleep once for the duration of an output 1587 // buffer size, then write 0s to the output 1588 if (sleepTime == 0) { 1589 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1590 sleepTime = activeSleepTime; 1591 } else { 1592 sleepTime = idleSleepTime; 1593 } 1594 } else if (mBytesWritten != 0 || 1595 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 1596 memset (mMixBuffer, 0, mixBufferSize); 1597 sleepTime = 0; 1598 LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 1599 } 1600 // TODO add standby time extension fct of effect tail 1601 } 1602 1603 if (mSuspended) { 1604 sleepTime = suspendSleepTimeUs(); 1605 } 1606 // sleepTime == 0 means we must write to audio hardware 1607 if (sleepTime == 0) { 1608 for (size_t i = 0; i < effectChains.size(); i ++) { 1609 effectChains[i]->process_l(); 1610 } 1611 // enable changes in effect chain 1612 unlockEffectChains(effectChains); 1613#ifdef LVMX 1614 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1615 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1616 LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize); 1617 } 1618#endif 1619 mLastWriteTime = systemTime(); 1620 mInWrite = true; 1621 mBytesWritten += mixBufferSize; 1622 1623 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 1624 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 1625 mNumWrites++; 1626 mInWrite = false; 1627 nsecs_t now = systemTime(); 1628 nsecs_t delta = now - mLastWriteTime; 1629 if (delta > maxPeriod) { 1630 mNumDelayedWrites++; 1631 if ((now - lastWarning) > kWarningThrottle) { 1632 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 1633 ns2ms(delta), mNumDelayedWrites, this); 1634 lastWarning = now; 1635 } 1636 if (mStandby) { 1637 longStandbyExit = true; 1638 } 1639 } 1640 mStandby = false; 1641 } else { 1642 // enable changes in effect chain 1643 unlockEffectChains(effectChains); 1644 usleep(sleepTime); 1645 } 1646 1647 // finally let go of all our tracks, without the lock held 1648 // since we can't guarantee the destructors won't acquire that 1649 // same lock. 1650 tracksToRemove.clear(); 1651 1652 // Effect chains will be actually deleted here if they were removed from 1653 // mEffectChains list during mixing or effects processing 1654 effectChains.clear(); 1655 } 1656 1657 if (!mStandby) { 1658 mOutput->standby(); 1659 } 1660 1661 LOGV("MixerThread %p exiting", this); 1662 return false; 1663} 1664 1665// prepareTracks_l() must be called with ThreadBase::mLock held 1666uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 1667{ 1668 1669 uint32_t mixerStatus = MIXER_IDLE; 1670 // find out which tracks need to be processed 1671 size_t count = activeTracks.size(); 1672 size_t mixedTracks = 0; 1673 size_t tracksWithEffect = 0; 1674 1675 float masterVolume = mMasterVolume; 1676 bool masterMute = mMasterMute; 1677 1678 if (masterMute) { 1679 masterVolume = 0; 1680 } 1681#ifdef LVMX 1682 bool tracksConnectedChanged = false; 1683 bool stateChanged = false; 1684 1685 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1686 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1687 { 1688 int activeTypes = 0; 1689 for (size_t i=0 ; i<count ; i++) { 1690 sp<Track> t = activeTracks[i].promote(); 1691 if (t == 0) continue; 1692 Track* const track = t.get(); 1693 int iTracktype=track->type(); 1694 activeTypes |= 1<<track->type(); 1695 } 1696 LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute); 1697 } 1698#endif 1699 // Delegate master volume control to effect in output mix effect chain if needed 1700 sp<EffectChain> chain = getEffectChain_l(AudioSystem::SESSION_OUTPUT_MIX); 1701 if (chain != 0) { 1702 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 1703 chain->setVolume_l(&v, &v); 1704 masterVolume = (float)((v + (1 << 23)) >> 24); 1705 chain.clear(); 1706 } 1707 1708 for (size_t i=0 ; i<count ; i++) { 1709 sp<Track> t = activeTracks[i].promote(); 1710 if (t == 0) continue; 1711 1712 Track* const track = t.get(); 1713 audio_track_cblk_t* cblk = track->cblk(); 1714 1715 // The first time a track is added we wait 1716 // for all its buffers to be filled before processing it 1717 mAudioMixer->setActiveTrack(track->name()); 1718 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 1719 !track->isPaused() && !track->isTerminated()) 1720 { 1721 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 1722 1723 mixedTracks++; 1724 1725 // track->mainBuffer() != mMixBuffer means there is an effect chain 1726 // connected to the track 1727 chain.clear(); 1728 if (track->mainBuffer() != mMixBuffer) { 1729 chain = getEffectChain_l(track->sessionId()); 1730 // Delegate volume control to effect in track effect chain if needed 1731 if (chain != 0) { 1732 tracksWithEffect++; 1733 } else { 1734 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 1735 track->name(), track->sessionId()); 1736 } 1737 } 1738 1739 1740 int param = AudioMixer::VOLUME; 1741 if (track->mFillingUpStatus == Track::FS_FILLED) { 1742 // no ramp for the first volume setting 1743 track->mFillingUpStatus = Track::FS_ACTIVE; 1744 if (track->mState == TrackBase::RESUMING) { 1745 track->mState = TrackBase::ACTIVE; 1746 param = AudioMixer::RAMP_VOLUME; 1747 } 1748 } else if (cblk->server != 0) { 1749 // If the track is stopped before the first frame was mixed, 1750 // do not apply ramp 1751 param = AudioMixer::RAMP_VOLUME; 1752 } 1753 1754 // compute volume for this track 1755 uint32_t vl, vr, va; 1756 if (track->isMuted() || track->isPausing() || 1757 mStreamTypes[track->type()].mute) { 1758 vl = vr = va = 0; 1759 if (track->isPausing()) { 1760 track->setPaused(); 1761 } 1762 } else { 1763 1764 // read original volumes with volume control 1765 float typeVolume = mStreamTypes[track->type()].volume; 1766#ifdef LVMX 1767 bool streamMute=false; 1768 // read the volume from the LivesVibes audio engine. 1769 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1770 { 1771 LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); 1772 if (streamMute) { 1773 typeVolume = 0; 1774 } 1775 } 1776#endif 1777 float v = masterVolume * typeVolume; 1778 vl = (uint32_t)(v * cblk->volume[0]) << 12; 1779 vr = (uint32_t)(v * cblk->volume[1]) << 12; 1780 1781 va = (uint32_t)(v * cblk->sendLevel); 1782 } 1783 // Delegate volume control to effect in track effect chain if needed 1784 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1785 // Do not ramp volume if volume is controlled by effect 1786 param = AudioMixer::VOLUME; 1787 track->mHasVolumeController = true; 1788 } else { 1789 // force no volume ramp when volume controller was just disabled or removed 1790 // from effect chain to avoid volume spike 1791 if (track->mHasVolumeController) { 1792 param = AudioMixer::VOLUME; 1793 } 1794 track->mHasVolumeController = false; 1795 } 1796 1797 // Convert volumes from 8.24 to 4.12 format 1798 int16_t left, right, aux; 1799 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1800 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1801 left = int16_t(v_clamped); 1802 v_clamped = (vr + (1 << 11)) >> 12; 1803 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1804 right = int16_t(v_clamped); 1805 1806 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 1807 aux = int16_t(va); 1808 1809#ifdef LVMX 1810 if ( tracksConnectedChanged || stateChanged ) 1811 { 1812 // only do the ramp when the volume is changed by the user / application 1813 param = AudioMixer::VOLUME; 1814 } 1815#endif 1816 1817 // XXX: these things DON'T need to be done each time 1818 mAudioMixer->setBufferProvider(track); 1819 mAudioMixer->enable(AudioMixer::MIXING); 1820 1821 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1822 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1823 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1824 mAudioMixer->setParameter( 1825 AudioMixer::TRACK, 1826 AudioMixer::FORMAT, (void *)track->format()); 1827 mAudioMixer->setParameter( 1828 AudioMixer::TRACK, 1829 AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); 1830 mAudioMixer->setParameter( 1831 AudioMixer::RESAMPLE, 1832 AudioMixer::SAMPLE_RATE, 1833 (void *)(cblk->sampleRate)); 1834 mAudioMixer->setParameter( 1835 AudioMixer::TRACK, 1836 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1837 mAudioMixer->setParameter( 1838 AudioMixer::TRACK, 1839 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1840 1841 // reset retry count 1842 track->mRetryCount = kMaxTrackRetries; 1843 mixerStatus = MIXER_TRACKS_READY; 1844 } else { 1845 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1846 if (track->isStopped()) { 1847 track->reset(); 1848 } 1849 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1850 // We have consumed all the buffers of this track. 1851 // Remove it from the list of active tracks. 1852 tracksToRemove->add(track); 1853 } else { 1854 // No buffers for this track. Give it a few chances to 1855 // fill a buffer, then remove it from active list. 1856 if (--(track->mRetryCount) <= 0) { 1857 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1858 tracksToRemove->add(track); 1859 } else if (mixerStatus != MIXER_TRACKS_READY) { 1860 mixerStatus = MIXER_TRACKS_ENABLED; 1861 } 1862 } 1863 mAudioMixer->disable(AudioMixer::MIXING); 1864 } 1865 } 1866 1867 // remove all the tracks that need to be... 1868 count = tracksToRemove->size(); 1869 if (UNLIKELY(count)) { 1870 for (size_t i=0 ; i<count ; i++) { 1871 const sp<Track>& track = tracksToRemove->itemAt(i); 1872 mActiveTracks.remove(track); 1873 if (track->mainBuffer() != mMixBuffer) { 1874 chain = getEffectChain_l(track->sessionId()); 1875 if (chain != 0) { 1876 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1877 chain->stopTrack(); 1878 } 1879 } 1880 if (track->isTerminated()) { 1881 mTracks.remove(track); 1882 deleteTrackName_l(track->mName); 1883 } 1884 } 1885 } 1886 1887 // mix buffer must be cleared if all tracks are connected to an 1888 // effect chain as in this case the mixer will not write to 1889 // mix buffer and track effects will accumulate into it 1890 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1891 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1892 } 1893 1894 return mixerStatus; 1895} 1896 1897void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1898{ 1899 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1900 this, streamType, mTracks.size()); 1901 Mutex::Autolock _l(mLock); 1902 1903 size_t size = mTracks.size(); 1904 for (size_t i = 0; i < size; i++) { 1905 sp<Track> t = mTracks[i]; 1906 if (t->type() == streamType) { 1907 t->mCblk->lock.lock(); 1908 t->mCblk->flags |= CBLK_INVALID_ON; 1909 t->mCblk->cv.signal(); 1910 t->mCblk->lock.unlock(); 1911 } 1912 } 1913} 1914 1915 1916// getTrackName_l() must be called with ThreadBase::mLock held 1917int AudioFlinger::MixerThread::getTrackName_l() 1918{ 1919 return mAudioMixer->getTrackName(); 1920} 1921 1922// deleteTrackName_l() must be called with ThreadBase::mLock held 1923void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1924{ 1925 LOGV("remove track (%d) and delete from mixer", name); 1926 mAudioMixer->deleteTrackName(name); 1927} 1928 1929// checkForNewParameters_l() must be called with ThreadBase::mLock held 1930bool AudioFlinger::MixerThread::checkForNewParameters_l() 1931{ 1932 bool reconfig = false; 1933 1934 while (!mNewParameters.isEmpty()) { 1935 status_t status = NO_ERROR; 1936 String8 keyValuePair = mNewParameters[0]; 1937 AudioParameter param = AudioParameter(keyValuePair); 1938 int value; 1939 1940 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1941 reconfig = true; 1942 } 1943 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1944 if (value != AudioSystem::PCM_16_BIT) { 1945 status = BAD_VALUE; 1946 } else { 1947 reconfig = true; 1948 } 1949 } 1950 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1951 if (value != AudioSystem::CHANNEL_OUT_STEREO) { 1952 status = BAD_VALUE; 1953 } else { 1954 reconfig = true; 1955 } 1956 } 1957 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1958 // do not accept frame count changes if tracks are open as the track buffer 1959 // size depends on frame count and correct behavior would not be garantied 1960 // if frame count is changed after track creation 1961 if (!mTracks.isEmpty()) { 1962 status = INVALID_OPERATION; 1963 } else { 1964 reconfig = true; 1965 } 1966 } 1967 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1968 // forward device change to effects that have requested to be 1969 // aware of attached audio device. 1970 mDevice = (uint32_t)value; 1971 for (size_t i = 0; i < mEffectChains.size(); i++) { 1972 mEffectChains[i]->setDevice_l(mDevice); 1973 } 1974 } 1975 1976 if (status == NO_ERROR) { 1977 status = mOutput->setParameters(keyValuePair); 1978 if (!mStandby && status == INVALID_OPERATION) { 1979 mOutput->standby(); 1980 mStandby = true; 1981 mBytesWritten = 0; 1982 status = mOutput->setParameters(keyValuePair); 1983 } 1984 if (status == NO_ERROR && reconfig) { 1985 delete mAudioMixer; 1986 readOutputParameters(); 1987 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1988 for (size_t i = 0; i < mTracks.size() ; i++) { 1989 int name = getTrackName_l(); 1990 if (name < 0) break; 1991 mTracks[i]->mName = name; 1992 // limit track sample rate to 2 x new output sample rate 1993 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 1994 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 1995 } 1996 } 1997 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 1998 } 1999 } 2000 2001 mNewParameters.removeAt(0); 2002 2003 mParamStatus = status; 2004 mParamCond.signal(); 2005 mWaitWorkCV.wait(mLock); 2006 } 2007 return reconfig; 2008} 2009 2010status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2011{ 2012 const size_t SIZE = 256; 2013 char buffer[SIZE]; 2014 String8 result; 2015 2016 PlaybackThread::dumpInternals(fd, args); 2017 2018 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2019 result.append(buffer); 2020 write(fd, result.string(), result.size()); 2021 return NO_ERROR; 2022} 2023 2024uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 2025{ 2026 return (uint32_t)(mOutput->latency() * 1000) / 2; 2027} 2028 2029uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2030{ 2031 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2032} 2033 2034uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2035{ 2036 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2037} 2038 2039// ---------------------------------------------------------------------------- 2040AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2041 : PlaybackThread(audioFlinger, output, id, device) 2042{ 2043 mType = PlaybackThread::DIRECT; 2044} 2045 2046AudioFlinger::DirectOutputThread::~DirectOutputThread() 2047{ 2048} 2049 2050 2051static inline int16_t clamp16(int32_t sample) 2052{ 2053 if ((sample>>15) ^ (sample>>31)) 2054 sample = 0x7FFF ^ (sample>>31); 2055 return sample; 2056} 2057 2058static inline 2059int32_t mul(int16_t in, int16_t v) 2060{ 2061#if defined(__arm__) && !defined(__thumb__) 2062 int32_t out; 2063 asm( "smulbb %[out], %[in], %[v] \n" 2064 : [out]"=r"(out) 2065 : [in]"%r"(in), [v]"r"(v) 2066 : ); 2067 return out; 2068#else 2069 return in * int32_t(v); 2070#endif 2071} 2072 2073void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2074{ 2075 // Do not apply volume on compressed audio 2076 if (!AudioSystem::isLinearPCM(mFormat)) { 2077 return; 2078 } 2079 2080 // convert to signed 16 bit before volume calculation 2081 if (mFormat == AudioSystem::PCM_8_BIT) { 2082 size_t count = mFrameCount * mChannelCount; 2083 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2084 int16_t *dst = mMixBuffer + count-1; 2085 while(count--) { 2086 *dst-- = (int16_t)(*src--^0x80) << 8; 2087 } 2088 } 2089 2090 size_t frameCount = mFrameCount; 2091 int16_t *out = mMixBuffer; 2092 if (ramp) { 2093 if (mChannelCount == 1) { 2094 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2095 int32_t vlInc = d / (int32_t)frameCount; 2096 int32_t vl = ((int32_t)mLeftVolShort << 16); 2097 do { 2098 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2099 out++; 2100 vl += vlInc; 2101 } while (--frameCount); 2102 2103 } else { 2104 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2105 int32_t vlInc = d / (int32_t)frameCount; 2106 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2107 int32_t vrInc = d / (int32_t)frameCount; 2108 int32_t vl = ((int32_t)mLeftVolShort << 16); 2109 int32_t vr = ((int32_t)mRightVolShort << 16); 2110 do { 2111 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2112 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2113 out += 2; 2114 vl += vlInc; 2115 vr += vrInc; 2116 } while (--frameCount); 2117 } 2118 } else { 2119 if (mChannelCount == 1) { 2120 do { 2121 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2122 out++; 2123 } while (--frameCount); 2124 } else { 2125 do { 2126 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2127 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2128 out += 2; 2129 } while (--frameCount); 2130 } 2131 } 2132 2133 // convert back to unsigned 8 bit after volume calculation 2134 if (mFormat == AudioSystem::PCM_8_BIT) { 2135 size_t count = mFrameCount * mChannelCount; 2136 int16_t *src = mMixBuffer; 2137 uint8_t *dst = (uint8_t *)mMixBuffer; 2138 while(count--) { 2139 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2140 } 2141 } 2142 2143 mLeftVolShort = leftVol; 2144 mRightVolShort = rightVol; 2145} 2146 2147bool AudioFlinger::DirectOutputThread::threadLoop() 2148{ 2149 uint32_t mixerStatus = MIXER_IDLE; 2150 sp<Track> trackToRemove; 2151 sp<Track> activeTrack; 2152 nsecs_t standbyTime = systemTime(); 2153 int8_t *curBuf; 2154 size_t mixBufferSize = mFrameCount*mFrameSize; 2155 uint32_t activeSleepTime = activeSleepTimeUs(); 2156 uint32_t idleSleepTime = idleSleepTimeUs(); 2157 uint32_t sleepTime = idleSleepTime; 2158 // use shorter standby delay as on normal output to release 2159 // hardware resources as soon as possible 2160 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2161 2162 while (!exitPending()) 2163 { 2164 bool rampVolume; 2165 uint16_t leftVol; 2166 uint16_t rightVol; 2167 Vector< sp<EffectChain> > effectChains; 2168 2169 processConfigEvents(); 2170 2171 mixerStatus = MIXER_IDLE; 2172 2173 { // scope for the mLock 2174 2175 Mutex::Autolock _l(mLock); 2176 2177 if (checkForNewParameters_l()) { 2178 mixBufferSize = mFrameCount*mFrameSize; 2179 activeSleepTime = activeSleepTimeUs(); 2180 idleSleepTime = idleSleepTimeUs(); 2181 standbyDelay = microseconds(activeSleepTime*2); 2182 } 2183 2184 // put audio hardware into standby after short delay 2185 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2186 mSuspended) { 2187 // wait until we have something to do... 2188 if (!mStandby) { 2189 LOGV("Audio hardware entering standby, mixer %p\n", this); 2190 mOutput->standby(); 2191 mStandby = true; 2192 mBytesWritten = 0; 2193 } 2194 2195 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2196 // we're about to wait, flush the binder command buffer 2197 IPCThreadState::self()->flushCommands(); 2198 2199 if (exitPending()) break; 2200 2201 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2202 mWaitWorkCV.wait(mLock); 2203 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2204 2205 if (mMasterMute == false) { 2206 char value[PROPERTY_VALUE_MAX]; 2207 property_get("ro.audio.silent", value, "0"); 2208 if (atoi(value)) { 2209 LOGD("Silence is golden"); 2210 setMasterMute(true); 2211 } 2212 } 2213 2214 standbyTime = systemTime() + standbyDelay; 2215 sleepTime = idleSleepTime; 2216 continue; 2217 } 2218 } 2219 2220 effectChains = mEffectChains; 2221 2222 // find out which tracks need to be processed 2223 if (mActiveTracks.size() != 0) { 2224 sp<Track> t = mActiveTracks[0].promote(); 2225 if (t == 0) continue; 2226 2227 Track* const track = t.get(); 2228 audio_track_cblk_t* cblk = track->cblk(); 2229 2230 // The first time a track is added we wait 2231 // for all its buffers to be filled before processing it 2232 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 2233 !track->isPaused() && !track->isTerminated()) 2234 { 2235 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2236 2237 if (track->mFillingUpStatus == Track::FS_FILLED) { 2238 track->mFillingUpStatus = Track::FS_ACTIVE; 2239 mLeftVolFloat = mRightVolFloat = 0; 2240 mLeftVolShort = mRightVolShort = 0; 2241 if (track->mState == TrackBase::RESUMING) { 2242 track->mState = TrackBase::ACTIVE; 2243 rampVolume = true; 2244 } 2245 } else if (cblk->server != 0) { 2246 // If the track is stopped before the first frame was mixed, 2247 // do not apply ramp 2248 rampVolume = true; 2249 } 2250 // compute volume for this track 2251 float left, right; 2252 if (track->isMuted() || mMasterMute || track->isPausing() || 2253 mStreamTypes[track->type()].mute) { 2254 left = right = 0; 2255 if (track->isPausing()) { 2256 track->setPaused(); 2257 } 2258 } else { 2259 float typeVolume = mStreamTypes[track->type()].volume; 2260 float v = mMasterVolume * typeVolume; 2261 float v_clamped = v * cblk->volume[0]; 2262 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2263 left = v_clamped/MAX_GAIN; 2264 v_clamped = v * cblk->volume[1]; 2265 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2266 right = v_clamped/MAX_GAIN; 2267 } 2268 2269 if (left != mLeftVolFloat || right != mRightVolFloat) { 2270 mLeftVolFloat = left; 2271 mRightVolFloat = right; 2272 2273 // If audio HAL implements volume control, 2274 // force software volume to nominal value 2275 if (mOutput->setVolume(left, right) == NO_ERROR) { 2276 left = 1.0f; 2277 right = 1.0f; 2278 } 2279 2280 // Convert volumes from float to 8.24 2281 uint32_t vl = (uint32_t)(left * (1 << 24)); 2282 uint32_t vr = (uint32_t)(right * (1 << 24)); 2283 2284 // Delegate volume control to effect in track effect chain if needed 2285 // only one effect chain can be present on DirectOutputThread, so if 2286 // there is one, the track is connected to it 2287 if (!effectChains.isEmpty()) { 2288 // Do not ramp volume if volume is controlled by effect 2289 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2290 rampVolume = false; 2291 } 2292 } 2293 2294 // Convert volumes from 8.24 to 4.12 format 2295 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2296 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2297 leftVol = (uint16_t)v_clamped; 2298 v_clamped = (vr + (1 << 11)) >> 12; 2299 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2300 rightVol = (uint16_t)v_clamped; 2301 } else { 2302 leftVol = mLeftVolShort; 2303 rightVol = mRightVolShort; 2304 rampVolume = false; 2305 } 2306 2307 // reset retry count 2308 track->mRetryCount = kMaxTrackRetriesDirect; 2309 activeTrack = t; 2310 mixerStatus = MIXER_TRACKS_READY; 2311 } else { 2312 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2313 if (track->isStopped()) { 2314 track->reset(); 2315 } 2316 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2317 // We have consumed all the buffers of this track. 2318 // Remove it from the list of active tracks. 2319 trackToRemove = track; 2320 } else { 2321 // No buffers for this track. Give it a few chances to 2322 // fill a buffer, then remove it from active list. 2323 if (--(track->mRetryCount) <= 0) { 2324 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2325 trackToRemove = track; 2326 } else { 2327 mixerStatus = MIXER_TRACKS_ENABLED; 2328 } 2329 } 2330 } 2331 } 2332 2333 // remove all the tracks that need to be... 2334 if (UNLIKELY(trackToRemove != 0)) { 2335 mActiveTracks.remove(trackToRemove); 2336 if (!effectChains.isEmpty()) { 2337 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2338 trackToRemove->sessionId()); 2339 effectChains[0]->stopTrack(); 2340 } 2341 if (trackToRemove->isTerminated()) { 2342 mTracks.remove(trackToRemove); 2343 deleteTrackName_l(trackToRemove->mName); 2344 } 2345 } 2346 2347 lockEffectChains_l(effectChains); 2348 } 2349 2350 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2351 AudioBufferProvider::Buffer buffer; 2352 size_t frameCount = mFrameCount; 2353 curBuf = (int8_t *)mMixBuffer; 2354 // output audio to hardware 2355 while (frameCount) { 2356 buffer.frameCount = frameCount; 2357 activeTrack->getNextBuffer(&buffer); 2358 if (UNLIKELY(buffer.raw == 0)) { 2359 memset(curBuf, 0, frameCount * mFrameSize); 2360 break; 2361 } 2362 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2363 frameCount -= buffer.frameCount; 2364 curBuf += buffer.frameCount * mFrameSize; 2365 activeTrack->releaseBuffer(&buffer); 2366 } 2367 sleepTime = 0; 2368 standbyTime = systemTime() + standbyDelay; 2369 } else { 2370 if (sleepTime == 0) { 2371 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2372 sleepTime = activeSleepTime; 2373 } else { 2374 sleepTime = idleSleepTime; 2375 } 2376 } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { 2377 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2378 sleepTime = 0; 2379 } 2380 } 2381 2382 if (mSuspended) { 2383 sleepTime = suspendSleepTimeUs(); 2384 } 2385 // sleepTime == 0 means we must write to audio hardware 2386 if (sleepTime == 0) { 2387 if (mixerStatus == MIXER_TRACKS_READY) { 2388 applyVolume(leftVol, rightVol, rampVolume); 2389 } 2390 for (size_t i = 0; i < effectChains.size(); i ++) { 2391 effectChains[i]->process_l(); 2392 } 2393 unlockEffectChains(effectChains); 2394 2395 mLastWriteTime = systemTime(); 2396 mInWrite = true; 2397 mBytesWritten += mixBufferSize; 2398 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 2399 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2400 mNumWrites++; 2401 mInWrite = false; 2402 mStandby = false; 2403 } else { 2404 unlockEffectChains(effectChains); 2405 usleep(sleepTime); 2406 } 2407 2408 // finally let go of removed track, without the lock held 2409 // since we can't guarantee the destructors won't acquire that 2410 // same lock. 2411 trackToRemove.clear(); 2412 activeTrack.clear(); 2413 2414 // Effect chains will be actually deleted here if they were removed from 2415 // mEffectChains list during mixing or effects processing 2416 effectChains.clear(); 2417 } 2418 2419 if (!mStandby) { 2420 mOutput->standby(); 2421 } 2422 2423 LOGV("DirectOutputThread %p exiting", this); 2424 return false; 2425} 2426 2427// getTrackName_l() must be called with ThreadBase::mLock held 2428int AudioFlinger::DirectOutputThread::getTrackName_l() 2429{ 2430 return 0; 2431} 2432 2433// deleteTrackName_l() must be called with ThreadBase::mLock held 2434void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2435{ 2436} 2437 2438// checkForNewParameters_l() must be called with ThreadBase::mLock held 2439bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2440{ 2441 bool reconfig = false; 2442 2443 while (!mNewParameters.isEmpty()) { 2444 status_t status = NO_ERROR; 2445 String8 keyValuePair = mNewParameters[0]; 2446 AudioParameter param = AudioParameter(keyValuePair); 2447 int value; 2448 2449 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2450 // do not accept frame count changes if tracks are open as the track buffer 2451 // size depends on frame count and correct behavior would not be garantied 2452 // if frame count is changed after track creation 2453 if (!mTracks.isEmpty()) { 2454 status = INVALID_OPERATION; 2455 } else { 2456 reconfig = true; 2457 } 2458 } 2459 if (status == NO_ERROR) { 2460 status = mOutput->setParameters(keyValuePair); 2461 if (!mStandby && status == INVALID_OPERATION) { 2462 mOutput->standby(); 2463 mStandby = true; 2464 mBytesWritten = 0; 2465 status = mOutput->setParameters(keyValuePair); 2466 } 2467 if (status == NO_ERROR && reconfig) { 2468 readOutputParameters(); 2469 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2470 } 2471 } 2472 2473 mNewParameters.removeAt(0); 2474 2475 mParamStatus = status; 2476 mParamCond.signal(); 2477 mWaitWorkCV.wait(mLock); 2478 } 2479 return reconfig; 2480} 2481 2482uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2483{ 2484 uint32_t time; 2485 if (AudioSystem::isLinearPCM(mFormat)) { 2486 time = (uint32_t)(mOutput->latency() * 1000) / 2; 2487 } else { 2488 time = 10000; 2489 } 2490 return time; 2491} 2492 2493uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2494{ 2495 uint32_t time; 2496 if (AudioSystem::isLinearPCM(mFormat)) { 2497 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2498 } else { 2499 time = 10000; 2500 } 2501 return time; 2502} 2503 2504uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2505{ 2506 uint32_t time; 2507 if (AudioSystem::isLinearPCM(mFormat)) { 2508 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2509 } else { 2510 time = 10000; 2511 } 2512 return time; 2513} 2514 2515 2516// ---------------------------------------------------------------------------- 2517 2518AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2519 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2520{ 2521 mType = PlaybackThread::DUPLICATING; 2522 addOutputTrack(mainThread); 2523} 2524 2525AudioFlinger::DuplicatingThread::~DuplicatingThread() 2526{ 2527 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2528 mOutputTracks[i]->destroy(); 2529 } 2530 mOutputTracks.clear(); 2531} 2532 2533bool AudioFlinger::DuplicatingThread::threadLoop() 2534{ 2535 Vector< sp<Track> > tracksToRemove; 2536 uint32_t mixerStatus = MIXER_IDLE; 2537 nsecs_t standbyTime = systemTime(); 2538 size_t mixBufferSize = mFrameCount*mFrameSize; 2539 SortedVector< sp<OutputTrack> > outputTracks; 2540 uint32_t writeFrames = 0; 2541 uint32_t activeSleepTime = activeSleepTimeUs(); 2542 uint32_t idleSleepTime = idleSleepTimeUs(); 2543 uint32_t sleepTime = idleSleepTime; 2544 Vector< sp<EffectChain> > effectChains; 2545 2546 while (!exitPending()) 2547 { 2548 processConfigEvents(); 2549 2550 mixerStatus = MIXER_IDLE; 2551 { // scope for the mLock 2552 2553 Mutex::Autolock _l(mLock); 2554 2555 if (checkForNewParameters_l()) { 2556 mixBufferSize = mFrameCount*mFrameSize; 2557 updateWaitTime(); 2558 activeSleepTime = activeSleepTimeUs(); 2559 idleSleepTime = idleSleepTimeUs(); 2560 } 2561 2562 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2563 2564 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2565 outputTracks.add(mOutputTracks[i]); 2566 } 2567 2568 // put audio hardware into standby after short delay 2569 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2570 mSuspended) { 2571 if (!mStandby) { 2572 for (size_t i = 0; i < outputTracks.size(); i++) { 2573 outputTracks[i]->stop(); 2574 } 2575 mStandby = true; 2576 mBytesWritten = 0; 2577 } 2578 2579 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2580 // we're about to wait, flush the binder command buffer 2581 IPCThreadState::self()->flushCommands(); 2582 outputTracks.clear(); 2583 2584 if (exitPending()) break; 2585 2586 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2587 mWaitWorkCV.wait(mLock); 2588 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2589 if (mMasterMute == false) { 2590 char value[PROPERTY_VALUE_MAX]; 2591 property_get("ro.audio.silent", value, "0"); 2592 if (atoi(value)) { 2593 LOGD("Silence is golden"); 2594 setMasterMute(true); 2595 } 2596 } 2597 2598 standbyTime = systemTime() + kStandbyTimeInNsecs; 2599 sleepTime = idleSleepTime; 2600 continue; 2601 } 2602 } 2603 2604 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2605 2606 // prevent any changes in effect chain list and in each effect chain 2607 // during mixing and effect process as the audio buffers could be deleted 2608 // or modified if an effect is created or deleted 2609 lockEffectChains_l(effectChains); 2610 } 2611 2612 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2613 // mix buffers... 2614 if (outputsReady(outputTracks)) { 2615 mAudioMixer->process(); 2616 } else { 2617 memset(mMixBuffer, 0, mixBufferSize); 2618 } 2619 sleepTime = 0; 2620 writeFrames = mFrameCount; 2621 } else { 2622 if (sleepTime == 0) { 2623 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2624 sleepTime = activeSleepTime; 2625 } else { 2626 sleepTime = idleSleepTime; 2627 } 2628 } else if (mBytesWritten != 0) { 2629 // flush remaining overflow buffers in output tracks 2630 for (size_t i = 0; i < outputTracks.size(); i++) { 2631 if (outputTracks[i]->isActive()) { 2632 sleepTime = 0; 2633 writeFrames = 0; 2634 memset(mMixBuffer, 0, mixBufferSize); 2635 break; 2636 } 2637 } 2638 } 2639 } 2640 2641 if (mSuspended) { 2642 sleepTime = suspendSleepTimeUs(); 2643 } 2644 // sleepTime == 0 means we must write to audio hardware 2645 if (sleepTime == 0) { 2646 for (size_t i = 0; i < effectChains.size(); i ++) { 2647 effectChains[i]->process_l(); 2648 } 2649 // enable changes in effect chain 2650 unlockEffectChains(effectChains); 2651 2652 standbyTime = systemTime() + kStandbyTimeInNsecs; 2653 for (size_t i = 0; i < outputTracks.size(); i++) { 2654 outputTracks[i]->write(mMixBuffer, writeFrames); 2655 } 2656 mStandby = false; 2657 mBytesWritten += mixBufferSize; 2658 } else { 2659 // enable changes in effect chain 2660 unlockEffectChains(effectChains); 2661 usleep(sleepTime); 2662 } 2663 2664 // finally let go of all our tracks, without the lock held 2665 // since we can't guarantee the destructors won't acquire that 2666 // same lock. 2667 tracksToRemove.clear(); 2668 outputTracks.clear(); 2669 2670 // Effect chains will be actually deleted here if they were removed from 2671 // mEffectChains list during mixing or effects processing 2672 effectChains.clear(); 2673 } 2674 2675 return false; 2676} 2677 2678void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2679{ 2680 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2681 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2682 this, 2683 mSampleRate, 2684 mFormat, 2685 mChannelCount, 2686 frameCount); 2687 if (outputTrack->cblk() != NULL) { 2688 thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); 2689 mOutputTracks.add(outputTrack); 2690 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2691 updateWaitTime(); 2692 } 2693} 2694 2695void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2696{ 2697 Mutex::Autolock _l(mLock); 2698 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2699 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2700 mOutputTracks[i]->destroy(); 2701 mOutputTracks.removeAt(i); 2702 updateWaitTime(); 2703 return; 2704 } 2705 } 2706 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2707} 2708 2709void AudioFlinger::DuplicatingThread::updateWaitTime() 2710{ 2711 mWaitTimeMs = UINT_MAX; 2712 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2713 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2714 if (strong != NULL) { 2715 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2716 if (waitTimeMs < mWaitTimeMs) { 2717 mWaitTimeMs = waitTimeMs; 2718 } 2719 } 2720 } 2721} 2722 2723 2724bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2725{ 2726 for (size_t i = 0; i < outputTracks.size(); i++) { 2727 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2728 if (thread == 0) { 2729 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2730 return false; 2731 } 2732 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2733 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2734 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2735 return false; 2736 } 2737 } 2738 return true; 2739} 2740 2741uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2742{ 2743 return (mWaitTimeMs * 1000) / 2; 2744} 2745 2746// ---------------------------------------------------------------------------- 2747 2748// TrackBase constructor must be called with AudioFlinger::mLock held 2749AudioFlinger::ThreadBase::TrackBase::TrackBase( 2750 const wp<ThreadBase>& thread, 2751 const sp<Client>& client, 2752 uint32_t sampleRate, 2753 int format, 2754 int channelCount, 2755 int frameCount, 2756 uint32_t flags, 2757 const sp<IMemory>& sharedBuffer, 2758 int sessionId) 2759 : RefBase(), 2760 mThread(thread), 2761 mClient(client), 2762 mCblk(0), 2763 mFrameCount(0), 2764 mState(IDLE), 2765 mClientTid(-1), 2766 mFormat(format), 2767 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2768 mSessionId(sessionId) 2769{ 2770 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2771 2772 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2773 size_t size = sizeof(audio_track_cblk_t); 2774 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2775 if (sharedBuffer == 0) { 2776 size += bufferSize; 2777 } 2778 2779 if (client != NULL) { 2780 mCblkMemory = client->heap()->allocate(size); 2781 if (mCblkMemory != 0) { 2782 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2783 if (mCblk) { // construct the shared structure in-place. 2784 new(mCblk) audio_track_cblk_t(); 2785 // clear all buffers 2786 mCblk->frameCount = frameCount; 2787 mCblk->sampleRate = sampleRate; 2788 mCblk->channelCount = (uint8_t)channelCount; 2789 if (sharedBuffer == 0) { 2790 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2791 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2792 // Force underrun condition to avoid false underrun callback until first data is 2793 // written to buffer 2794 mCblk->flags = CBLK_UNDERRUN_ON; 2795 } else { 2796 mBuffer = sharedBuffer->pointer(); 2797 } 2798 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2799 } 2800 } else { 2801 LOGE("not enough memory for AudioTrack size=%u", size); 2802 client->heap()->dump("AudioTrack"); 2803 return; 2804 } 2805 } else { 2806 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2807 if (mCblk) { // construct the shared structure in-place. 2808 new(mCblk) audio_track_cblk_t(); 2809 // clear all buffers 2810 mCblk->frameCount = frameCount; 2811 mCblk->sampleRate = sampleRate; 2812 mCblk->channelCount = (uint8_t)channelCount; 2813 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2814 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2815 // Force underrun condition to avoid false underrun callback until first data is 2816 // written to buffer 2817 mCblk->flags = CBLK_UNDERRUN_ON; 2818 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2819 } 2820 } 2821} 2822 2823AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2824{ 2825 if (mCblk) { 2826 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2827 if (mClient == NULL) { 2828 delete mCblk; 2829 } 2830 } 2831 mCblkMemory.clear(); // and free the shared memory 2832 if (mClient != NULL) { 2833 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2834 mClient.clear(); 2835 } 2836} 2837 2838void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2839{ 2840 buffer->raw = 0; 2841 mFrameCount = buffer->frameCount; 2842 step(); 2843 buffer->frameCount = 0; 2844} 2845 2846bool AudioFlinger::ThreadBase::TrackBase::step() { 2847 bool result; 2848 audio_track_cblk_t* cblk = this->cblk(); 2849 2850 result = cblk->stepServer(mFrameCount); 2851 if (!result) { 2852 LOGV("stepServer failed acquiring cblk mutex"); 2853 mFlags |= STEPSERVER_FAILED; 2854 } 2855 return result; 2856} 2857 2858void AudioFlinger::ThreadBase::TrackBase::reset() { 2859 audio_track_cblk_t* cblk = this->cblk(); 2860 2861 cblk->user = 0; 2862 cblk->server = 0; 2863 cblk->userBase = 0; 2864 cblk->serverBase = 0; 2865 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2866 LOGV("TrackBase::reset"); 2867} 2868 2869sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2870{ 2871 return mCblkMemory; 2872} 2873 2874int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2875 return (int)mCblk->sampleRate; 2876} 2877 2878int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2879 return (int)mCblk->channelCount; 2880} 2881 2882void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2883 audio_track_cblk_t* cblk = this->cblk(); 2884 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2885 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2886 2887 // Check validity of returned pointer in case the track control block would have been corrupted. 2888 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2889 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2890 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2891 server %d, serverBase %d, user %d, userBase %d, channelCount %d", 2892 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2893 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); 2894 return 0; 2895 } 2896 2897 return bufferStart; 2898} 2899 2900// ---------------------------------------------------------------------------- 2901 2902// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2903AudioFlinger::PlaybackThread::Track::Track( 2904 const wp<ThreadBase>& thread, 2905 const sp<Client>& client, 2906 int streamType, 2907 uint32_t sampleRate, 2908 int format, 2909 int channelCount, 2910 int frameCount, 2911 const sp<IMemory>& sharedBuffer, 2912 int sessionId) 2913 : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), 2914 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 2915 mAuxEffectId(0), mHasVolumeController(false) 2916{ 2917 if (mCblk != NULL) { 2918 sp<ThreadBase> baseThread = thread.promote(); 2919 if (baseThread != 0) { 2920 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2921 mName = playbackThread->getTrackName_l(); 2922 mMainBuffer = playbackThread->mixBuffer(); 2923 } 2924 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2925 if (mName < 0) { 2926 LOGE("no more track names available"); 2927 } 2928 mVolume[0] = 1.0f; 2929 mVolume[1] = 1.0f; 2930 mStreamType = streamType; 2931 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2932 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2933 mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); 2934 } 2935} 2936 2937AudioFlinger::PlaybackThread::Track::~Track() 2938{ 2939 LOGV("PlaybackThread::Track destructor"); 2940 sp<ThreadBase> thread = mThread.promote(); 2941 if (thread != 0) { 2942 Mutex::Autolock _l(thread->mLock); 2943 mState = TERMINATED; 2944 } 2945} 2946 2947void AudioFlinger::PlaybackThread::Track::destroy() 2948{ 2949 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2950 // by removing it from mTracks vector, so there is a risk that this Tracks's 2951 // desctructor is called. As the destructor needs to lock mLock, 2952 // we must acquire a strong reference on this Track before locking mLock 2953 // here so that the destructor is called only when exiting this function. 2954 // On the other hand, as long as Track::destroy() is only called by 2955 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2956 // this Track with its member mTrack. 2957 sp<Track> keep(this); 2958 { // scope for mLock 2959 sp<ThreadBase> thread = mThread.promote(); 2960 if (thread != 0) { 2961 if (!isOutputTrack()) { 2962 if (mState == ACTIVE || mState == RESUMING) { 2963 AudioSystem::stopOutput(thread->id(), 2964 (AudioSystem::stream_type)mStreamType, 2965 mSessionId); 2966 } 2967 AudioSystem::releaseOutput(thread->id()); 2968 } 2969 Mutex::Autolock _l(thread->mLock); 2970 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2971 playbackThread->destroyTrack_l(this); 2972 } 2973 } 2974} 2975 2976void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 2977{ 2978 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", 2979 mName - AudioMixer::TRACK0, 2980 (mClient == NULL) ? getpid() : mClient->pid(), 2981 mStreamType, 2982 mFormat, 2983 mCblk->channelCount, 2984 mSessionId, 2985 mFrameCount, 2986 mState, 2987 mMute, 2988 mFillingUpStatus, 2989 mCblk->sampleRate, 2990 mCblk->volume[0], 2991 mCblk->volume[1], 2992 mCblk->server, 2993 mCblk->user, 2994 (int)mMainBuffer, 2995 (int)mAuxBuffer); 2996} 2997 2998status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 2999{ 3000 audio_track_cblk_t* cblk = this->cblk(); 3001 uint32_t framesReady; 3002 uint32_t framesReq = buffer->frameCount; 3003 3004 // Check if last stepServer failed, try to step now 3005 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3006 if (!step()) goto getNextBuffer_exit; 3007 LOGV("stepServer recovered"); 3008 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3009 } 3010 3011 framesReady = cblk->framesReady(); 3012 3013 if (LIKELY(framesReady)) { 3014 uint32_t s = cblk->server; 3015 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3016 3017 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3018 if (framesReq > framesReady) { 3019 framesReq = framesReady; 3020 } 3021 if (s + framesReq > bufferEnd) { 3022 framesReq = bufferEnd - s; 3023 } 3024 3025 buffer->raw = getBuffer(s, framesReq); 3026 if (buffer->raw == 0) goto getNextBuffer_exit; 3027 3028 buffer->frameCount = framesReq; 3029 return NO_ERROR; 3030 } 3031 3032getNextBuffer_exit: 3033 buffer->raw = 0; 3034 buffer->frameCount = 0; 3035 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3036 return NOT_ENOUGH_DATA; 3037} 3038 3039bool AudioFlinger::PlaybackThread::Track::isReady() const { 3040 if (mFillingUpStatus != FS_FILLING) return true; 3041 3042 if (mCblk->framesReady() >= mCblk->frameCount || 3043 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3044 mFillingUpStatus = FS_FILLED; 3045 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3046 return true; 3047 } 3048 return false; 3049} 3050 3051status_t AudioFlinger::PlaybackThread::Track::start() 3052{ 3053 status_t status = NO_ERROR; 3054 LOGV("start(%d), calling thread %d session %d", 3055 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3056 sp<ThreadBase> thread = mThread.promote(); 3057 if (thread != 0) { 3058 Mutex::Autolock _l(thread->mLock); 3059 int state = mState; 3060 // here the track could be either new, or restarted 3061 // in both cases "unstop" the track 3062 if (mState == PAUSED) { 3063 mState = TrackBase::RESUMING; 3064 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3065 } else { 3066 mState = TrackBase::ACTIVE; 3067 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3068 } 3069 3070 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3071 thread->mLock.unlock(); 3072 status = AudioSystem::startOutput(thread->id(), 3073 (AudioSystem::stream_type)mStreamType, 3074 mSessionId); 3075 thread->mLock.lock(); 3076 } 3077 if (status == NO_ERROR) { 3078 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3079 playbackThread->addTrack_l(this); 3080 } else { 3081 mState = state; 3082 } 3083 } else { 3084 status = BAD_VALUE; 3085 } 3086 return status; 3087} 3088 3089void AudioFlinger::PlaybackThread::Track::stop() 3090{ 3091 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3092 sp<ThreadBase> thread = mThread.promote(); 3093 if (thread != 0) { 3094 Mutex::Autolock _l(thread->mLock); 3095 int state = mState; 3096 if (mState > STOPPED) { 3097 mState = STOPPED; 3098 // If the track is not active (PAUSED and buffers full), flush buffers 3099 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3100 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3101 reset(); 3102 } 3103 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3104 } 3105 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3106 thread->mLock.unlock(); 3107 AudioSystem::stopOutput(thread->id(), 3108 (AudioSystem::stream_type)mStreamType, 3109 mSessionId); 3110 thread->mLock.lock(); 3111 } 3112 } 3113} 3114 3115void AudioFlinger::PlaybackThread::Track::pause() 3116{ 3117 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3118 sp<ThreadBase> thread = mThread.promote(); 3119 if (thread != 0) { 3120 Mutex::Autolock _l(thread->mLock); 3121 if (mState == ACTIVE || mState == RESUMING) { 3122 mState = PAUSING; 3123 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3124 if (!isOutputTrack()) { 3125 thread->mLock.unlock(); 3126 AudioSystem::stopOutput(thread->id(), 3127 (AudioSystem::stream_type)mStreamType, 3128 mSessionId); 3129 thread->mLock.lock(); 3130 } 3131 } 3132 } 3133} 3134 3135void AudioFlinger::PlaybackThread::Track::flush() 3136{ 3137 LOGV("flush(%d)", mName); 3138 sp<ThreadBase> thread = mThread.promote(); 3139 if (thread != 0) { 3140 Mutex::Autolock _l(thread->mLock); 3141 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3142 return; 3143 } 3144 // No point remaining in PAUSED state after a flush => go to 3145 // STOPPED state 3146 mState = STOPPED; 3147 3148 mCblk->lock.lock(); 3149 // NOTE: reset() will reset cblk->user and cblk->server with 3150 // the risk that at the same time, the AudioMixer is trying to read 3151 // data. In this case, getNextBuffer() would return a NULL pointer 3152 // as audio buffer => the AudioMixer code MUST always test that pointer 3153 // returned by getNextBuffer() is not NULL! 3154 reset(); 3155 mCblk->lock.unlock(); 3156 } 3157} 3158 3159void AudioFlinger::PlaybackThread::Track::reset() 3160{ 3161 // Do not reset twice to avoid discarding data written just after a flush and before 3162 // the audioflinger thread detects the track is stopped. 3163 if (!mResetDone) { 3164 TrackBase::reset(); 3165 // Force underrun condition to avoid false underrun callback until first data is 3166 // written to buffer 3167 mCblk->flags |= CBLK_UNDERRUN_ON; 3168 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3169 mFillingUpStatus = FS_FILLING; 3170 mResetDone = true; 3171 } 3172} 3173 3174void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3175{ 3176 mMute = muted; 3177} 3178 3179void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3180{ 3181 mVolume[0] = left; 3182 mVolume[1] = right; 3183} 3184 3185status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3186{ 3187 status_t status = DEAD_OBJECT; 3188 sp<ThreadBase> thread = mThread.promote(); 3189 if (thread != 0) { 3190 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3191 status = playbackThread->attachAuxEffect(this, EffectId); 3192 } 3193 return status; 3194} 3195 3196void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3197{ 3198 mAuxEffectId = EffectId; 3199 mAuxBuffer = buffer; 3200} 3201 3202// ---------------------------------------------------------------------------- 3203 3204// RecordTrack constructor must be called with AudioFlinger::mLock held 3205AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3206 const wp<ThreadBase>& thread, 3207 const sp<Client>& client, 3208 uint32_t sampleRate, 3209 int format, 3210 int channelCount, 3211 int frameCount, 3212 uint32_t flags, 3213 int sessionId) 3214 : TrackBase(thread, client, sampleRate, format, 3215 channelCount, frameCount, flags, 0, sessionId), 3216 mOverflow(false) 3217{ 3218 if (mCblk != NULL) { 3219 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3220 if (format == AudioSystem::PCM_16_BIT) { 3221 mCblk->frameSize = channelCount * sizeof(int16_t); 3222 } else if (format == AudioSystem::PCM_8_BIT) { 3223 mCblk->frameSize = channelCount * sizeof(int8_t); 3224 } else { 3225 mCblk->frameSize = sizeof(int8_t); 3226 } 3227 } 3228} 3229 3230AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3231{ 3232 sp<ThreadBase> thread = mThread.promote(); 3233 if (thread != 0) { 3234 AudioSystem::releaseInput(thread->id()); 3235 } 3236} 3237 3238status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3239{ 3240 audio_track_cblk_t* cblk = this->cblk(); 3241 uint32_t framesAvail; 3242 uint32_t framesReq = buffer->frameCount; 3243 3244 // Check if last stepServer failed, try to step now 3245 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3246 if (!step()) goto getNextBuffer_exit; 3247 LOGV("stepServer recovered"); 3248 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3249 } 3250 3251 framesAvail = cblk->framesAvailable_l(); 3252 3253 if (LIKELY(framesAvail)) { 3254 uint32_t s = cblk->server; 3255 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3256 3257 if (framesReq > framesAvail) { 3258 framesReq = framesAvail; 3259 } 3260 if (s + framesReq > bufferEnd) { 3261 framesReq = bufferEnd - s; 3262 } 3263 3264 buffer->raw = getBuffer(s, framesReq); 3265 if (buffer->raw == 0) goto getNextBuffer_exit; 3266 3267 buffer->frameCount = framesReq; 3268 return NO_ERROR; 3269 } 3270 3271getNextBuffer_exit: 3272 buffer->raw = 0; 3273 buffer->frameCount = 0; 3274 return NOT_ENOUGH_DATA; 3275} 3276 3277status_t AudioFlinger::RecordThread::RecordTrack::start() 3278{ 3279 sp<ThreadBase> thread = mThread.promote(); 3280 if (thread != 0) { 3281 RecordThread *recordThread = (RecordThread *)thread.get(); 3282 return recordThread->start(this); 3283 } else { 3284 return BAD_VALUE; 3285 } 3286} 3287 3288void AudioFlinger::RecordThread::RecordTrack::stop() 3289{ 3290 sp<ThreadBase> thread = mThread.promote(); 3291 if (thread != 0) { 3292 RecordThread *recordThread = (RecordThread *)thread.get(); 3293 recordThread->stop(this); 3294 TrackBase::reset(); 3295 // Force overerrun condition to avoid false overrun callback until first data is 3296 // read from buffer 3297 mCblk->flags |= CBLK_UNDERRUN_ON; 3298 } 3299} 3300 3301void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3302{ 3303 snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", 3304 (mClient == NULL) ? getpid() : mClient->pid(), 3305 mFormat, 3306 mCblk->channelCount, 3307 mSessionId, 3308 mFrameCount, 3309 mState, 3310 mCblk->sampleRate, 3311 mCblk->server, 3312 mCblk->user); 3313} 3314 3315 3316// ---------------------------------------------------------------------------- 3317 3318AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3319 const wp<ThreadBase>& thread, 3320 DuplicatingThread *sourceThread, 3321 uint32_t sampleRate, 3322 int format, 3323 int channelCount, 3324 int frameCount) 3325 : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), 3326 mActive(false), mSourceThread(sourceThread) 3327{ 3328 3329 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3330 if (mCblk != NULL) { 3331 mCblk->flags |= CBLK_DIRECTION_OUT; 3332 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3333 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3334 mOutBuffer.frameCount = 0; 3335 playbackThread->mTracks.add(this); 3336 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", 3337 mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); 3338 } else { 3339 LOGW("Error creating output track on thread %p", playbackThread); 3340 } 3341} 3342 3343AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3344{ 3345 clearBufferQueue(); 3346} 3347 3348status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3349{ 3350 status_t status = Track::start(); 3351 if (status != NO_ERROR) { 3352 return status; 3353 } 3354 3355 mActive = true; 3356 mRetryCount = 127; 3357 return status; 3358} 3359 3360void AudioFlinger::PlaybackThread::OutputTrack::stop() 3361{ 3362 Track::stop(); 3363 clearBufferQueue(); 3364 mOutBuffer.frameCount = 0; 3365 mActive = false; 3366} 3367 3368bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3369{ 3370 Buffer *pInBuffer; 3371 Buffer inBuffer; 3372 uint32_t channelCount = mCblk->channelCount; 3373 bool outputBufferFull = false; 3374 inBuffer.frameCount = frames; 3375 inBuffer.i16 = data; 3376 3377 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3378 3379 if (!mActive && frames != 0) { 3380 start(); 3381 sp<ThreadBase> thread = mThread.promote(); 3382 if (thread != 0) { 3383 MixerThread *mixerThread = (MixerThread *)thread.get(); 3384 if (mCblk->frameCount > frames){ 3385 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3386 uint32_t startFrames = (mCblk->frameCount - frames); 3387 pInBuffer = new Buffer; 3388 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3389 pInBuffer->frameCount = startFrames; 3390 pInBuffer->i16 = pInBuffer->mBuffer; 3391 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3392 mBufferQueue.add(pInBuffer); 3393 } else { 3394 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3395 } 3396 } 3397 } 3398 } 3399 3400 while (waitTimeLeftMs) { 3401 // First write pending buffers, then new data 3402 if (mBufferQueue.size()) { 3403 pInBuffer = mBufferQueue.itemAt(0); 3404 } else { 3405 pInBuffer = &inBuffer; 3406 } 3407 3408 if (pInBuffer->frameCount == 0) { 3409 break; 3410 } 3411 3412 if (mOutBuffer.frameCount == 0) { 3413 mOutBuffer.frameCount = pInBuffer->frameCount; 3414 nsecs_t startTime = systemTime(); 3415 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3416 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3417 outputBufferFull = true; 3418 break; 3419 } 3420 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3421 if (waitTimeLeftMs >= waitTimeMs) { 3422 waitTimeLeftMs -= waitTimeMs; 3423 } else { 3424 waitTimeLeftMs = 0; 3425 } 3426 } 3427 3428 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3429 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3430 mCblk->stepUser(outFrames); 3431 pInBuffer->frameCount -= outFrames; 3432 pInBuffer->i16 += outFrames * channelCount; 3433 mOutBuffer.frameCount -= outFrames; 3434 mOutBuffer.i16 += outFrames * channelCount; 3435 3436 if (pInBuffer->frameCount == 0) { 3437 if (mBufferQueue.size()) { 3438 mBufferQueue.removeAt(0); 3439 delete [] pInBuffer->mBuffer; 3440 delete pInBuffer; 3441 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3442 } else { 3443 break; 3444 } 3445 } 3446 } 3447 3448 // If we could not write all frames, allocate a buffer and queue it for next time. 3449 if (inBuffer.frameCount) { 3450 sp<ThreadBase> thread = mThread.promote(); 3451 if (thread != 0 && !thread->standby()) { 3452 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3453 pInBuffer = new Buffer; 3454 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3455 pInBuffer->frameCount = inBuffer.frameCount; 3456 pInBuffer->i16 = pInBuffer->mBuffer; 3457 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3458 mBufferQueue.add(pInBuffer); 3459 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3460 } else { 3461 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3462 } 3463 } 3464 } 3465 3466 // Calling write() with a 0 length buffer, means that no more data will be written: 3467 // If no more buffers are pending, fill output track buffer to make sure it is started 3468 // by output mixer. 3469 if (frames == 0 && mBufferQueue.size() == 0) { 3470 if (mCblk->user < mCblk->frameCount) { 3471 frames = mCblk->frameCount - mCblk->user; 3472 pInBuffer = new Buffer; 3473 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3474 pInBuffer->frameCount = frames; 3475 pInBuffer->i16 = pInBuffer->mBuffer; 3476 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3477 mBufferQueue.add(pInBuffer); 3478 } else if (mActive) { 3479 stop(); 3480 } 3481 } 3482 3483 return outputBufferFull; 3484} 3485 3486status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3487{ 3488 int active; 3489 status_t result; 3490 audio_track_cblk_t* cblk = mCblk; 3491 uint32_t framesReq = buffer->frameCount; 3492 3493// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3494 buffer->frameCount = 0; 3495 3496 uint32_t framesAvail = cblk->framesAvailable(); 3497 3498 3499 if (framesAvail == 0) { 3500 Mutex::Autolock _l(cblk->lock); 3501 goto start_loop_here; 3502 while (framesAvail == 0) { 3503 active = mActive; 3504 if (UNLIKELY(!active)) { 3505 LOGV("Not active and NO_MORE_BUFFERS"); 3506 return AudioTrack::NO_MORE_BUFFERS; 3507 } 3508 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3509 if (result != NO_ERROR) { 3510 return AudioTrack::NO_MORE_BUFFERS; 3511 } 3512 // read the server count again 3513 start_loop_here: 3514 framesAvail = cblk->framesAvailable_l(); 3515 } 3516 } 3517 3518// if (framesAvail < framesReq) { 3519// return AudioTrack::NO_MORE_BUFFERS; 3520// } 3521 3522 if (framesReq > framesAvail) { 3523 framesReq = framesAvail; 3524 } 3525 3526 uint32_t u = cblk->user; 3527 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3528 3529 if (u + framesReq > bufferEnd) { 3530 framesReq = bufferEnd - u; 3531 } 3532 3533 buffer->frameCount = framesReq; 3534 buffer->raw = (void *)cblk->buffer(u); 3535 return NO_ERROR; 3536} 3537 3538 3539void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3540{ 3541 size_t size = mBufferQueue.size(); 3542 Buffer *pBuffer; 3543 3544 for (size_t i = 0; i < size; i++) { 3545 pBuffer = mBufferQueue.itemAt(i); 3546 delete [] pBuffer->mBuffer; 3547 delete pBuffer; 3548 } 3549 mBufferQueue.clear(); 3550} 3551 3552// ---------------------------------------------------------------------------- 3553 3554AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3555 : RefBase(), 3556 mAudioFlinger(audioFlinger), 3557 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3558 mPid(pid) 3559{ 3560 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3561} 3562 3563// Client destructor must be called with AudioFlinger::mLock held 3564AudioFlinger::Client::~Client() 3565{ 3566 mAudioFlinger->removeClient_l(mPid); 3567} 3568 3569const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3570{ 3571 return mMemoryDealer; 3572} 3573 3574// ---------------------------------------------------------------------------- 3575 3576AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3577 const sp<IAudioFlingerClient>& client, 3578 pid_t pid) 3579 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3580{ 3581} 3582 3583AudioFlinger::NotificationClient::~NotificationClient() 3584{ 3585 mClient.clear(); 3586} 3587 3588void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3589{ 3590 sp<NotificationClient> keep(this); 3591 { 3592 mAudioFlinger->removeNotificationClient(mPid); 3593 } 3594} 3595 3596// ---------------------------------------------------------------------------- 3597 3598AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3599 : BnAudioTrack(), 3600 mTrack(track) 3601{ 3602} 3603 3604AudioFlinger::TrackHandle::~TrackHandle() { 3605 // just stop the track on deletion, associated resources 3606 // will be freed from the main thread once all pending buffers have 3607 // been played. Unless it's not in the active track list, in which 3608 // case we free everything now... 3609 mTrack->destroy(); 3610} 3611 3612status_t AudioFlinger::TrackHandle::start() { 3613 return mTrack->start(); 3614} 3615 3616void AudioFlinger::TrackHandle::stop() { 3617 mTrack->stop(); 3618} 3619 3620void AudioFlinger::TrackHandle::flush() { 3621 mTrack->flush(); 3622} 3623 3624void AudioFlinger::TrackHandle::mute(bool e) { 3625 mTrack->mute(e); 3626} 3627 3628void AudioFlinger::TrackHandle::pause() { 3629 mTrack->pause(); 3630} 3631 3632void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3633 mTrack->setVolume(left, right); 3634} 3635 3636sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3637 return mTrack->getCblk(); 3638} 3639 3640status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3641{ 3642 return mTrack->attachAuxEffect(EffectId); 3643} 3644 3645status_t AudioFlinger::TrackHandle::onTransact( 3646 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3647{ 3648 return BnAudioTrack::onTransact(code, data, reply, flags); 3649} 3650 3651// ---------------------------------------------------------------------------- 3652 3653sp<IAudioRecord> AudioFlinger::openRecord( 3654 pid_t pid, 3655 int input, 3656 uint32_t sampleRate, 3657 int format, 3658 int channelCount, 3659 int frameCount, 3660 uint32_t flags, 3661 int *sessionId, 3662 status_t *status) 3663{ 3664 sp<RecordThread::RecordTrack> recordTrack; 3665 sp<RecordHandle> recordHandle; 3666 sp<Client> client; 3667 wp<Client> wclient; 3668 status_t lStatus; 3669 RecordThread *thread; 3670 size_t inFrameCount; 3671 int lSessionId; 3672 3673 // check calling permissions 3674 if (!recordingAllowed()) { 3675 lStatus = PERMISSION_DENIED; 3676 goto Exit; 3677 } 3678 3679 // add client to list 3680 { // scope for mLock 3681 Mutex::Autolock _l(mLock); 3682 thread = checkRecordThread_l(input); 3683 if (thread == NULL) { 3684 lStatus = BAD_VALUE; 3685 goto Exit; 3686 } 3687 3688 wclient = mClients.valueFor(pid); 3689 if (wclient != NULL) { 3690 client = wclient.promote(); 3691 } else { 3692 client = new Client(this, pid); 3693 mClients.add(pid, client); 3694 } 3695 3696 // If no audio session id is provided, create one here 3697 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 3698 lSessionId = *sessionId; 3699 } else { 3700 lSessionId = nextUniqueId(); 3701 if (sessionId != NULL) { 3702 *sessionId = lSessionId; 3703 } 3704 } 3705 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3706 recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, 3707 format, channelCount, frameCount, flags, lSessionId); 3708 } 3709 if (recordTrack->getCblk() == NULL) { 3710 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3711 // destructor is called by the TrackBase destructor with mLock held 3712 client.clear(); 3713 recordTrack.clear(); 3714 lStatus = NO_MEMORY; 3715 goto Exit; 3716 } 3717 3718 // return to handle to client 3719 recordHandle = new RecordHandle(recordTrack); 3720 lStatus = NO_ERROR; 3721 3722Exit: 3723 if (status) { 3724 *status = lStatus; 3725 } 3726 return recordHandle; 3727} 3728 3729// ---------------------------------------------------------------------------- 3730 3731AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3732 : BnAudioRecord(), 3733 mRecordTrack(recordTrack) 3734{ 3735} 3736 3737AudioFlinger::RecordHandle::~RecordHandle() { 3738 stop(); 3739} 3740 3741status_t AudioFlinger::RecordHandle::start() { 3742 LOGV("RecordHandle::start()"); 3743 return mRecordTrack->start(); 3744} 3745 3746void AudioFlinger::RecordHandle::stop() { 3747 LOGV("RecordHandle::stop()"); 3748 mRecordTrack->stop(); 3749} 3750 3751sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3752 return mRecordTrack->getCblk(); 3753} 3754 3755status_t AudioFlinger::RecordHandle::onTransact( 3756 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3757{ 3758 return BnAudioRecord::onTransact(code, data, reply, flags); 3759} 3760 3761// ---------------------------------------------------------------------------- 3762 3763AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : 3764 ThreadBase(audioFlinger, id), 3765 mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3766{ 3767 mReqChannelCount = AudioSystem::popCount(channels); 3768 mReqSampleRate = sampleRate; 3769 readInputParameters(); 3770} 3771 3772 3773AudioFlinger::RecordThread::~RecordThread() 3774{ 3775 delete[] mRsmpInBuffer; 3776 if (mResampler != 0) { 3777 delete mResampler; 3778 delete[] mRsmpOutBuffer; 3779 } 3780} 3781 3782void AudioFlinger::RecordThread::onFirstRef() 3783{ 3784 const size_t SIZE = 256; 3785 char buffer[SIZE]; 3786 3787 snprintf(buffer, SIZE, "Record Thread %p", this); 3788 3789 run(buffer, PRIORITY_URGENT_AUDIO); 3790} 3791 3792bool AudioFlinger::RecordThread::threadLoop() 3793{ 3794 AudioBufferProvider::Buffer buffer; 3795 sp<RecordTrack> activeTrack; 3796 3797 // start recording 3798 while (!exitPending()) { 3799 3800 processConfigEvents(); 3801 3802 { // scope for mLock 3803 Mutex::Autolock _l(mLock); 3804 checkForNewParameters_l(); 3805 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3806 if (!mStandby) { 3807 mInput->standby(); 3808 mStandby = true; 3809 } 3810 3811 if (exitPending()) break; 3812 3813 LOGV("RecordThread: loop stopping"); 3814 // go to sleep 3815 mWaitWorkCV.wait(mLock); 3816 LOGV("RecordThread: loop starting"); 3817 continue; 3818 } 3819 if (mActiveTrack != 0) { 3820 if (mActiveTrack->mState == TrackBase::PAUSING) { 3821 if (!mStandby) { 3822 mInput->standby(); 3823 mStandby = true; 3824 } 3825 mActiveTrack.clear(); 3826 mStartStopCond.broadcast(); 3827 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3828 if (mReqChannelCount != mActiveTrack->channelCount()) { 3829 mActiveTrack.clear(); 3830 mStartStopCond.broadcast(); 3831 } else if (mBytesRead != 0) { 3832 // record start succeeds only if first read from audio input 3833 // succeeds 3834 if (mBytesRead > 0) { 3835 mActiveTrack->mState = TrackBase::ACTIVE; 3836 } else { 3837 mActiveTrack.clear(); 3838 } 3839 mStartStopCond.broadcast(); 3840 } 3841 mStandby = false; 3842 } 3843 } 3844 } 3845 3846 if (mActiveTrack != 0) { 3847 if (mActiveTrack->mState != TrackBase::ACTIVE && 3848 mActiveTrack->mState != TrackBase::RESUMING) { 3849 usleep(5000); 3850 continue; 3851 } 3852 buffer.frameCount = mFrameCount; 3853 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3854 size_t framesOut = buffer.frameCount; 3855 if (mResampler == 0) { 3856 // no resampling 3857 while (framesOut) { 3858 size_t framesIn = mFrameCount - mRsmpInIndex; 3859 if (framesIn) { 3860 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3861 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3862 if (framesIn > framesOut) 3863 framesIn = framesOut; 3864 mRsmpInIndex += framesIn; 3865 framesOut -= framesIn; 3866 if ((int)mChannelCount == mReqChannelCount || 3867 mFormat != AudioSystem::PCM_16_BIT) { 3868 memcpy(dst, src, framesIn * mFrameSize); 3869 } else { 3870 int16_t *src16 = (int16_t *)src; 3871 int16_t *dst16 = (int16_t *)dst; 3872 if (mChannelCount == 1) { 3873 while (framesIn--) { 3874 *dst16++ = *src16; 3875 *dst16++ = *src16++; 3876 } 3877 } else { 3878 while (framesIn--) { 3879 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3880 src16 += 2; 3881 } 3882 } 3883 } 3884 } 3885 if (framesOut && mFrameCount == mRsmpInIndex) { 3886 if (framesOut == mFrameCount && 3887 ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { 3888 mBytesRead = mInput->read(buffer.raw, mInputBytes); 3889 framesOut = 0; 3890 } else { 3891 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 3892 mRsmpInIndex = 0; 3893 } 3894 if (mBytesRead < 0) { 3895 LOGE("Error reading audio input"); 3896 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3897 // Force input into standby so that it tries to 3898 // recover at next read attempt 3899 mInput->standby(); 3900 usleep(5000); 3901 } 3902 mRsmpInIndex = mFrameCount; 3903 framesOut = 0; 3904 buffer.frameCount = 0; 3905 } 3906 } 3907 } 3908 } else { 3909 // resampling 3910 3911 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3912 // alter output frame count as if we were expecting stereo samples 3913 if (mChannelCount == 1 && mReqChannelCount == 1) { 3914 framesOut >>= 1; 3915 } 3916 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3917 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3918 // are 32 bit aligned which should be always true. 3919 if (mChannelCount == 2 && mReqChannelCount == 1) { 3920 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3921 // the resampler always outputs stereo samples: do post stereo to mono conversion 3922 int16_t *src = (int16_t *)mRsmpOutBuffer; 3923 int16_t *dst = buffer.i16; 3924 while (framesOut--) { 3925 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3926 src += 2; 3927 } 3928 } else { 3929 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3930 } 3931 3932 } 3933 mActiveTrack->releaseBuffer(&buffer); 3934 mActiveTrack->overflow(); 3935 } 3936 // client isn't retrieving buffers fast enough 3937 else { 3938 if (!mActiveTrack->setOverflow()) 3939 LOGW("RecordThread: buffer overflow"); 3940 // Release the processor for a while before asking for a new buffer. 3941 // This will give the application more chance to read from the buffer and 3942 // clear the overflow. 3943 usleep(5000); 3944 } 3945 } 3946 } 3947 3948 if (!mStandby) { 3949 mInput->standby(); 3950 } 3951 mActiveTrack.clear(); 3952 3953 mStartStopCond.broadcast(); 3954 3955 LOGV("RecordThread %p exiting", this); 3956 return false; 3957} 3958 3959status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 3960{ 3961 LOGV("RecordThread::start"); 3962 sp <ThreadBase> strongMe = this; 3963 status_t status = NO_ERROR; 3964 { 3965 AutoMutex lock(&mLock); 3966 if (mActiveTrack != 0) { 3967 if (recordTrack != mActiveTrack.get()) { 3968 status = -EBUSY; 3969 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3970 mActiveTrack->mState = TrackBase::ACTIVE; 3971 } 3972 return status; 3973 } 3974 3975 recordTrack->mState = TrackBase::IDLE; 3976 mActiveTrack = recordTrack; 3977 mLock.unlock(); 3978 status_t status = AudioSystem::startInput(mId); 3979 mLock.lock(); 3980 if (status != NO_ERROR) { 3981 mActiveTrack.clear(); 3982 return status; 3983 } 3984 mActiveTrack->mState = TrackBase::RESUMING; 3985 mRsmpInIndex = mFrameCount; 3986 mBytesRead = 0; 3987 // signal thread to start 3988 LOGV("Signal record thread"); 3989 mWaitWorkCV.signal(); 3990 // do not wait for mStartStopCond if exiting 3991 if (mExiting) { 3992 mActiveTrack.clear(); 3993 status = INVALID_OPERATION; 3994 goto startError; 3995 } 3996 mStartStopCond.wait(mLock); 3997 if (mActiveTrack == 0) { 3998 LOGV("Record failed to start"); 3999 status = BAD_VALUE; 4000 goto startError; 4001 } 4002 LOGV("Record started OK"); 4003 return status; 4004 } 4005startError: 4006 AudioSystem::stopInput(mId); 4007 return status; 4008} 4009 4010void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4011 LOGV("RecordThread::stop"); 4012 sp <ThreadBase> strongMe = this; 4013 { 4014 AutoMutex lock(&mLock); 4015 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4016 mActiveTrack->mState = TrackBase::PAUSING; 4017 // do not wait for mStartStopCond if exiting 4018 if (mExiting) { 4019 return; 4020 } 4021 mStartStopCond.wait(mLock); 4022 // if we have been restarted, recordTrack == mActiveTrack.get() here 4023 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4024 mLock.unlock(); 4025 AudioSystem::stopInput(mId); 4026 mLock.lock(); 4027 LOGV("Record stopped OK"); 4028 } 4029 } 4030 } 4031} 4032 4033status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4034{ 4035 const size_t SIZE = 256; 4036 char buffer[SIZE]; 4037 String8 result; 4038 pid_t pid = 0; 4039 4040 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4041 result.append(buffer); 4042 4043 if (mActiveTrack != 0) { 4044 result.append("Active Track:\n"); 4045 result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); 4046 mActiveTrack->dump(buffer, SIZE); 4047 result.append(buffer); 4048 4049 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4050 result.append(buffer); 4051 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4052 result.append(buffer); 4053 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4054 result.append(buffer); 4055 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4056 result.append(buffer); 4057 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4058 result.append(buffer); 4059 4060 4061 } else { 4062 result.append("No record client\n"); 4063 } 4064 write(fd, result.string(), result.size()); 4065 4066 dumpBase(fd, args); 4067 4068 return NO_ERROR; 4069} 4070 4071status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4072{ 4073 size_t framesReq = buffer->frameCount; 4074 size_t framesReady = mFrameCount - mRsmpInIndex; 4075 int channelCount; 4076 4077 if (framesReady == 0) { 4078 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 4079 if (mBytesRead < 0) { 4080 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4081 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4082 // Force input into standby so that it tries to 4083 // recover at next read attempt 4084 mInput->standby(); 4085 usleep(5000); 4086 } 4087 buffer->raw = 0; 4088 buffer->frameCount = 0; 4089 return NOT_ENOUGH_DATA; 4090 } 4091 mRsmpInIndex = 0; 4092 framesReady = mFrameCount; 4093 } 4094 4095 if (framesReq > framesReady) { 4096 framesReq = framesReady; 4097 } 4098 4099 if (mChannelCount == 1 && mReqChannelCount == 2) { 4100 channelCount = 1; 4101 } else { 4102 channelCount = 2; 4103 } 4104 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4105 buffer->frameCount = framesReq; 4106 return NO_ERROR; 4107} 4108 4109void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4110{ 4111 mRsmpInIndex += buffer->frameCount; 4112 buffer->frameCount = 0; 4113} 4114 4115bool AudioFlinger::RecordThread::checkForNewParameters_l() 4116{ 4117 bool reconfig = false; 4118 4119 while (!mNewParameters.isEmpty()) { 4120 status_t status = NO_ERROR; 4121 String8 keyValuePair = mNewParameters[0]; 4122 AudioParameter param = AudioParameter(keyValuePair); 4123 int value; 4124 int reqFormat = mFormat; 4125 int reqSamplingRate = mReqSampleRate; 4126 int reqChannelCount = mReqChannelCount; 4127 4128 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4129 reqSamplingRate = value; 4130 reconfig = true; 4131 } 4132 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4133 reqFormat = value; 4134 reconfig = true; 4135 } 4136 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4137 reqChannelCount = AudioSystem::popCount(value); 4138 reconfig = true; 4139 } 4140 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4141 // do not accept frame count changes if tracks are open as the track buffer 4142 // size depends on frame count and correct behavior would not be garantied 4143 // if frame count is changed after track creation 4144 if (mActiveTrack != 0) { 4145 status = INVALID_OPERATION; 4146 } else { 4147 reconfig = true; 4148 } 4149 } 4150 if (status == NO_ERROR) { 4151 status = mInput->setParameters(keyValuePair); 4152 if (status == INVALID_OPERATION) { 4153 mInput->standby(); 4154 status = mInput->setParameters(keyValuePair); 4155 } 4156 if (reconfig) { 4157 if (status == BAD_VALUE && 4158 reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && 4159 ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && 4160 (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { 4161 status = NO_ERROR; 4162 } 4163 if (status == NO_ERROR) { 4164 readInputParameters(); 4165 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4166 } 4167 } 4168 } 4169 4170 mNewParameters.removeAt(0); 4171 4172 mParamStatus = status; 4173 mParamCond.signal(); 4174 mWaitWorkCV.wait(mLock); 4175 } 4176 return reconfig; 4177} 4178 4179String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4180{ 4181 return mInput->getParameters(keys); 4182} 4183 4184void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4185 AudioSystem::OutputDescriptor desc; 4186 void *param2 = 0; 4187 4188 switch (event) { 4189 case AudioSystem::INPUT_OPENED: 4190 case AudioSystem::INPUT_CONFIG_CHANGED: 4191 desc.channels = mChannels; 4192 desc.samplingRate = mSampleRate; 4193 desc.format = mFormat; 4194 desc.frameCount = mFrameCount; 4195 desc.latency = 0; 4196 param2 = &desc; 4197 break; 4198 4199 case AudioSystem::INPUT_CLOSED: 4200 default: 4201 break; 4202 } 4203 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4204} 4205 4206void AudioFlinger::RecordThread::readInputParameters() 4207{ 4208 if (mRsmpInBuffer) delete mRsmpInBuffer; 4209 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4210 if (mResampler) delete mResampler; 4211 mResampler = 0; 4212 4213 mSampleRate = mInput->sampleRate(); 4214 mChannels = mInput->channels(); 4215 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 4216 mFormat = mInput->format(); 4217 mFrameSize = (uint16_t)mInput->frameSize(); 4218 mInputBytes = mInput->bufferSize(); 4219 mFrameCount = mInputBytes / mFrameSize; 4220 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4221 4222 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4223 { 4224 int channelCount; 4225 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4226 // stereo to mono post process as the resampler always outputs stereo. 4227 if (mChannelCount == 1 && mReqChannelCount == 2) { 4228 channelCount = 1; 4229 } else { 4230 channelCount = 2; 4231 } 4232 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4233 mResampler->setSampleRate(mSampleRate); 4234 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4235 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4236 4237 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4238 if (mChannelCount == 1 && mReqChannelCount == 1) { 4239 mFrameCount >>= 1; 4240 } 4241 4242 } 4243 mRsmpInIndex = mFrameCount; 4244} 4245 4246unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4247{ 4248 return mInput->getInputFramesLost(); 4249} 4250 4251// ---------------------------------------------------------------------------- 4252 4253int AudioFlinger::openOutput(uint32_t *pDevices, 4254 uint32_t *pSamplingRate, 4255 uint32_t *pFormat, 4256 uint32_t *pChannels, 4257 uint32_t *pLatencyMs, 4258 uint32_t flags) 4259{ 4260 status_t status; 4261 PlaybackThread *thread = NULL; 4262 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4263 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4264 uint32_t format = pFormat ? *pFormat : 0; 4265 uint32_t channels = pChannels ? *pChannels : 0; 4266 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4267 4268 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4269 pDevices ? *pDevices : 0, 4270 samplingRate, 4271 format, 4272 channels, 4273 flags); 4274 4275 if (pDevices == NULL || *pDevices == 0) { 4276 return 0; 4277 } 4278 Mutex::Autolock _l(mLock); 4279 4280 AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, 4281 (int *)&format, 4282 &channels, 4283 &samplingRate, 4284 &status); 4285 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4286 output, 4287 samplingRate, 4288 format, 4289 channels, 4290 status); 4291 4292 mHardwareStatus = AUDIO_HW_IDLE; 4293 if (output != 0) { 4294 int id = nextUniqueId(); 4295 if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || 4296 (format != AudioSystem::PCM_16_BIT) || 4297 (channels != AudioSystem::CHANNEL_OUT_STEREO)) { 4298 thread = new DirectOutputThread(this, output, id, *pDevices); 4299 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4300 } else { 4301 thread = new MixerThread(this, output, id, *pDevices); 4302 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4303 4304#ifdef LVMX 4305 unsigned bitsPerSample = 4306 (format == AudioSystem::PCM_16_BIT) ? 16 : 4307 ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); 4308 unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; 4309 int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); 4310 4311 LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); 4312 LifeVibes::setDevice(audioOutputType, *pDevices); 4313#endif 4314 4315 } 4316 mPlaybackThreads.add(id, thread); 4317 4318 if (pSamplingRate) *pSamplingRate = samplingRate; 4319 if (pFormat) *pFormat = format; 4320 if (pChannels) *pChannels = channels; 4321 if (pLatencyMs) *pLatencyMs = thread->latency(); 4322 4323 // notify client processes of the new output creation 4324 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4325 return id; 4326 } 4327 4328 return 0; 4329} 4330 4331int AudioFlinger::openDuplicateOutput(int output1, int output2) 4332{ 4333 Mutex::Autolock _l(mLock); 4334 MixerThread *thread1 = checkMixerThread_l(output1); 4335 MixerThread *thread2 = checkMixerThread_l(output2); 4336 4337 if (thread1 == NULL || thread2 == NULL) { 4338 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4339 return 0; 4340 } 4341 4342 int id = nextUniqueId(); 4343 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4344 thread->addOutputTrack(thread2); 4345 mPlaybackThreads.add(id, thread); 4346 // notify client processes of the new output creation 4347 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4348 return id; 4349} 4350 4351status_t AudioFlinger::closeOutput(int output) 4352{ 4353 // keep strong reference on the playback thread so that 4354 // it is not destroyed while exit() is executed 4355 sp <PlaybackThread> thread; 4356 { 4357 Mutex::Autolock _l(mLock); 4358 thread = checkPlaybackThread_l(output); 4359 if (thread == NULL) { 4360 return BAD_VALUE; 4361 } 4362 4363 LOGV("closeOutput() %d", output); 4364 4365 if (thread->type() == PlaybackThread::MIXER) { 4366 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4367 if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { 4368 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4369 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4370 } 4371 } 4372 } 4373 void *param2 = 0; 4374 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4375 mPlaybackThreads.removeItem(output); 4376 } 4377 thread->exit(); 4378 4379 if (thread->type() != PlaybackThread::DUPLICATING) { 4380 mAudioHardware->closeOutputStream(thread->getOutput()); 4381 } 4382 return NO_ERROR; 4383} 4384 4385status_t AudioFlinger::suspendOutput(int output) 4386{ 4387 Mutex::Autolock _l(mLock); 4388 PlaybackThread *thread = checkPlaybackThread_l(output); 4389 4390 if (thread == NULL) { 4391 return BAD_VALUE; 4392 } 4393 4394 LOGV("suspendOutput() %d", output); 4395 thread->suspend(); 4396 4397 return NO_ERROR; 4398} 4399 4400status_t AudioFlinger::restoreOutput(int output) 4401{ 4402 Mutex::Autolock _l(mLock); 4403 PlaybackThread *thread = checkPlaybackThread_l(output); 4404 4405 if (thread == NULL) { 4406 return BAD_VALUE; 4407 } 4408 4409 LOGV("restoreOutput() %d", output); 4410 4411 thread->restore(); 4412 4413 return NO_ERROR; 4414} 4415 4416int AudioFlinger::openInput(uint32_t *pDevices, 4417 uint32_t *pSamplingRate, 4418 uint32_t *pFormat, 4419 uint32_t *pChannels, 4420 uint32_t acoustics) 4421{ 4422 status_t status; 4423 RecordThread *thread = NULL; 4424 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4425 uint32_t format = pFormat ? *pFormat : 0; 4426 uint32_t channels = pChannels ? *pChannels : 0; 4427 uint32_t reqSamplingRate = samplingRate; 4428 uint32_t reqFormat = format; 4429 uint32_t reqChannels = channels; 4430 4431 if (pDevices == NULL || *pDevices == 0) { 4432 return 0; 4433 } 4434 Mutex::Autolock _l(mLock); 4435 4436 AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, 4437 (int *)&format, 4438 &channels, 4439 &samplingRate, 4440 &status, 4441 (AudioSystem::audio_in_acoustics)acoustics); 4442 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4443 input, 4444 samplingRate, 4445 format, 4446 channels, 4447 acoustics, 4448 status); 4449 4450 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4451 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4452 // or stereo to mono conversions on 16 bit PCM inputs. 4453 if (input == 0 && status == BAD_VALUE && 4454 reqFormat == format && format == AudioSystem::PCM_16_BIT && 4455 (samplingRate <= 2 * reqSamplingRate) && 4456 (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) { 4457 LOGV("openInput() reopening with proposed sampling rate and channels"); 4458 input = mAudioHardware->openInputStream(*pDevices, 4459 (int *)&format, 4460 &channels, 4461 &samplingRate, 4462 &status, 4463 (AudioSystem::audio_in_acoustics)acoustics); 4464 } 4465 4466 if (input != 0) { 4467 int id = nextUniqueId(); 4468 // Start record thread 4469 thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); 4470 mRecordThreads.add(id, thread); 4471 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4472 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4473 if (pFormat) *pFormat = format; 4474 if (pChannels) *pChannels = reqChannels; 4475 4476 input->standby(); 4477 4478 // notify client processes of the new input creation 4479 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4480 return id; 4481 } 4482 4483 return 0; 4484} 4485 4486status_t AudioFlinger::closeInput(int input) 4487{ 4488 // keep strong reference on the record thread so that 4489 // it is not destroyed while exit() is executed 4490 sp <RecordThread> thread; 4491 { 4492 Mutex::Autolock _l(mLock); 4493 thread = checkRecordThread_l(input); 4494 if (thread == NULL) { 4495 return BAD_VALUE; 4496 } 4497 4498 LOGV("closeInput() %d", input); 4499 void *param2 = 0; 4500 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4501 mRecordThreads.removeItem(input); 4502 } 4503 thread->exit(); 4504 4505 mAudioHardware->closeInputStream(thread->getInput()); 4506 4507 return NO_ERROR; 4508} 4509 4510status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4511{ 4512 Mutex::Autolock _l(mLock); 4513 MixerThread *dstThread = checkMixerThread_l(output); 4514 if (dstThread == NULL) { 4515 LOGW("setStreamOutput() bad output id %d", output); 4516 return BAD_VALUE; 4517 } 4518 4519 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4520 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4521 4522 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4523 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4524 if (thread != dstThread && 4525 thread->type() != PlaybackThread::DIRECT) { 4526 MixerThread *srcThread = (MixerThread *)thread; 4527 srcThread->invalidateTracks(stream); 4528 } 4529 } 4530 4531 return NO_ERROR; 4532} 4533 4534 4535int AudioFlinger::newAudioSessionId() 4536{ 4537 return nextUniqueId(); 4538} 4539 4540// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4541AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4542{ 4543 PlaybackThread *thread = NULL; 4544 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4545 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4546 } 4547 return thread; 4548} 4549 4550// checkMixerThread_l() must be called with AudioFlinger::mLock held 4551AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4552{ 4553 PlaybackThread *thread = checkPlaybackThread_l(output); 4554 if (thread != NULL) { 4555 if (thread->type() == PlaybackThread::DIRECT) { 4556 thread = NULL; 4557 } 4558 } 4559 return (MixerThread *)thread; 4560} 4561 4562// checkRecordThread_l() must be called with AudioFlinger::mLock held 4563AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4564{ 4565 RecordThread *thread = NULL; 4566 if (mRecordThreads.indexOfKey(input) >= 0) { 4567 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4568 } 4569 return thread; 4570} 4571 4572int AudioFlinger::nextUniqueId() 4573{ 4574 return android_atomic_inc(&mNextUniqueId); 4575} 4576 4577// ---------------------------------------------------------------------------- 4578// Effect management 4579// ---------------------------------------------------------------------------- 4580 4581 4582status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) 4583{ 4584 // check calling permissions 4585 if (!settingsAllowed()) { 4586 return PERMISSION_DENIED; 4587 } 4588 // only allow libraries loaded from /system/lib/soundfx for now 4589 if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) { 4590 return PERMISSION_DENIED; 4591 } 4592 4593 Mutex::Autolock _l(mLock); 4594 return EffectLoadLibrary(libPath, handle); 4595} 4596 4597status_t AudioFlinger::unloadEffectLibrary(int handle) 4598{ 4599 // check calling permissions 4600 if (!settingsAllowed()) { 4601 return PERMISSION_DENIED; 4602 } 4603 4604 Mutex::Autolock _l(mLock); 4605 return EffectUnloadLibrary(handle); 4606} 4607 4608status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4609{ 4610 Mutex::Autolock _l(mLock); 4611 return EffectQueryNumberEffects(numEffects); 4612} 4613 4614status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4615{ 4616 Mutex::Autolock _l(mLock); 4617 return EffectQueryEffect(index, descriptor); 4618} 4619 4620status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4621{ 4622 Mutex::Autolock _l(mLock); 4623 return EffectGetDescriptor(pUuid, descriptor); 4624} 4625 4626 4627// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4628static const effect_uuid_t VISUALIZATION_UUID_ = 4629 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4630 4631sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4632 effect_descriptor_t *pDesc, 4633 const sp<IEffectClient>& effectClient, 4634 int32_t priority, 4635 int output, 4636 int sessionId, 4637 status_t *status, 4638 int *id, 4639 int *enabled) 4640{ 4641 status_t lStatus = NO_ERROR; 4642 sp<EffectHandle> handle; 4643 effect_interface_t itfe; 4644 effect_descriptor_t desc; 4645 sp<Client> client; 4646 wp<Client> wclient; 4647 4648 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", 4649 pid, effectClient.get(), priority, sessionId, output); 4650 4651 if (pDesc == NULL) { 4652 lStatus = BAD_VALUE; 4653 goto Exit; 4654 } 4655 4656 { 4657 Mutex::Autolock _l(mLock); 4658 4659 // check recording permission for visualizer 4660 if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4661 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) { 4662 if (!recordingAllowed()) { 4663 lStatus = PERMISSION_DENIED; 4664 goto Exit; 4665 } 4666 } 4667 4668 if (!EffectIsNullUuid(&pDesc->uuid)) { 4669 // if uuid is specified, request effect descriptor 4670 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4671 if (lStatus < 0) { 4672 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4673 goto Exit; 4674 } 4675 } else { 4676 // if uuid is not specified, look for an available implementation 4677 // of the required type in effect factory 4678 if (EffectIsNullUuid(&pDesc->type)) { 4679 LOGW("createEffect() no effect type"); 4680 lStatus = BAD_VALUE; 4681 goto Exit; 4682 } 4683 uint32_t numEffects = 0; 4684 effect_descriptor_t d; 4685 bool found = false; 4686 4687 lStatus = EffectQueryNumberEffects(&numEffects); 4688 if (lStatus < 0) { 4689 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4690 goto Exit; 4691 } 4692 for (uint32_t i = 0; i < numEffects; i++) { 4693 lStatus = EffectQueryEffect(i, &desc); 4694 if (lStatus < 0) { 4695 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4696 continue; 4697 } 4698 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4699 // If matching type found save effect descriptor. If the session is 4700 // 0 and the effect is not auxiliary, continue enumeration in case 4701 // an auxiliary version of this effect type is available 4702 found = true; 4703 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4704 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX || 4705 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4706 break; 4707 } 4708 } 4709 } 4710 if (!found) { 4711 lStatus = BAD_VALUE; 4712 LOGW("createEffect() effect not found"); 4713 goto Exit; 4714 } 4715 // For same effect type, chose auxiliary version over insert version if 4716 // connect to output mix (Compliance to OpenSL ES) 4717 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && 4718 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4719 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4720 } 4721 } 4722 4723 // Do not allow auxiliary effects on a session different from 0 (output mix) 4724 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX && 4725 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4726 lStatus = INVALID_OPERATION; 4727 goto Exit; 4728 } 4729 4730 // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects 4731 // that can only be created by audio policy manager (running in same process) 4732 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && 4733 getpid() != pid) { 4734 lStatus = INVALID_OPERATION; 4735 goto Exit; 4736 } 4737 4738 // return effect descriptor 4739 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4740 4741 // If output is not specified try to find a matching audio session ID in one of the 4742 // output threads. 4743 // TODO: allow attachment of effect to inputs 4744 if (output == 0) { 4745 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) { 4746 // output must be specified by AudioPolicyManager when using session 4747 // AudioSystem::SESSION_OUTPUT_STAGE 4748 lStatus = BAD_VALUE; 4749 goto Exit; 4750 } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 4751 output = AudioSystem::getOutputForEffect(&desc); 4752 LOGV("createEffect() got output %d for effect %s", output, desc.name); 4753 } else { 4754 // look for the thread where the specified audio session is present 4755 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4756 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4757 output = mPlaybackThreads.keyAt(i); 4758 break; 4759 } 4760 } 4761 // If no output thread contains the requested session ID, default to 4762 // first output. The effect chain will be moved to the correct output 4763 // thread when a track with the same session ID is created 4764 if (output == 0 && mPlaybackThreads.size()) { 4765 output = mPlaybackThreads.keyAt(0); 4766 } 4767 } 4768 } 4769 PlaybackThread *thread = checkPlaybackThread_l(output); 4770 if (thread == NULL) { 4771 LOGE("createEffect() unknown output thread"); 4772 lStatus = BAD_VALUE; 4773 goto Exit; 4774 } 4775 4776 wclient = mClients.valueFor(pid); 4777 4778 if (wclient != NULL) { 4779 client = wclient.promote(); 4780 } else { 4781 client = new Client(this, pid); 4782 mClients.add(pid, client); 4783 } 4784 4785 // create effect on selected output trhead 4786 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4787 &desc, enabled, &lStatus); 4788 if (handle != 0 && id != NULL) { 4789 *id = handle->id(); 4790 } 4791 } 4792 4793Exit: 4794 if(status) { 4795 *status = lStatus; 4796 } 4797 return handle; 4798} 4799 4800status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4801{ 4802 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4803 session, srcOutput, dstOutput); 4804 Mutex::Autolock _l(mLock); 4805 if (srcOutput == dstOutput) { 4806 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4807 return NO_ERROR; 4808 } 4809 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4810 if (srcThread == NULL) { 4811 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4812 return BAD_VALUE; 4813 } 4814 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 4815 if (dstThread == NULL) { 4816 LOGW("moveEffects() bad dstOutput %d", dstOutput); 4817 return BAD_VALUE; 4818 } 4819 4820 Mutex::Autolock _dl(dstThread->mLock); 4821 Mutex::Autolock _sl(srcThread->mLock); 4822 moveEffectChain_l(session, srcThread, dstThread, false); 4823 4824 return NO_ERROR; 4825} 4826 4827// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 4828status_t AudioFlinger::moveEffectChain_l(int session, 4829 AudioFlinger::PlaybackThread *srcThread, 4830 AudioFlinger::PlaybackThread *dstThread, 4831 bool reRegister) 4832{ 4833 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 4834 session, srcThread, dstThread); 4835 4836 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 4837 if (chain == 0) { 4838 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 4839 session, srcThread); 4840 return INVALID_OPERATION; 4841 } 4842 4843 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 4844 // so that a new chain is created with correct parameters when first effect is added. This is 4845 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 4846 // removed. 4847 srcThread->removeEffectChain_l(chain); 4848 4849 // transfer all effects one by one so that new effect chain is created on new thread with 4850 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 4851 int dstOutput = dstThread->id(); 4852 sp<EffectChain> dstChain; 4853 uint32_t strategy; 4854 sp<EffectModule> effect = chain->getEffectFromId_l(0); 4855 while (effect != 0) { 4856 srcThread->removeEffect_l(effect); 4857 dstThread->addEffect_l(effect); 4858 // if the move request is not received from audio policy manager, the effect must be 4859 // re-registered with the new strategy and output 4860 if (dstChain == 0) { 4861 dstChain = effect->chain().promote(); 4862 if (dstChain == 0) { 4863 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 4864 srcThread->addEffect_l(effect); 4865 return NO_INIT; 4866 } 4867 strategy = dstChain->strategy(); 4868 } 4869 if (reRegister) { 4870 AudioSystem::unregisterEffect(effect->id()); 4871 AudioSystem::registerEffect(&effect->desc(), 4872 dstOutput, 4873 strategy, 4874 session, 4875 effect->id()); 4876 } 4877 effect = chain->getEffectFromId_l(0); 4878 } 4879 4880 return NO_ERROR; 4881} 4882 4883// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 4884sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( 4885 const sp<AudioFlinger::Client>& client, 4886 const sp<IEffectClient>& effectClient, 4887 int32_t priority, 4888 int sessionId, 4889 effect_descriptor_t *desc, 4890 int *enabled, 4891 status_t *status 4892 ) 4893{ 4894 sp<EffectModule> effect; 4895 sp<EffectHandle> handle; 4896 status_t lStatus; 4897 sp<Track> track; 4898 sp<EffectChain> chain; 4899 bool chainCreated = false; 4900 bool effectCreated = false; 4901 bool effectRegistered = false; 4902 4903 if (mOutput == 0) { 4904 LOGW("createEffect_l() Audio driver not initialized."); 4905 lStatus = NO_INIT; 4906 goto Exit; 4907 } 4908 4909 // Do not allow auxiliary effect on session other than 0 4910 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && 4911 sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 4912 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4913 desc->name, sessionId); 4914 lStatus = BAD_VALUE; 4915 goto Exit; 4916 } 4917 4918 // Do not allow effects with session ID 0 on direct output or duplicating threads 4919 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 4920 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) { 4921 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4922 desc->name, sessionId); 4923 lStatus = BAD_VALUE; 4924 goto Exit; 4925 } 4926 4927 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 4928 4929 { // scope for mLock 4930 Mutex::Autolock _l(mLock); 4931 4932 // check for existing effect chain with the requested audio session 4933 chain = getEffectChain_l(sessionId); 4934 if (chain == 0) { 4935 // create a new chain for this session 4936 LOGV("createEffect_l() new effect chain for session %d", sessionId); 4937 chain = new EffectChain(this, sessionId); 4938 addEffectChain_l(chain); 4939 chain->setStrategy(getStrategyForSession_l(sessionId)); 4940 chainCreated = true; 4941 } else { 4942 effect = chain->getEffectFromDesc_l(desc); 4943 } 4944 4945 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 4946 4947 if (effect == 0) { 4948 int id = mAudioFlinger->nextUniqueId(); 4949 // Check CPU and memory usage 4950 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 4951 if (lStatus != NO_ERROR) { 4952 goto Exit; 4953 } 4954 effectRegistered = true; 4955 // create a new effect module if none present in the chain 4956 effect = new EffectModule(this, chain, desc, id, sessionId); 4957 lStatus = effect->status(); 4958 if (lStatus != NO_ERROR) { 4959 goto Exit; 4960 } 4961 lStatus = chain->addEffect_l(effect); 4962 if (lStatus != NO_ERROR) { 4963 goto Exit; 4964 } 4965 effectCreated = true; 4966 4967 effect->setDevice(mDevice); 4968 effect->setMode(mAudioFlinger->getMode()); 4969 } 4970 // create effect handle and connect it to effect module 4971 handle = new EffectHandle(effect, client, effectClient, priority); 4972 lStatus = effect->addHandle(handle); 4973 if (enabled) { 4974 *enabled = (int)effect->isEnabled(); 4975 } 4976 } 4977 4978Exit: 4979 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 4980 Mutex::Autolock _l(mLock); 4981 if (effectCreated) { 4982 chain->removeEffect_l(effect); 4983 } 4984 if (effectRegistered) { 4985 AudioSystem::unregisterEffect(effect->id()); 4986 } 4987 if (chainCreated) { 4988 removeEffectChain_l(chain); 4989 } 4990 handle.clear(); 4991 } 4992 4993 if(status) { 4994 *status = lStatus; 4995 } 4996 return handle; 4997} 4998 4999// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5000// PlaybackThread::mLock held 5001status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect) 5002{ 5003 // check for existing effect chain with the requested audio session 5004 int sessionId = effect->sessionId(); 5005 sp<EffectChain> chain = getEffectChain_l(sessionId); 5006 bool chainCreated = false; 5007 5008 if (chain == 0) { 5009 // create a new chain for this session 5010 LOGV("addEffect_l() new effect chain for session %d", sessionId); 5011 chain = new EffectChain(this, sessionId); 5012 addEffectChain_l(chain); 5013 chain->setStrategy(getStrategyForSession_l(sessionId)); 5014 chainCreated = true; 5015 } 5016 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5017 5018 if (chain->getEffectFromId_l(effect->id()) != 0) { 5019 LOGW("addEffect_l() %p effect %s already present in chain %p", 5020 this, effect->desc().name, chain.get()); 5021 return BAD_VALUE; 5022 } 5023 5024 status_t status = chain->addEffect_l(effect); 5025 if (status != NO_ERROR) { 5026 if (chainCreated) { 5027 removeEffectChain_l(chain); 5028 } 5029 return status; 5030 } 5031 5032 effect->setDevice(mDevice); 5033 effect->setMode(mAudioFlinger->getMode()); 5034 return NO_ERROR; 5035} 5036 5037void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) { 5038 5039 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5040 effect_descriptor_t desc = effect->desc(); 5041 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5042 detachAuxEffect_l(effect->id()); 5043 } 5044 5045 sp<EffectChain> chain = effect->chain().promote(); 5046 if (chain != 0) { 5047 // remove effect chain if removing last effect 5048 if (chain->removeEffect_l(effect) == 0) { 5049 removeEffectChain_l(chain); 5050 } 5051 } else { 5052 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5053 } 5054} 5055 5056void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect, 5057 const wp<EffectHandle>& handle) { 5058 Mutex::Autolock _l(mLock); 5059 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5060 // delete the effect module if removing last handle on it 5061 if (effect->removeHandle(handle) == 0) { 5062 removeEffect_l(effect); 5063 AudioSystem::unregisterEffect(effect->id()); 5064 } 5065} 5066 5067status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5068{ 5069 int session = chain->sessionId(); 5070 int16_t *buffer = mMixBuffer; 5071 bool ownsBuffer = false; 5072 5073 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5074 if (session > 0) { 5075 // Only one effect chain can be present in direct output thread and it uses 5076 // the mix buffer as input 5077 if (mType != DIRECT) { 5078 size_t numSamples = mFrameCount * mChannelCount; 5079 buffer = new int16_t[numSamples]; 5080 memset(buffer, 0, numSamples * sizeof(int16_t)); 5081 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5082 ownsBuffer = true; 5083 } 5084 5085 // Attach all tracks with same session ID to this chain. 5086 for (size_t i = 0; i < mTracks.size(); ++i) { 5087 sp<Track> track = mTracks[i]; 5088 if (session == track->sessionId()) { 5089 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5090 track->setMainBuffer(buffer); 5091 } 5092 } 5093 5094 // indicate all active tracks in the chain 5095 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5096 sp<Track> track = mActiveTracks[i].promote(); 5097 if (track == 0) continue; 5098 if (session == track->sessionId()) { 5099 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5100 chain->startTrack(); 5101 } 5102 } 5103 } 5104 5105 chain->setInBuffer(buffer, ownsBuffer); 5106 chain->setOutBuffer(mMixBuffer); 5107 // Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect 5108 // chains list in order to be processed last as it contains output stage effects 5109 // Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before 5110 // session AudioSystem::SESSION_OUTPUT_STAGE to be processed 5111 // after track specific effects and before output stage 5112 // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and 5113 // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX 5114 // Effect chain for other sessions are inserted at beginning of effect 5115 // chains list to be processed before output mix effects. Relative order between other 5116 // sessions is not important 5117 size_t size = mEffectChains.size(); 5118 size_t i = 0; 5119 for (i = 0; i < size; i++) { 5120 if (mEffectChains[i]->sessionId() < session) break; 5121 } 5122 mEffectChains.insertAt(chain, i); 5123 5124 return NO_ERROR; 5125} 5126 5127size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5128{ 5129 int session = chain->sessionId(); 5130 5131 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5132 5133 for (size_t i = 0; i < mEffectChains.size(); i++) { 5134 if (chain == mEffectChains[i]) { 5135 mEffectChains.removeAt(i); 5136 // detach all tracks with same session ID from this chain 5137 for (size_t i = 0; i < mTracks.size(); ++i) { 5138 sp<Track> track = mTracks[i]; 5139 if (session == track->sessionId()) { 5140 track->setMainBuffer(mMixBuffer); 5141 } 5142 } 5143 break; 5144 } 5145 } 5146 return mEffectChains.size(); 5147} 5148 5149void AudioFlinger::PlaybackThread::lockEffectChains_l( 5150 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5151{ 5152 effectChains = mEffectChains; 5153 for (size_t i = 0; i < mEffectChains.size(); i++) { 5154 mEffectChains[i]->lock(); 5155 } 5156} 5157 5158void AudioFlinger::PlaybackThread::unlockEffectChains( 5159 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5160{ 5161 for (size_t i = 0; i < effectChains.size(); i++) { 5162 effectChains[i]->unlock(); 5163 } 5164} 5165 5166 5167sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) 5168{ 5169 sp<EffectModule> effect; 5170 5171 sp<EffectChain> chain = getEffectChain_l(sessionId); 5172 if (chain != 0) { 5173 effect = chain->getEffectFromId_l(effectId); 5174 } 5175 return effect; 5176} 5177 5178status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5179 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5180{ 5181 Mutex::Autolock _l(mLock); 5182 return attachAuxEffect_l(track, EffectId); 5183} 5184 5185status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5186 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5187{ 5188 status_t status = NO_ERROR; 5189 5190 if (EffectId == 0) { 5191 track->setAuxBuffer(0, NULL); 5192 } else { 5193 // Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX 5194 sp<EffectModule> effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId); 5195 if (effect != 0) { 5196 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5197 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5198 } else { 5199 status = INVALID_OPERATION; 5200 } 5201 } else { 5202 status = BAD_VALUE; 5203 } 5204 } 5205 return status; 5206} 5207 5208void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5209{ 5210 for (size_t i = 0; i < mTracks.size(); ++i) { 5211 sp<Track> track = mTracks[i]; 5212 if (track->auxEffectId() == effectId) { 5213 attachAuxEffect_l(track, 0); 5214 } 5215 } 5216} 5217 5218// ---------------------------------------------------------------------------- 5219// EffectModule implementation 5220// ---------------------------------------------------------------------------- 5221 5222#undef LOG_TAG 5223#define LOG_TAG "AudioFlinger::EffectModule" 5224 5225AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5226 const wp<AudioFlinger::EffectChain>& chain, 5227 effect_descriptor_t *desc, 5228 int id, 5229 int sessionId) 5230 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5231 mStatus(NO_INIT), mState(IDLE) 5232{ 5233 LOGV("Constructor %p", this); 5234 int lStatus; 5235 sp<ThreadBase> thread = mThread.promote(); 5236 if (thread == 0) { 5237 return; 5238 } 5239 PlaybackThread *p = (PlaybackThread *)thread.get(); 5240 5241 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5242 5243 // create effect engine from effect factory 5244 mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); 5245 5246 if (mStatus != NO_ERROR) { 5247 return; 5248 } 5249 lStatus = init(); 5250 if (lStatus < 0) { 5251 mStatus = lStatus; 5252 goto Error; 5253 } 5254 5255 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5256 return; 5257Error: 5258 EffectRelease(mEffectInterface); 5259 mEffectInterface = NULL; 5260 LOGV("Constructor Error %d", mStatus); 5261} 5262 5263AudioFlinger::EffectModule::~EffectModule() 5264{ 5265 LOGV("Destructor %p", this); 5266 if (mEffectInterface != NULL) { 5267 // release effect engine 5268 EffectRelease(mEffectInterface); 5269 } 5270} 5271 5272status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5273{ 5274 status_t status; 5275 5276 Mutex::Autolock _l(mLock); 5277 // First handle in mHandles has highest priority and controls the effect module 5278 int priority = handle->priority(); 5279 size_t size = mHandles.size(); 5280 sp<EffectHandle> h; 5281 size_t i; 5282 for (i = 0; i < size; i++) { 5283 h = mHandles[i].promote(); 5284 if (h == 0) continue; 5285 if (h->priority() <= priority) break; 5286 } 5287 // if inserted in first place, move effect control from previous owner to this handle 5288 if (i == 0) { 5289 if (h != 0) { 5290 h->setControl(false, true); 5291 } 5292 handle->setControl(true, false); 5293 status = NO_ERROR; 5294 } else { 5295 status = ALREADY_EXISTS; 5296 } 5297 mHandles.insertAt(handle, i); 5298 return status; 5299} 5300 5301size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5302{ 5303 Mutex::Autolock _l(mLock); 5304 size_t size = mHandles.size(); 5305 size_t i; 5306 for (i = 0; i < size; i++) { 5307 if (mHandles[i] == handle) break; 5308 } 5309 if (i == size) { 5310 return size; 5311 } 5312 mHandles.removeAt(i); 5313 size = mHandles.size(); 5314 // if removed from first place, move effect control from this handle to next in line 5315 if (i == 0 && size != 0) { 5316 sp<EffectHandle> h = mHandles[0].promote(); 5317 if (h != 0) { 5318 h->setControl(true, true); 5319 } 5320 } 5321 5322 return size; 5323} 5324 5325void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5326{ 5327 // keep a strong reference on this EffectModule to avoid calling the 5328 // destructor before we exit 5329 sp<EffectModule> keep(this); 5330 { 5331 sp<ThreadBase> thread = mThread.promote(); 5332 if (thread != 0) { 5333 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5334 playbackThread->disconnectEffect(keep, handle); 5335 } 5336 } 5337} 5338 5339void AudioFlinger::EffectModule::updateState() { 5340 Mutex::Autolock _l(mLock); 5341 5342 switch (mState) { 5343 case RESTART: 5344 reset_l(); 5345 // FALL THROUGH 5346 5347 case STARTING: 5348 // clear auxiliary effect input buffer for next accumulation 5349 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5350 memset(mConfig.inputCfg.buffer.raw, 5351 0, 5352 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5353 } 5354 start_l(); 5355 mState = ACTIVE; 5356 break; 5357 case STOPPING: 5358 stop_l(); 5359 mDisableWaitCnt = mMaxDisableWaitCnt; 5360 mState = STOPPED; 5361 break; 5362 case STOPPED: 5363 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5364 // turn off sequence. 5365 if (--mDisableWaitCnt == 0) { 5366 reset_l(); 5367 mState = IDLE; 5368 } 5369 break; 5370 default: //IDLE , ACTIVE 5371 break; 5372 } 5373} 5374 5375void AudioFlinger::EffectModule::process() 5376{ 5377 Mutex::Autolock _l(mLock); 5378 5379 if (mEffectInterface == NULL || 5380 mConfig.inputCfg.buffer.raw == NULL || 5381 mConfig.outputCfg.buffer.raw == NULL) { 5382 return; 5383 } 5384 5385 if (isProcessEnabled()) { 5386 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5387 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5388 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5389 mConfig.inputCfg.buffer.s32, 5390 mConfig.inputCfg.buffer.frameCount/2); 5391 } 5392 5393 // do the actual processing in the effect engine 5394 int ret = (*mEffectInterface)->process(mEffectInterface, 5395 &mConfig.inputCfg.buffer, 5396 &mConfig.outputCfg.buffer); 5397 5398 // force transition to IDLE state when engine is ready 5399 if (mState == STOPPED && ret == -ENODATA) { 5400 mDisableWaitCnt = 1; 5401 } 5402 5403 // clear auxiliary effect input buffer for next accumulation 5404 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5405 memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5406 } 5407 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5408 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){ 5409 // If an insert effect is idle and input buffer is different from output buffer, copy input to 5410 // output 5411 sp<EffectChain> chain = mChain.promote(); 5412 if (chain != 0 && chain->activeTracks() != 0) { 5413 size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t); 5414 if (mConfig.inputCfg.channels == CHANNEL_STEREO) { 5415 size *= 2; 5416 } 5417 memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size); 5418 } 5419 } 5420} 5421 5422void AudioFlinger::EffectModule::reset_l() 5423{ 5424 if (mEffectInterface == NULL) { 5425 return; 5426 } 5427 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5428} 5429 5430status_t AudioFlinger::EffectModule::configure() 5431{ 5432 uint32_t channels; 5433 if (mEffectInterface == NULL) { 5434 return NO_INIT; 5435 } 5436 5437 sp<ThreadBase> thread = mThread.promote(); 5438 if (thread == 0) { 5439 return DEAD_OBJECT; 5440 } 5441 5442 // TODO: handle configuration of effects replacing track process 5443 if (thread->channelCount() == 1) { 5444 channels = CHANNEL_MONO; 5445 } else { 5446 channels = CHANNEL_STEREO; 5447 } 5448 5449 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5450 mConfig.inputCfg.channels = CHANNEL_MONO; 5451 } else { 5452 mConfig.inputCfg.channels = channels; 5453 } 5454 mConfig.outputCfg.channels = channels; 5455 mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; 5456 mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; 5457 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5458 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5459 mConfig.inputCfg.bufferProvider.cookie = NULL; 5460 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5461 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5462 mConfig.outputCfg.bufferProvider.cookie = NULL; 5463 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5464 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5465 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5466 // Insert effect: 5467 // - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE, 5468 // always overwrites output buffer: input buffer == output buffer 5469 // - in other sessions: 5470 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5471 // other effect: overwrites output buffer: input buffer == output buffer 5472 // Auxiliary effect: 5473 // accumulates in output buffer: input buffer != output buffer 5474 // Therefore: accumulate <=> input buffer != output buffer 5475 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5476 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5477 } else { 5478 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5479 } 5480 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5481 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5482 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5483 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5484 5485 LOGV("configure() %p thread %p buffer %p framecount %d", 5486 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5487 5488 status_t cmdStatus; 5489 uint32_t size = sizeof(int); 5490 status_t status = (*mEffectInterface)->command(mEffectInterface, 5491 EFFECT_CMD_CONFIGURE, 5492 sizeof(effect_config_t), 5493 &mConfig, 5494 &size, 5495 &cmdStatus); 5496 if (status == 0) { 5497 status = cmdStatus; 5498 } 5499 5500 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5501 (1000 * mConfig.outputCfg.buffer.frameCount); 5502 5503 return status; 5504} 5505 5506status_t AudioFlinger::EffectModule::init() 5507{ 5508 Mutex::Autolock _l(mLock); 5509 if (mEffectInterface == NULL) { 5510 return NO_INIT; 5511 } 5512 status_t cmdStatus; 5513 uint32_t size = sizeof(status_t); 5514 status_t status = (*mEffectInterface)->command(mEffectInterface, 5515 EFFECT_CMD_INIT, 5516 0, 5517 NULL, 5518 &size, 5519 &cmdStatus); 5520 if (status == 0) { 5521 status = cmdStatus; 5522 } 5523 return status; 5524} 5525 5526status_t AudioFlinger::EffectModule::start_l() 5527{ 5528 if (mEffectInterface == NULL) { 5529 return NO_INIT; 5530 } 5531 status_t cmdStatus; 5532 uint32_t size = sizeof(status_t); 5533 status_t status = (*mEffectInterface)->command(mEffectInterface, 5534 EFFECT_CMD_ENABLE, 5535 0, 5536 NULL, 5537 &size, 5538 &cmdStatus); 5539 if (status == 0) { 5540 status = cmdStatus; 5541 } 5542 return status; 5543} 5544 5545status_t AudioFlinger::EffectModule::stop_l() 5546{ 5547 if (mEffectInterface == NULL) { 5548 return NO_INIT; 5549 } 5550 status_t cmdStatus; 5551 uint32_t size = sizeof(status_t); 5552 status_t status = (*mEffectInterface)->command(mEffectInterface, 5553 EFFECT_CMD_DISABLE, 5554 0, 5555 NULL, 5556 &size, 5557 &cmdStatus); 5558 if (status == 0) { 5559 status = cmdStatus; 5560 } 5561 return status; 5562} 5563 5564status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5565 uint32_t cmdSize, 5566 void *pCmdData, 5567 uint32_t *replySize, 5568 void *pReplyData) 5569{ 5570 Mutex::Autolock _l(mLock); 5571// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5572 5573 if (mEffectInterface == NULL) { 5574 return NO_INIT; 5575 } 5576 status_t status = (*mEffectInterface)->command(mEffectInterface, 5577 cmdCode, 5578 cmdSize, 5579 pCmdData, 5580 replySize, 5581 pReplyData); 5582 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5583 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5584 for (size_t i = 1; i < mHandles.size(); i++) { 5585 sp<EffectHandle> h = mHandles[i].promote(); 5586 if (h != 0) { 5587 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5588 } 5589 } 5590 } 5591 return status; 5592} 5593 5594status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5595{ 5596 Mutex::Autolock _l(mLock); 5597 LOGV("setEnabled %p enabled %d", this, enabled); 5598 5599 if (enabled != isEnabled()) { 5600 switch (mState) { 5601 // going from disabled to enabled 5602 case IDLE: 5603 mState = STARTING; 5604 break; 5605 case STOPPED: 5606 mState = RESTART; 5607 break; 5608 case STOPPING: 5609 mState = ACTIVE; 5610 break; 5611 5612 // going from enabled to disabled 5613 case RESTART: 5614 mState = STOPPED; 5615 break; 5616 case STARTING: 5617 mState = IDLE; 5618 break; 5619 case ACTIVE: 5620 mState = STOPPING; 5621 break; 5622 } 5623 for (size_t i = 1; i < mHandles.size(); i++) { 5624 sp<EffectHandle> h = mHandles[i].promote(); 5625 if (h != 0) { 5626 h->setEnabled(enabled); 5627 } 5628 } 5629 } 5630 return NO_ERROR; 5631} 5632 5633bool AudioFlinger::EffectModule::isEnabled() 5634{ 5635 switch (mState) { 5636 case RESTART: 5637 case STARTING: 5638 case ACTIVE: 5639 return true; 5640 case IDLE: 5641 case STOPPING: 5642 case STOPPED: 5643 default: 5644 return false; 5645 } 5646} 5647 5648bool AudioFlinger::EffectModule::isProcessEnabled() 5649{ 5650 switch (mState) { 5651 case RESTART: 5652 case ACTIVE: 5653 case STOPPING: 5654 case STOPPED: 5655 return true; 5656 case IDLE: 5657 case STARTING: 5658 default: 5659 return false; 5660 } 5661} 5662 5663status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5664{ 5665 Mutex::Autolock _l(mLock); 5666 status_t status = NO_ERROR; 5667 5668 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5669 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5670 if (isProcessEnabled() && 5671 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5672 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5673 status_t cmdStatus; 5674 uint32_t volume[2]; 5675 uint32_t *pVolume = NULL; 5676 uint32_t size = sizeof(volume); 5677 volume[0] = *left; 5678 volume[1] = *right; 5679 if (controller) { 5680 pVolume = volume; 5681 } 5682 status = (*mEffectInterface)->command(mEffectInterface, 5683 EFFECT_CMD_SET_VOLUME, 5684 size, 5685 volume, 5686 &size, 5687 pVolume); 5688 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5689 *left = volume[0]; 5690 *right = volume[1]; 5691 } 5692 } 5693 return status; 5694} 5695 5696status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5697{ 5698 Mutex::Autolock _l(mLock); 5699 status_t status = NO_ERROR; 5700 if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5701 // convert device bit field from AudioSystem to EffectApi format. 5702 device = deviceAudioSystemToEffectApi(device); 5703 if (device == 0) { 5704 return BAD_VALUE; 5705 } 5706 status_t cmdStatus; 5707 uint32_t size = sizeof(status_t); 5708 status = (*mEffectInterface)->command(mEffectInterface, 5709 EFFECT_CMD_SET_DEVICE, 5710 sizeof(uint32_t), 5711 &device, 5712 &size, 5713 &cmdStatus); 5714 if (status == NO_ERROR) { 5715 status = cmdStatus; 5716 } 5717 } 5718 return status; 5719} 5720 5721status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 5722{ 5723 Mutex::Autolock _l(mLock); 5724 status_t status = NO_ERROR; 5725 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 5726 // convert audio mode from AudioSystem to EffectApi format. 5727 int effectMode = modeAudioSystemToEffectApi(mode); 5728 if (effectMode < 0) { 5729 return BAD_VALUE; 5730 } 5731 status_t cmdStatus; 5732 uint32_t size = sizeof(status_t); 5733 status = (*mEffectInterface)->command(mEffectInterface, 5734 EFFECT_CMD_SET_AUDIO_MODE, 5735 sizeof(int), 5736 &effectMode, 5737 &size, 5738 &cmdStatus); 5739 if (status == NO_ERROR) { 5740 status = cmdStatus; 5741 } 5742 } 5743 return status; 5744} 5745 5746// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified 5747const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { 5748 DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE 5749 DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER 5750 DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET 5751 DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE 5752 DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO 5753 DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET 5754 DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT 5755 DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP 5756 DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES 5757 DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER 5758 DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL 5759}; 5760 5761uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) 5762{ 5763 uint32_t deviceOut = 0; 5764 while (device) { 5765 const uint32_t i = 31 - __builtin_clz(device); 5766 device &= ~(1 << i); 5767 if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { 5768 LOGE("device convertion error for AudioSystem device 0x%08x", device); 5769 return 0; 5770 } 5771 deviceOut |= (uint32_t)sDeviceConvTable[i]; 5772 } 5773 return deviceOut; 5774} 5775 5776// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified 5777const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { 5778 AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL 5779 AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE 5780 AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL 5781}; 5782 5783int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) 5784{ 5785 int modeOut = -1; 5786 if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { 5787 modeOut = (int)sModeConvTable[mode]; 5788 } 5789 return modeOut; 5790} 5791 5792status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 5793{ 5794 const size_t SIZE = 256; 5795 char buffer[SIZE]; 5796 String8 result; 5797 5798 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 5799 result.append(buffer); 5800 5801 bool locked = tryLock(mLock); 5802 // failed to lock - AudioFlinger is probably deadlocked 5803 if (!locked) { 5804 result.append("\t\tCould not lock Fx mutex:\n"); 5805 } 5806 5807 result.append("\t\tSession Status State Engine:\n"); 5808 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 5809 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 5810 result.append(buffer); 5811 5812 result.append("\t\tDescriptor:\n"); 5813 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5814 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 5815 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 5816 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 5817 result.append(buffer); 5818 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5819 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 5820 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 5821 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 5822 result.append(buffer); 5823 snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", 5824 mDescriptor.apiVersion, 5825 mDescriptor.flags); 5826 result.append(buffer); 5827 snprintf(buffer, SIZE, "\t\t- name: %s\n", 5828 mDescriptor.name); 5829 result.append(buffer); 5830 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 5831 mDescriptor.implementor); 5832 result.append(buffer); 5833 5834 result.append("\t\t- Input configuration:\n"); 5835 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5836 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5837 (uint32_t)mConfig.inputCfg.buffer.raw, 5838 mConfig.inputCfg.buffer.frameCount, 5839 mConfig.inputCfg.samplingRate, 5840 mConfig.inputCfg.channels, 5841 mConfig.inputCfg.format); 5842 result.append(buffer); 5843 5844 result.append("\t\t- Output configuration:\n"); 5845 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5846 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5847 (uint32_t)mConfig.outputCfg.buffer.raw, 5848 mConfig.outputCfg.buffer.frameCount, 5849 mConfig.outputCfg.samplingRate, 5850 mConfig.outputCfg.channels, 5851 mConfig.outputCfg.format); 5852 result.append(buffer); 5853 5854 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 5855 result.append(buffer); 5856 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 5857 for (size_t i = 0; i < mHandles.size(); ++i) { 5858 sp<EffectHandle> handle = mHandles[i].promote(); 5859 if (handle != 0) { 5860 handle->dump(buffer, SIZE); 5861 result.append(buffer); 5862 } 5863 } 5864 5865 result.append("\n"); 5866 5867 write(fd, result.string(), result.length()); 5868 5869 if (locked) { 5870 mLock.unlock(); 5871 } 5872 5873 return NO_ERROR; 5874} 5875 5876// ---------------------------------------------------------------------------- 5877// EffectHandle implementation 5878// ---------------------------------------------------------------------------- 5879 5880#undef LOG_TAG 5881#define LOG_TAG "AudioFlinger::EffectHandle" 5882 5883AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 5884 const sp<AudioFlinger::Client>& client, 5885 const sp<IEffectClient>& effectClient, 5886 int32_t priority) 5887 : BnEffect(), 5888 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 5889{ 5890 LOGV("constructor %p", this); 5891 5892 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 5893 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 5894 if (mCblkMemory != 0) { 5895 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 5896 5897 if (mCblk) { 5898 new(mCblk) effect_param_cblk_t(); 5899 mBuffer = (uint8_t *)mCblk + bufOffset; 5900 } 5901 } else { 5902 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 5903 return; 5904 } 5905} 5906 5907AudioFlinger::EffectHandle::~EffectHandle() 5908{ 5909 LOGV("Destructor %p", this); 5910 disconnect(); 5911} 5912 5913status_t AudioFlinger::EffectHandle::enable() 5914{ 5915 if (!mHasControl) return INVALID_OPERATION; 5916 if (mEffect == 0) return DEAD_OBJECT; 5917 5918 return mEffect->setEnabled(true); 5919} 5920 5921status_t AudioFlinger::EffectHandle::disable() 5922{ 5923 if (!mHasControl) return INVALID_OPERATION; 5924 if (mEffect == NULL) return DEAD_OBJECT; 5925 5926 return mEffect->setEnabled(false); 5927} 5928 5929void AudioFlinger::EffectHandle::disconnect() 5930{ 5931 if (mEffect == 0) { 5932 return; 5933 } 5934 mEffect->disconnect(this); 5935 // release sp on module => module destructor can be called now 5936 mEffect.clear(); 5937 if (mCblk) { 5938 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 5939 } 5940 mCblkMemory.clear(); // and free the shared memory 5941 if (mClient != 0) { 5942 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 5943 mClient.clear(); 5944 } 5945} 5946 5947status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 5948 uint32_t cmdSize, 5949 void *pCmdData, 5950 uint32_t *replySize, 5951 void *pReplyData) 5952{ 5953// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 5954// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 5955 5956 // only get parameter command is permitted for applications not controlling the effect 5957 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 5958 return INVALID_OPERATION; 5959 } 5960 if (mEffect == 0) return DEAD_OBJECT; 5961 5962 // handle commands that are not forwarded transparently to effect engine 5963 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 5964 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 5965 // no risk to block the whole media server process or mixer threads is we are stuck here 5966 Mutex::Autolock _l(mCblk->lock); 5967 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 5968 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 5969 mCblk->serverIndex = 0; 5970 mCblk->clientIndex = 0; 5971 return BAD_VALUE; 5972 } 5973 status_t status = NO_ERROR; 5974 while (mCblk->serverIndex < mCblk->clientIndex) { 5975 int reply; 5976 uint32_t rsize = sizeof(int); 5977 int *p = (int *)(mBuffer + mCblk->serverIndex); 5978 int size = *p++; 5979 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 5980 LOGW("command(): invalid parameter block size"); 5981 break; 5982 } 5983 effect_param_t *param = (effect_param_t *)p; 5984 if (param->psize == 0 || param->vsize == 0) { 5985 LOGW("command(): null parameter or value size"); 5986 mCblk->serverIndex += size; 5987 continue; 5988 } 5989 uint32_t psize = sizeof(effect_param_t) + 5990 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 5991 param->vsize; 5992 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 5993 psize, 5994 p, 5995 &rsize, 5996 &reply); 5997 // stop at first error encountered 5998 if (ret != NO_ERROR) { 5999 status = ret; 6000 *(int *)pReplyData = reply; 6001 break; 6002 } else if (reply != NO_ERROR) { 6003 *(int *)pReplyData = reply; 6004 break; 6005 } 6006 mCblk->serverIndex += size; 6007 } 6008 mCblk->serverIndex = 0; 6009 mCblk->clientIndex = 0; 6010 return status; 6011 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6012 *(int *)pReplyData = NO_ERROR; 6013 return enable(); 6014 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6015 *(int *)pReplyData = NO_ERROR; 6016 return disable(); 6017 } 6018 6019 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6020} 6021 6022sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 6023 return mCblkMemory; 6024} 6025 6026void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 6027{ 6028 LOGV("setControl %p control %d", this, hasControl); 6029 6030 mHasControl = hasControl; 6031 if (signal && mEffectClient != 0) { 6032 mEffectClient->controlStatusChanged(hasControl); 6033 } 6034} 6035 6036void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6037 uint32_t cmdSize, 6038 void *pCmdData, 6039 uint32_t replySize, 6040 void *pReplyData) 6041{ 6042 if (mEffectClient != 0) { 6043 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6044 } 6045} 6046 6047 6048 6049void AudioFlinger::EffectHandle::setEnabled(bool enabled) 6050{ 6051 if (mEffectClient != 0) { 6052 mEffectClient->enableStatusChanged(enabled); 6053 } 6054} 6055 6056status_t AudioFlinger::EffectHandle::onTransact( 6057 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6058{ 6059 return BnEffect::onTransact(code, data, reply, flags); 6060} 6061 6062 6063void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 6064{ 6065 bool locked = tryLock(mCblk->lock); 6066 6067 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 6068 (mClient == NULL) ? getpid() : mClient->pid(), 6069 mPriority, 6070 mHasControl, 6071 !locked, 6072 mCblk->clientIndex, 6073 mCblk->serverIndex 6074 ); 6075 6076 if (locked) { 6077 mCblk->lock.unlock(); 6078 } 6079} 6080 6081#undef LOG_TAG 6082#define LOG_TAG "AudioFlinger::EffectChain" 6083 6084AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 6085 int sessionId) 6086 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false), 6087 mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 6088 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 6089{ 6090 mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 6091} 6092 6093AudioFlinger::EffectChain::~EffectChain() 6094{ 6095 if (mOwnInBuffer) { 6096 delete mInBuffer; 6097 } 6098 6099} 6100 6101// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6102sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6103{ 6104 sp<EffectModule> effect; 6105 size_t size = mEffects.size(); 6106 6107 for (size_t i = 0; i < size; i++) { 6108 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6109 effect = mEffects[i]; 6110 break; 6111 } 6112 } 6113 return effect; 6114} 6115 6116// getEffectFromId_l() must be called with PlaybackThread::mLock held 6117sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6118{ 6119 sp<EffectModule> effect; 6120 size_t size = mEffects.size(); 6121 6122 for (size_t i = 0; i < size; i++) { 6123 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6124 if (id == 0 || mEffects[i]->id() == id) { 6125 effect = mEffects[i]; 6126 break; 6127 } 6128 } 6129 return effect; 6130} 6131 6132// Must be called with EffectChain::mLock locked 6133void AudioFlinger::EffectChain::process_l() 6134{ 6135 size_t size = mEffects.size(); 6136 for (size_t i = 0; i < size; i++) { 6137 mEffects[i]->process(); 6138 } 6139 for (size_t i = 0; i < size; i++) { 6140 mEffects[i]->updateState(); 6141 } 6142 // if no track is active, input buffer must be cleared here as the mixer process 6143 // will not do it 6144 if (mSessionId > 0 && activeTracks() == 0) { 6145 sp<ThreadBase> thread = mThread.promote(); 6146 if (thread != 0) { 6147 size_t numSamples = thread->frameCount() * thread->channelCount(); 6148 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6149 } 6150 } 6151} 6152 6153// addEffect_l() must be called with PlaybackThread::mLock held 6154status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6155{ 6156 effect_descriptor_t desc = effect->desc(); 6157 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6158 6159 Mutex::Autolock _l(mLock); 6160 effect->setChain(this); 6161 sp<ThreadBase> thread = mThread.promote(); 6162 if (thread == 0) { 6163 return NO_INIT; 6164 } 6165 effect->setThread(thread); 6166 6167 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6168 // Auxiliary effects are inserted at the beginning of mEffects vector as 6169 // they are processed first and accumulated in chain input buffer 6170 mEffects.insertAt(effect, 0); 6171 6172 // the input buffer for auxiliary effect contains mono samples in 6173 // 32 bit format. This is to avoid saturation in AudoMixer 6174 // accumulation stage. Saturation is done in EffectModule::process() before 6175 // calling the process in effect engine 6176 size_t numSamples = thread->frameCount(); 6177 int32_t *buffer = new int32_t[numSamples]; 6178 memset(buffer, 0, numSamples * sizeof(int32_t)); 6179 effect->setInBuffer((int16_t *)buffer); 6180 // auxiliary effects output samples to chain input buffer for further processing 6181 // by insert effects 6182 effect->setOutBuffer(mInBuffer); 6183 } else { 6184 // Insert effects are inserted at the end of mEffects vector as they are processed 6185 // after track and auxiliary effects. 6186 // Insert effect order as a function of indicated preference: 6187 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6188 // another effect is present 6189 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6190 // last effect claiming first position 6191 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6192 // first effect claiming last position 6193 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6194 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6195 // already present 6196 6197 int size = (int)mEffects.size(); 6198 int idx_insert = size; 6199 int idx_insert_first = -1; 6200 int idx_insert_last = -1; 6201 6202 for (int i = 0; i < size; i++) { 6203 effect_descriptor_t d = mEffects[i]->desc(); 6204 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6205 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6206 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6207 // check invalid effect chaining combinations 6208 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6209 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6210 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6211 return INVALID_OPERATION; 6212 } 6213 // remember position of first insert effect and by default 6214 // select this as insert position for new effect 6215 if (idx_insert == size) { 6216 idx_insert = i; 6217 } 6218 // remember position of last insert effect claiming 6219 // first position 6220 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6221 idx_insert_first = i; 6222 } 6223 // remember position of first insert effect claiming 6224 // last position 6225 if (iPref == EFFECT_FLAG_INSERT_LAST && 6226 idx_insert_last == -1) { 6227 idx_insert_last = i; 6228 } 6229 } 6230 } 6231 6232 // modify idx_insert from first position if needed 6233 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6234 if (idx_insert_last != -1) { 6235 idx_insert = idx_insert_last; 6236 } else { 6237 idx_insert = size; 6238 } 6239 } else { 6240 if (idx_insert_first != -1) { 6241 idx_insert = idx_insert_first + 1; 6242 } 6243 } 6244 6245 // always read samples from chain input buffer 6246 effect->setInBuffer(mInBuffer); 6247 6248 // if last effect in the chain, output samples to chain 6249 // output buffer, otherwise to chain input buffer 6250 if (idx_insert == size) { 6251 if (idx_insert != 0) { 6252 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6253 mEffects[idx_insert-1]->configure(); 6254 } 6255 effect->setOutBuffer(mOutBuffer); 6256 } else { 6257 effect->setOutBuffer(mInBuffer); 6258 } 6259 mEffects.insertAt(effect, idx_insert); 6260 6261 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6262 } 6263 effect->configure(); 6264 return NO_ERROR; 6265} 6266 6267// removeEffect_l() must be called with PlaybackThread::mLock held 6268size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6269{ 6270 Mutex::Autolock _l(mLock); 6271 int size = (int)mEffects.size(); 6272 int i; 6273 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6274 6275 for (i = 0; i < size; i++) { 6276 if (effect == mEffects[i]) { 6277 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6278 delete[] effect->inBuffer(); 6279 } else { 6280 if (i == size - 1 && i != 0) { 6281 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6282 mEffects[i - 1]->configure(); 6283 } 6284 } 6285 mEffects.removeAt(i); 6286 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6287 break; 6288 } 6289 } 6290 6291 return mEffects.size(); 6292} 6293 6294// setDevice_l() must be called with PlaybackThread::mLock held 6295void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6296{ 6297 size_t size = mEffects.size(); 6298 for (size_t i = 0; i < size; i++) { 6299 mEffects[i]->setDevice(device); 6300 } 6301} 6302 6303// setMode_l() must be called with PlaybackThread::mLock held 6304void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6305{ 6306 size_t size = mEffects.size(); 6307 for (size_t i = 0; i < size; i++) { 6308 mEffects[i]->setMode(mode); 6309 } 6310} 6311 6312// setVolume_l() must be called with PlaybackThread::mLock held 6313bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6314{ 6315 uint32_t newLeft = *left; 6316 uint32_t newRight = *right; 6317 bool hasControl = false; 6318 int ctrlIdx = -1; 6319 size_t size = mEffects.size(); 6320 6321 // first update volume controller 6322 for (size_t i = size; i > 0; i--) { 6323 if (mEffects[i - 1]->isProcessEnabled() && 6324 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6325 ctrlIdx = i - 1; 6326 hasControl = true; 6327 break; 6328 } 6329 } 6330 6331 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6332 if (hasControl) { 6333 *left = mNewLeftVolume; 6334 *right = mNewRightVolume; 6335 } 6336 return hasControl; 6337 } 6338 6339 mVolumeCtrlIdx = ctrlIdx; 6340 mLeftVolume = newLeft; 6341 mRightVolume = newRight; 6342 6343 // second get volume update from volume controller 6344 if (ctrlIdx >= 0) { 6345 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6346 mNewLeftVolume = newLeft; 6347 mNewRightVolume = newRight; 6348 } 6349 // then indicate volume to all other effects in chain. 6350 // Pass altered volume to effects before volume controller 6351 // and requested volume to effects after controller 6352 uint32_t lVol = newLeft; 6353 uint32_t rVol = newRight; 6354 6355 for (size_t i = 0; i < size; i++) { 6356 if ((int)i == ctrlIdx) continue; 6357 // this also works for ctrlIdx == -1 when there is no volume controller 6358 if ((int)i > ctrlIdx) { 6359 lVol = *left; 6360 rVol = *right; 6361 } 6362 mEffects[i]->setVolume(&lVol, &rVol, false); 6363 } 6364 *left = newLeft; 6365 *right = newRight; 6366 6367 return hasControl; 6368} 6369 6370status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6371{ 6372 const size_t SIZE = 256; 6373 char buffer[SIZE]; 6374 String8 result; 6375 6376 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6377 result.append(buffer); 6378 6379 bool locked = tryLock(mLock); 6380 // failed to lock - AudioFlinger is probably deadlocked 6381 if (!locked) { 6382 result.append("\tCould not lock mutex:\n"); 6383 } 6384 6385 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6386 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6387 mEffects.size(), 6388 (uint32_t)mInBuffer, 6389 (uint32_t)mOutBuffer, 6390 mActiveTrackCnt); 6391 result.append(buffer); 6392 write(fd, result.string(), result.size()); 6393 6394 for (size_t i = 0; i < mEffects.size(); ++i) { 6395 sp<EffectModule> effect = mEffects[i]; 6396 if (effect != 0) { 6397 effect->dump(fd, args); 6398 } 6399 } 6400 6401 if (locked) { 6402 mLock.unlock(); 6403 } 6404 6405 return NO_ERROR; 6406} 6407 6408#undef LOG_TAG 6409#define LOG_TAG "AudioFlinger" 6410 6411// ---------------------------------------------------------------------------- 6412 6413status_t AudioFlinger::onTransact( 6414 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6415{ 6416 return BnAudioFlinger::onTransact(code, data, reply, flags); 6417} 6418 6419}; // namespace android 6420