AudioFlinger.cpp revision aeae3de947fa0b1e670c8472b32288962f97b4f5
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IServiceManager.h> 28#include <utils/Log.h> 29#include <binder/Parcel.h> 30#include <binder/IPCThreadState.h> 31#include <utils/String16.h> 32#include <utils/threads.h> 33 34#include <cutils/properties.h> 35 36#include <media/AudioTrack.h> 37#include <media/AudioRecord.h> 38 39#include <private/media/AudioTrackShared.h> 40#include <private/media/AudioEffectShared.h> 41#include <hardware_legacy/AudioHardwareInterface.h> 42 43#include "AudioMixer.h" 44#include "AudioFlinger.h" 45 46#ifdef WITH_A2DP 47#include "A2dpAudioInterface.h" 48#endif 49 50#ifdef LVMX 51#include "lifevibes.h" 52#endif 53 54#include <media/EffectsFactoryApi.h> 55#include <media/EffectVisualizerApi.h> 56 57// ---------------------------------------------------------------------------- 58// the sim build doesn't have gettid 59 60#ifndef HAVE_GETTID 61# define gettid getpid 62#endif 63 64// ---------------------------------------------------------------------------- 65 66extern const char * const gEffectLibPath; 67 68namespace android { 69 70static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; 71static const char* kHardwareLockedString = "Hardware lock is taken\n"; 72 73//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 74static const float MAX_GAIN = 4096.0f; 75static const float MAX_GAIN_INT = 0x1000; 76 77// retry counts for buffer fill timeout 78// 50 * ~20msecs = 1 second 79static const int8_t kMaxTrackRetries = 50; 80static const int8_t kMaxTrackStartupRetries = 50; 81// allow less retry attempts on direct output thread. 82// direct outputs can be a scarce resource in audio hardware and should 83// be released as quickly as possible. 84static const int8_t kMaxTrackRetriesDirect = 2; 85 86static const int kDumpLockRetries = 50; 87static const int kDumpLockSleep = 20000; 88 89static const nsecs_t kWarningThrottle = seconds(5); 90 91 92#define AUDIOFLINGER_SECURITY_ENABLED 1 93 94// ---------------------------------------------------------------------------- 95 96static bool recordingAllowed() { 97#ifndef HAVE_ANDROID_OS 98 return true; 99#endif 100#if AUDIOFLINGER_SECURITY_ENABLED 101 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 102 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 103 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 104 return ok; 105#else 106 if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO"))) 107 LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest"); 108 return true; 109#endif 110} 111 112static bool settingsAllowed() { 113#ifndef HAVE_ANDROID_OS 114 return true; 115#endif 116#if AUDIOFLINGER_SECURITY_ENABLED 117 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 118 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 119 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 120 return ok; 121#else 122 if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"))) 123 LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest"); 124 return true; 125#endif 126} 127 128// ---------------------------------------------------------------------------- 129 130AudioFlinger::AudioFlinger() 131 : BnAudioFlinger(), 132 mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1) 133{ 134 mHardwareStatus = AUDIO_HW_IDLE; 135 136 mAudioHardware = AudioHardwareInterface::create(); 137 138 mHardwareStatus = AUDIO_HW_INIT; 139 if (mAudioHardware->initCheck() == NO_ERROR) { 140 // open 16-bit output stream for s/w mixer 141 mMode = AudioSystem::MODE_NORMAL; 142 setMode(mMode); 143 144 setMasterVolume(1.0f); 145 setMasterMute(false); 146 } else { 147 LOGE("Couldn't even initialize the stubbed audio hardware!"); 148 } 149#ifdef LVMX 150 LifeVibes::init(); 151 mLifeVibesClientPid = -1; 152#endif 153} 154 155AudioFlinger::~AudioFlinger() 156{ 157 while (!mRecordThreads.isEmpty()) { 158 // closeInput() will remove first entry from mRecordThreads 159 closeInput(mRecordThreads.keyAt(0)); 160 } 161 while (!mPlaybackThreads.isEmpty()) { 162 // closeOutput() will remove first entry from mPlaybackThreads 163 closeOutput(mPlaybackThreads.keyAt(0)); 164 } 165 if (mAudioHardware) { 166 delete mAudioHardware; 167 } 168} 169 170 171 172status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 173{ 174 const size_t SIZE = 256; 175 char buffer[SIZE]; 176 String8 result; 177 178 result.append("Clients:\n"); 179 for (size_t i = 0; i < mClients.size(); ++i) { 180 wp<Client> wClient = mClients.valueAt(i); 181 if (wClient != 0) { 182 sp<Client> client = wClient.promote(); 183 if (client != 0) { 184 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 185 result.append(buffer); 186 } 187 } 188 } 189 write(fd, result.string(), result.size()); 190 return NO_ERROR; 191} 192 193 194status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 195{ 196 const size_t SIZE = 256; 197 char buffer[SIZE]; 198 String8 result; 199 int hardwareStatus = mHardwareStatus; 200 201 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 202 result.append(buffer); 203 write(fd, result.string(), result.size()); 204 return NO_ERROR; 205} 206 207status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 208{ 209 const size_t SIZE = 256; 210 char buffer[SIZE]; 211 String8 result; 212 snprintf(buffer, SIZE, "Permission Denial: " 213 "can't dump AudioFlinger from pid=%d, uid=%d\n", 214 IPCThreadState::self()->getCallingPid(), 215 IPCThreadState::self()->getCallingUid()); 216 result.append(buffer); 217 write(fd, result.string(), result.size()); 218 return NO_ERROR; 219} 220 221static bool tryLock(Mutex& mutex) 222{ 223 bool locked = false; 224 for (int i = 0; i < kDumpLockRetries; ++i) { 225 if (mutex.tryLock() == NO_ERROR) { 226 locked = true; 227 break; 228 } 229 usleep(kDumpLockSleep); 230 } 231 return locked; 232} 233 234status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 235{ 236 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 237 dumpPermissionDenial(fd, args); 238 } else { 239 // get state of hardware lock 240 bool hardwareLocked = tryLock(mHardwareLock); 241 if (!hardwareLocked) { 242 String8 result(kHardwareLockedString); 243 write(fd, result.string(), result.size()); 244 } else { 245 mHardwareLock.unlock(); 246 } 247 248 bool locked = tryLock(mLock); 249 250 // failed to lock - AudioFlinger is probably deadlocked 251 if (!locked) { 252 String8 result(kDeadlockedString); 253 write(fd, result.string(), result.size()); 254 } 255 256 dumpClients(fd, args); 257 dumpInternals(fd, args); 258 259 // dump playback threads 260 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 261 mPlaybackThreads.valueAt(i)->dump(fd, args); 262 } 263 264 // dump record threads 265 for (size_t i = 0; i < mRecordThreads.size(); i++) { 266 mRecordThreads.valueAt(i)->dump(fd, args); 267 } 268 269 if (mAudioHardware) { 270 mAudioHardware->dumpState(fd, args); 271 } 272 if (locked) mLock.unlock(); 273 } 274 return NO_ERROR; 275} 276 277 278// IAudioFlinger interface 279 280 281sp<IAudioTrack> AudioFlinger::createTrack( 282 pid_t pid, 283 int streamType, 284 uint32_t sampleRate, 285 int format, 286 int channelCount, 287 int frameCount, 288 uint32_t flags, 289 const sp<IMemory>& sharedBuffer, 290 int output, 291 int *sessionId, 292 status_t *status) 293{ 294 sp<PlaybackThread::Track> track; 295 sp<TrackHandle> trackHandle; 296 sp<Client> client; 297 wp<Client> wclient; 298 status_t lStatus; 299 int lSessionId; 300 301 if (streamType >= AudioSystem::NUM_STREAM_TYPES) { 302 LOGE("invalid stream type"); 303 lStatus = BAD_VALUE; 304 goto Exit; 305 } 306 307 { 308 Mutex::Autolock _l(mLock); 309 PlaybackThread *thread = checkPlaybackThread_l(output); 310 PlaybackThread *effectThread = NULL; 311 if (thread == NULL) { 312 LOGE("unknown output thread"); 313 lStatus = BAD_VALUE; 314 goto Exit; 315 } 316 317 wclient = mClients.valueFor(pid); 318 319 if (wclient != NULL) { 320 client = wclient.promote(); 321 } else { 322 client = new Client(this, pid); 323 mClients.add(pid, client); 324 } 325 326 LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 327 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 328 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 329 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 330 if (mPlaybackThreads.keyAt(i) != output) { 331 // prevent same audio session on different output threads 332 uint32_t sessions = t->hasAudioSession(*sessionId); 333 if (sessions & PlaybackThread::TRACK_SESSION) { 334 lStatus = BAD_VALUE; 335 goto Exit; 336 } 337 // check if an effect with same session ID is waiting for a track to be created 338 if (sessions & PlaybackThread::EFFECT_SESSION) { 339 effectThread = t.get(); 340 } 341 } 342 } 343 lSessionId = *sessionId; 344 } else { 345 // if no audio session id is provided, create one here 346 lSessionId = nextUniqueId(); 347 if (sessionId != NULL) { 348 *sessionId = lSessionId; 349 } 350 } 351 LOGV("createTrack() lSessionId: %d", lSessionId); 352 353 track = thread->createTrack_l(client, streamType, sampleRate, format, 354 channelCount, frameCount, sharedBuffer, lSessionId, &lStatus); 355 356 // move effect chain to this output thread if an effect on same session was waiting 357 // for a track to be created 358 if (lStatus == NO_ERROR && effectThread != NULL) { 359 Mutex::Autolock _dl(thread->mLock); 360 Mutex::Autolock _sl(effectThread->mLock); 361 moveEffectChain_l(lSessionId, effectThread, thread, true); 362 } 363 } 364 if (lStatus == NO_ERROR) { 365 trackHandle = new TrackHandle(track); 366 } else { 367 // remove local strong reference to Client before deleting the Track so that the Client 368 // destructor is called by the TrackBase destructor with mLock held 369 client.clear(); 370 track.clear(); 371 } 372 373Exit: 374 if(status) { 375 *status = lStatus; 376 } 377 return trackHandle; 378} 379 380uint32_t AudioFlinger::sampleRate(int output) const 381{ 382 Mutex::Autolock _l(mLock); 383 PlaybackThread *thread = checkPlaybackThread_l(output); 384 if (thread == NULL) { 385 LOGW("sampleRate() unknown thread %d", output); 386 return 0; 387 } 388 return thread->sampleRate(); 389} 390 391int AudioFlinger::channelCount(int output) const 392{ 393 Mutex::Autolock _l(mLock); 394 PlaybackThread *thread = checkPlaybackThread_l(output); 395 if (thread == NULL) { 396 LOGW("channelCount() unknown thread %d", output); 397 return 0; 398 } 399 return thread->channelCount(); 400} 401 402int AudioFlinger::format(int output) const 403{ 404 Mutex::Autolock _l(mLock); 405 PlaybackThread *thread = checkPlaybackThread_l(output); 406 if (thread == NULL) { 407 LOGW("format() unknown thread %d", output); 408 return 0; 409 } 410 return thread->format(); 411} 412 413size_t AudioFlinger::frameCount(int output) const 414{ 415 Mutex::Autolock _l(mLock); 416 PlaybackThread *thread = checkPlaybackThread_l(output); 417 if (thread == NULL) { 418 LOGW("frameCount() unknown thread %d", output); 419 return 0; 420 } 421 return thread->frameCount(); 422} 423 424uint32_t AudioFlinger::latency(int output) const 425{ 426 Mutex::Autolock _l(mLock); 427 PlaybackThread *thread = checkPlaybackThread_l(output); 428 if (thread == NULL) { 429 LOGW("latency() unknown thread %d", output); 430 return 0; 431 } 432 return thread->latency(); 433} 434 435status_t AudioFlinger::setMasterVolume(float value) 436{ 437 // check calling permissions 438 if (!settingsAllowed()) { 439 return PERMISSION_DENIED; 440 } 441 442 // when hw supports master volume, don't scale in sw mixer 443 AutoMutex lock(mHardwareLock); 444 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 445 if (mAudioHardware->setMasterVolume(value) == NO_ERROR) { 446 value = 1.0f; 447 } 448 mHardwareStatus = AUDIO_HW_IDLE; 449 450 mMasterVolume = value; 451 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 452 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 453 454 return NO_ERROR; 455} 456 457status_t AudioFlinger::setMode(int mode) 458{ 459 status_t ret; 460 461 // check calling permissions 462 if (!settingsAllowed()) { 463 return PERMISSION_DENIED; 464 } 465 if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) { 466 LOGW("Illegal value: setMode(%d)", mode); 467 return BAD_VALUE; 468 } 469 470 { // scope for the lock 471 AutoMutex lock(mHardwareLock); 472 mHardwareStatus = AUDIO_HW_SET_MODE; 473 ret = mAudioHardware->setMode(mode); 474 mHardwareStatus = AUDIO_HW_IDLE; 475 } 476 477 if (NO_ERROR == ret) { 478 Mutex::Autolock _l(mLock); 479 mMode = mode; 480 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 481 mPlaybackThreads.valueAt(i)->setMode(mode); 482#ifdef LVMX 483 LifeVibes::setMode(mode); 484#endif 485 } 486 487 return ret; 488} 489 490status_t AudioFlinger::setMicMute(bool state) 491{ 492 // check calling permissions 493 if (!settingsAllowed()) { 494 return PERMISSION_DENIED; 495 } 496 497 AutoMutex lock(mHardwareLock); 498 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 499 status_t ret = mAudioHardware->setMicMute(state); 500 mHardwareStatus = AUDIO_HW_IDLE; 501 return ret; 502} 503 504bool AudioFlinger::getMicMute() const 505{ 506 bool state = AudioSystem::MODE_INVALID; 507 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 508 mAudioHardware->getMicMute(&state); 509 mHardwareStatus = AUDIO_HW_IDLE; 510 return state; 511} 512 513status_t AudioFlinger::setMasterMute(bool muted) 514{ 515 // check calling permissions 516 if (!settingsAllowed()) { 517 return PERMISSION_DENIED; 518 } 519 520 mMasterMute = muted; 521 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 522 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 523 524 return NO_ERROR; 525} 526 527float AudioFlinger::masterVolume() const 528{ 529 return mMasterVolume; 530} 531 532bool AudioFlinger::masterMute() const 533{ 534 return mMasterMute; 535} 536 537status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 538{ 539 // check calling permissions 540 if (!settingsAllowed()) { 541 return PERMISSION_DENIED; 542 } 543 544 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { 545 return BAD_VALUE; 546 } 547 548 AutoMutex lock(mLock); 549 PlaybackThread *thread = NULL; 550 if (output) { 551 thread = checkPlaybackThread_l(output); 552 if (thread == NULL) { 553 return BAD_VALUE; 554 } 555 } 556 557 mStreamTypes[stream].volume = value; 558 559 if (thread == NULL) { 560 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 561 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 562 } 563 } else { 564 thread->setStreamVolume(stream, value); 565 } 566 567 return NO_ERROR; 568} 569 570status_t AudioFlinger::setStreamMute(int stream, bool muted) 571{ 572 // check calling permissions 573 if (!settingsAllowed()) { 574 return PERMISSION_DENIED; 575 } 576 577 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || 578 uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { 579 return BAD_VALUE; 580 } 581 582 mStreamTypes[stream].mute = muted; 583 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 584 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 585 586 return NO_ERROR; 587} 588 589float AudioFlinger::streamVolume(int stream, int output) const 590{ 591 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { 592 return 0.0f; 593 } 594 595 AutoMutex lock(mLock); 596 float volume; 597 if (output) { 598 PlaybackThread *thread = checkPlaybackThread_l(output); 599 if (thread == NULL) { 600 return 0.0f; 601 } 602 volume = thread->streamVolume(stream); 603 } else { 604 volume = mStreamTypes[stream].volume; 605 } 606 607 return volume; 608} 609 610bool AudioFlinger::streamMute(int stream) const 611{ 612 if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) { 613 return true; 614 } 615 616 return mStreamTypes[stream].mute; 617} 618 619bool AudioFlinger::isStreamActive(int stream) const 620{ 621 Mutex::Autolock _l(mLock); 622 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 623 if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) { 624 return true; 625 } 626 } 627 return false; 628} 629 630status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 631{ 632 status_t result; 633 634 LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 635 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 636 // check calling permissions 637 if (!settingsAllowed()) { 638 return PERMISSION_DENIED; 639 } 640 641#ifdef LVMX 642 AudioParameter param = AudioParameter(keyValuePairs); 643 LifeVibes::setParameters(ioHandle,keyValuePairs); 644 String8 key = String8(AudioParameter::keyRouting); 645 int device; 646 if (NO_ERROR != param.getInt(key, device)) { 647 device = -1; 648 } 649 650 key = String8(LifevibesTag); 651 String8 value; 652 int musicEnabled = -1; 653 if (NO_ERROR == param.get(key, value)) { 654 if (value == LifevibesEnable) { 655 mLifeVibesClientPid = IPCThreadState::self()->getCallingPid(); 656 musicEnabled = 1; 657 } else if (value == LifevibesDisable) { 658 mLifeVibesClientPid = -1; 659 musicEnabled = 0; 660 } 661 } 662#endif 663 664 // ioHandle == 0 means the parameters are global to the audio hardware interface 665 if (ioHandle == 0) { 666 AutoMutex lock(mHardwareLock); 667 mHardwareStatus = AUDIO_SET_PARAMETER; 668 result = mAudioHardware->setParameters(keyValuePairs); 669#ifdef LVMX 670 if (musicEnabled != -1) { 671 LifeVibes::enableMusic((bool) musicEnabled); 672 } 673#endif 674 mHardwareStatus = AUDIO_HW_IDLE; 675 return result; 676 } 677 678 // hold a strong ref on thread in case closeOutput() or closeInput() is called 679 // and the thread is exited once the lock is released 680 sp<ThreadBase> thread; 681 { 682 Mutex::Autolock _l(mLock); 683 thread = checkPlaybackThread_l(ioHandle); 684 if (thread == NULL) { 685 thread = checkRecordThread_l(ioHandle); 686 } 687 } 688 if (thread != NULL) { 689 result = thread->setParameters(keyValuePairs); 690#ifdef LVMX 691 if ((NO_ERROR == result) && (device != -1)) { 692 LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device); 693 } 694#endif 695 return result; 696 } 697 return BAD_VALUE; 698} 699 700String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 701{ 702// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 703// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 704 705 if (ioHandle == 0) { 706 return mAudioHardware->getParameters(keys); 707 } 708 709 Mutex::Autolock _l(mLock); 710 711 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 712 if (playbackThread != NULL) { 713 return playbackThread->getParameters(keys); 714 } 715 RecordThread *recordThread = checkRecordThread_l(ioHandle); 716 if (recordThread != NULL) { 717 return recordThread->getParameters(keys); 718 } 719 return String8(""); 720} 721 722size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 723{ 724 return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); 725} 726 727unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 728{ 729 if (ioHandle == 0) { 730 return 0; 731 } 732 733 Mutex::Autolock _l(mLock); 734 735 RecordThread *recordThread = checkRecordThread_l(ioHandle); 736 if (recordThread != NULL) { 737 return recordThread->getInputFramesLost(); 738 } 739 return 0; 740} 741 742status_t AudioFlinger::setVoiceVolume(float value) 743{ 744 // check calling permissions 745 if (!settingsAllowed()) { 746 return PERMISSION_DENIED; 747 } 748 749 AutoMutex lock(mHardwareLock); 750 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 751 status_t ret = mAudioHardware->setVoiceVolume(value); 752 mHardwareStatus = AUDIO_HW_IDLE; 753 754 return ret; 755} 756 757status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 758{ 759 status_t status; 760 761 Mutex::Autolock _l(mLock); 762 763 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 764 if (playbackThread != NULL) { 765 return playbackThread->getRenderPosition(halFrames, dspFrames); 766 } 767 768 return BAD_VALUE; 769} 770 771void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 772{ 773 774 Mutex::Autolock _l(mLock); 775 776 int pid = IPCThreadState::self()->getCallingPid(); 777 if (mNotificationClients.indexOfKey(pid) < 0) { 778 sp<NotificationClient> notificationClient = new NotificationClient(this, 779 client, 780 pid); 781 LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 782 783 mNotificationClients.add(pid, notificationClient); 784 785 sp<IBinder> binder = client->asBinder(); 786 binder->linkToDeath(notificationClient); 787 788 // the config change is always sent from playback or record threads to avoid deadlock 789 // with AudioSystem::gLock 790 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 791 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 792 } 793 794 for (size_t i = 0; i < mRecordThreads.size(); i++) { 795 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 796 } 797 } 798} 799 800void AudioFlinger::removeNotificationClient(pid_t pid) 801{ 802 Mutex::Autolock _l(mLock); 803 804 int index = mNotificationClients.indexOfKey(pid); 805 if (index >= 0) { 806 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 807 LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 808#ifdef LVMX 809 if (pid == mLifeVibesClientPid) { 810 LOGV("Disabling lifevibes"); 811 LifeVibes::enableMusic(false); 812 mLifeVibesClientPid = -1; 813 } 814#endif 815 mNotificationClients.removeItem(pid); 816 } 817} 818 819// audioConfigChanged_l() must be called with AudioFlinger::mLock held 820void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 821{ 822 size_t size = mNotificationClients.size(); 823 for (size_t i = 0; i < size; i++) { 824 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 825 } 826} 827 828// removeClient_l() must be called with AudioFlinger::mLock held 829void AudioFlinger::removeClient_l(pid_t pid) 830{ 831 LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 832 mClients.removeItem(pid); 833} 834 835 836// ---------------------------------------------------------------------------- 837 838AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id) 839 : Thread(false), 840 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 841 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false) 842{ 843} 844 845AudioFlinger::ThreadBase::~ThreadBase() 846{ 847 mParamCond.broadcast(); 848 mNewParameters.clear(); 849} 850 851void AudioFlinger::ThreadBase::exit() 852{ 853 // keep a strong ref on ourself so that we wont get 854 // destroyed in the middle of requestExitAndWait() 855 sp <ThreadBase> strongMe = this; 856 857 LOGV("ThreadBase::exit"); 858 { 859 AutoMutex lock(&mLock); 860 mExiting = true; 861 requestExit(); 862 mWaitWorkCV.signal(); 863 } 864 requestExitAndWait(); 865} 866 867uint32_t AudioFlinger::ThreadBase::sampleRate() const 868{ 869 return mSampleRate; 870} 871 872int AudioFlinger::ThreadBase::channelCount() const 873{ 874 return (int)mChannelCount; 875} 876 877int AudioFlinger::ThreadBase::format() const 878{ 879 return mFormat; 880} 881 882size_t AudioFlinger::ThreadBase::frameCount() const 883{ 884 return mFrameCount; 885} 886 887status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 888{ 889 status_t status; 890 891 LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 892 Mutex::Autolock _l(mLock); 893 894 mNewParameters.add(keyValuePairs); 895 mWaitWorkCV.signal(); 896 // wait condition with timeout in case the thread loop has exited 897 // before the request could be processed 898 if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { 899 status = mParamStatus; 900 mWaitWorkCV.signal(); 901 } else { 902 status = TIMED_OUT; 903 } 904 return status; 905} 906 907void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 908{ 909 Mutex::Autolock _l(mLock); 910 sendConfigEvent_l(event, param); 911} 912 913// sendConfigEvent_l() must be called with ThreadBase::mLock held 914void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 915{ 916 ConfigEvent *configEvent = new ConfigEvent(); 917 configEvent->mEvent = event; 918 configEvent->mParam = param; 919 mConfigEvents.add(configEvent); 920 LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 921 mWaitWorkCV.signal(); 922} 923 924void AudioFlinger::ThreadBase::processConfigEvents() 925{ 926 mLock.lock(); 927 while(!mConfigEvents.isEmpty()) { 928 LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 929 ConfigEvent *configEvent = mConfigEvents[0]; 930 mConfigEvents.removeAt(0); 931 // release mLock before locking AudioFlinger mLock: lock order is always 932 // AudioFlinger then ThreadBase to avoid cross deadlock 933 mLock.unlock(); 934 mAudioFlinger->mLock.lock(); 935 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 936 mAudioFlinger->mLock.unlock(); 937 delete configEvent; 938 mLock.lock(); 939 } 940 mLock.unlock(); 941} 942 943status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 944{ 945 const size_t SIZE = 256; 946 char buffer[SIZE]; 947 String8 result; 948 949 bool locked = tryLock(mLock); 950 if (!locked) { 951 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 952 write(fd, buffer, strlen(buffer)); 953 } 954 955 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 956 result.append(buffer); 957 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 958 result.append(buffer); 959 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 960 result.append(buffer); 961 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 962 result.append(buffer); 963 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 964 result.append(buffer); 965 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 966 result.append(buffer); 967 968 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 969 result.append(buffer); 970 result.append(" Index Command"); 971 for (size_t i = 0; i < mNewParameters.size(); ++i) { 972 snprintf(buffer, SIZE, "\n %02d ", i); 973 result.append(buffer); 974 result.append(mNewParameters[i]); 975 } 976 977 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 978 result.append(buffer); 979 snprintf(buffer, SIZE, " Index event param\n"); 980 result.append(buffer); 981 for (size_t i = 0; i < mConfigEvents.size(); i++) { 982 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 983 result.append(buffer); 984 } 985 result.append("\n"); 986 987 write(fd, result.string(), result.size()); 988 989 if (locked) { 990 mLock.unlock(); 991 } 992 return NO_ERROR; 993} 994 995 996// ---------------------------------------------------------------------------- 997 998AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 999 : ThreadBase(audioFlinger, id), 1000 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 1001 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1002 mDevice(device) 1003{ 1004 readOutputParameters(); 1005 1006 mMasterVolume = mAudioFlinger->masterVolume(); 1007 mMasterMute = mAudioFlinger->masterMute(); 1008 1009 for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { 1010 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1011 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1012 } 1013} 1014 1015AudioFlinger::PlaybackThread::~PlaybackThread() 1016{ 1017 delete [] mMixBuffer; 1018} 1019 1020status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1021{ 1022 dumpInternals(fd, args); 1023 dumpTracks(fd, args); 1024 dumpEffectChains(fd, args); 1025 return NO_ERROR; 1026} 1027 1028status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1029{ 1030 const size_t SIZE = 256; 1031 char buffer[SIZE]; 1032 String8 result; 1033 1034 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1035 result.append(buffer); 1036 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1037 for (size_t i = 0; i < mTracks.size(); ++i) { 1038 sp<Track> track = mTracks[i]; 1039 if (track != 0) { 1040 track->dump(buffer, SIZE); 1041 result.append(buffer); 1042 } 1043 } 1044 1045 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1046 result.append(buffer); 1047 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1048 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1049 wp<Track> wTrack = mActiveTracks[i]; 1050 if (wTrack != 0) { 1051 sp<Track> track = wTrack.promote(); 1052 if (track != 0) { 1053 track->dump(buffer, SIZE); 1054 result.append(buffer); 1055 } 1056 } 1057 } 1058 write(fd, result.string(), result.size()); 1059 return NO_ERROR; 1060} 1061 1062status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args) 1063{ 1064 const size_t SIZE = 256; 1065 char buffer[SIZE]; 1066 String8 result; 1067 1068 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1069 write(fd, buffer, strlen(buffer)); 1070 1071 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1072 sp<EffectChain> chain = mEffectChains[i]; 1073 if (chain != 0) { 1074 chain->dump(fd, args); 1075 } 1076 } 1077 return NO_ERROR; 1078} 1079 1080status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1081{ 1082 const size_t SIZE = 256; 1083 char buffer[SIZE]; 1084 String8 result; 1085 1086 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1087 result.append(buffer); 1088 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1089 result.append(buffer); 1090 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1091 result.append(buffer); 1092 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1093 result.append(buffer); 1094 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1095 result.append(buffer); 1096 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1097 result.append(buffer); 1098 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1099 result.append(buffer); 1100 write(fd, result.string(), result.size()); 1101 1102 dumpBase(fd, args); 1103 1104 return NO_ERROR; 1105} 1106 1107// Thread virtuals 1108status_t AudioFlinger::PlaybackThread::readyToRun() 1109{ 1110 if (mSampleRate == 0) { 1111 LOGE("No working audio driver found."); 1112 return NO_INIT; 1113 } 1114 LOGI("AudioFlinger's thread %p ready to run", this); 1115 return NO_ERROR; 1116} 1117 1118void AudioFlinger::PlaybackThread::onFirstRef() 1119{ 1120 const size_t SIZE = 256; 1121 char buffer[SIZE]; 1122 1123 snprintf(buffer, SIZE, "Playback Thread %p", this); 1124 1125 run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); 1126} 1127 1128// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1129sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1130 const sp<AudioFlinger::Client>& client, 1131 int streamType, 1132 uint32_t sampleRate, 1133 int format, 1134 int channelCount, 1135 int frameCount, 1136 const sp<IMemory>& sharedBuffer, 1137 int sessionId, 1138 status_t *status) 1139{ 1140 sp<Track> track; 1141 status_t lStatus; 1142 1143 if (mType == DIRECT) { 1144 if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) { 1145 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p", 1146 sampleRate, format, channelCount, mOutput); 1147 lStatus = BAD_VALUE; 1148 goto Exit; 1149 } 1150 } else { 1151 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1152 if (sampleRate > mSampleRate*2) { 1153 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1154 lStatus = BAD_VALUE; 1155 goto Exit; 1156 } 1157 } 1158 1159 if (mOutput == 0) { 1160 LOGE("Audio driver not initialized."); 1161 lStatus = NO_INIT; 1162 goto Exit; 1163 } 1164 1165 { // scope for mLock 1166 Mutex::Autolock _l(mLock); 1167 1168 // all tracks in same audio session must share the same routing strategy otherwise 1169 // conflicts will happen when tracks are moved from one output to another by audio policy 1170 // manager 1171 uint32_t strategy = 1172 AudioSystem::getStrategyForStream((AudioSystem::stream_type)streamType); 1173 for (size_t i = 0; i < mTracks.size(); ++i) { 1174 sp<Track> t = mTracks[i]; 1175 if (t != 0) { 1176 if (sessionId == t->sessionId() && 1177 strategy != AudioSystem::getStrategyForStream((AudioSystem::stream_type)t->type())) { 1178 lStatus = BAD_VALUE; 1179 goto Exit; 1180 } 1181 } 1182 } 1183 1184 track = new Track(this, client, streamType, sampleRate, format, 1185 channelCount, frameCount, sharedBuffer, sessionId); 1186 if (track->getCblk() == NULL || track->name() < 0) { 1187 lStatus = NO_MEMORY; 1188 goto Exit; 1189 } 1190 mTracks.add(track); 1191 1192 sp<EffectChain> chain = getEffectChain_l(sessionId); 1193 if (chain != 0) { 1194 LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1195 track->setMainBuffer(chain->inBuffer()); 1196 chain->setStrategy(AudioSystem::getStrategyForStream((AudioSystem::stream_type)track->type())); 1197 } 1198 } 1199 lStatus = NO_ERROR; 1200 1201Exit: 1202 if(status) { 1203 *status = lStatus; 1204 } 1205 return track; 1206} 1207 1208uint32_t AudioFlinger::PlaybackThread::latency() const 1209{ 1210 if (mOutput) { 1211 return mOutput->latency(); 1212 } 1213 else { 1214 return 0; 1215 } 1216} 1217 1218status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1219{ 1220#ifdef LVMX 1221 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1222 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1223 LifeVibes::setMasterVolume(audioOutputType, value); 1224 } 1225#endif 1226 mMasterVolume = value; 1227 return NO_ERROR; 1228} 1229 1230status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1231{ 1232#ifdef LVMX 1233 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1234 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1235 LifeVibes::setMasterMute(audioOutputType, muted); 1236 } 1237#endif 1238 mMasterMute = muted; 1239 return NO_ERROR; 1240} 1241 1242float AudioFlinger::PlaybackThread::masterVolume() const 1243{ 1244 return mMasterVolume; 1245} 1246 1247bool AudioFlinger::PlaybackThread::masterMute() const 1248{ 1249 return mMasterMute; 1250} 1251 1252status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1253{ 1254#ifdef LVMX 1255 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1256 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1257 LifeVibes::setStreamVolume(audioOutputType, stream, value); 1258 } 1259#endif 1260 mStreamTypes[stream].volume = value; 1261 return NO_ERROR; 1262} 1263 1264status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1265{ 1266#ifdef LVMX 1267 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1268 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1269 LifeVibes::setStreamMute(audioOutputType, stream, muted); 1270 } 1271#endif 1272 mStreamTypes[stream].mute = muted; 1273 return NO_ERROR; 1274} 1275 1276float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1277{ 1278 return mStreamTypes[stream].volume; 1279} 1280 1281bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1282{ 1283 return mStreamTypes[stream].mute; 1284} 1285 1286bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const 1287{ 1288 Mutex::Autolock _l(mLock); 1289 size_t count = mActiveTracks.size(); 1290 for (size_t i = 0 ; i < count ; ++i) { 1291 sp<Track> t = mActiveTracks[i].promote(); 1292 if (t == 0) continue; 1293 Track* const track = t.get(); 1294 if (t->type() == stream) 1295 return true; 1296 } 1297 return false; 1298} 1299 1300// addTrack_l() must be called with ThreadBase::mLock held 1301status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1302{ 1303 status_t status = ALREADY_EXISTS; 1304 1305 // set retry count for buffer fill 1306 track->mRetryCount = kMaxTrackStartupRetries; 1307 if (mActiveTracks.indexOf(track) < 0) { 1308 // the track is newly added, make sure it fills up all its 1309 // buffers before playing. This is to ensure the client will 1310 // effectively get the latency it requested. 1311 track->mFillingUpStatus = Track::FS_FILLING; 1312 track->mResetDone = false; 1313 mActiveTracks.add(track); 1314 if (track->mainBuffer() != mMixBuffer) { 1315 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1316 if (chain != 0) { 1317 LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1318 chain->startTrack(); 1319 } 1320 } 1321 1322 status = NO_ERROR; 1323 } 1324 1325 LOGV("mWaitWorkCV.broadcast"); 1326 mWaitWorkCV.broadcast(); 1327 1328 return status; 1329} 1330 1331// destroyTrack_l() must be called with ThreadBase::mLock held 1332void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1333{ 1334 track->mState = TrackBase::TERMINATED; 1335 if (mActiveTracks.indexOf(track) < 0) { 1336 mTracks.remove(track); 1337 deleteTrackName_l(track->name()); 1338 } 1339} 1340 1341String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1342{ 1343 return mOutput->getParameters(keys); 1344} 1345 1346// destroyTrack_l() must be called with AudioFlinger::mLock held 1347void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1348 AudioSystem::OutputDescriptor desc; 1349 void *param2 = 0; 1350 1351 LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1352 1353 switch (event) { 1354 case AudioSystem::OUTPUT_OPENED: 1355 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1356 desc.channels = mChannels; 1357 desc.samplingRate = mSampleRate; 1358 desc.format = mFormat; 1359 desc.frameCount = mFrameCount; 1360 desc.latency = latency(); 1361 param2 = &desc; 1362 break; 1363 1364 case AudioSystem::STREAM_CONFIG_CHANGED: 1365 param2 = ¶m; 1366 case AudioSystem::OUTPUT_CLOSED: 1367 default: 1368 break; 1369 } 1370 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1371} 1372 1373void AudioFlinger::PlaybackThread::readOutputParameters() 1374{ 1375 mSampleRate = mOutput->sampleRate(); 1376 mChannels = mOutput->channels(); 1377 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 1378 mFormat = mOutput->format(); 1379 mFrameSize = (uint16_t)mOutput->frameSize(); 1380 mFrameCount = mOutput->bufferSize() / mFrameSize; 1381 1382 // FIXME - Current mixer implementation only supports stereo output: Always 1383 // Allocate a stereo buffer even if HW output is mono. 1384 if (mMixBuffer != NULL) delete[] mMixBuffer; 1385 mMixBuffer = new int16_t[mFrameCount * 2]; 1386 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1387 1388 // force reconfiguration of effect chains and engines to take new buffer size and audio 1389 // parameters into account 1390 // Note that mLock is not held when readOutputParameters() is called from the constructor 1391 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1392 // matter. 1393 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1394 Vector< sp<EffectChain> > effectChains = mEffectChains; 1395 for (size_t i = 0; i < effectChains.size(); i ++) { 1396 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1397 } 1398} 1399 1400status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1401{ 1402 if (halFrames == 0 || dspFrames == 0) { 1403 return BAD_VALUE; 1404 } 1405 if (mOutput == 0) { 1406 return INVALID_OPERATION; 1407 } 1408 *halFrames = mBytesWritten/mOutput->frameSize(); 1409 1410 return mOutput->getRenderPosition(dspFrames); 1411} 1412 1413uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1414{ 1415 Mutex::Autolock _l(mLock); 1416 uint32_t result = 0; 1417 if (getEffectChain_l(sessionId) != 0) { 1418 result = EFFECT_SESSION; 1419 } 1420 1421 for (size_t i = 0; i < mTracks.size(); ++i) { 1422 sp<Track> track = mTracks[i]; 1423 if (sessionId == track->sessionId() && 1424 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1425 result |= TRACK_SESSION; 1426 break; 1427 } 1428 } 1429 1430 return result; 1431} 1432 1433uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1434{ 1435 // session AudioSystem::SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1436 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1437 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 1438 return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 1439 } 1440 for (size_t i = 0; i < mTracks.size(); i++) { 1441 sp<Track> track = mTracks[i]; 1442 if (sessionId == track->sessionId() && 1443 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1444 return AudioSystem::getStrategyForStream((AudioSystem::stream_type) track->type()); 1445 } 1446 } 1447 return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 1448} 1449 1450sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId) 1451{ 1452 Mutex::Autolock _l(mLock); 1453 return getEffectChain_l(sessionId); 1454} 1455 1456sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId) 1457{ 1458 sp<EffectChain> chain; 1459 1460 size_t size = mEffectChains.size(); 1461 for (size_t i = 0; i < size; i++) { 1462 if (mEffectChains[i]->sessionId() == sessionId) { 1463 chain = mEffectChains[i]; 1464 break; 1465 } 1466 } 1467 return chain; 1468} 1469 1470void AudioFlinger::PlaybackThread::setMode(uint32_t mode) 1471{ 1472 Mutex::Autolock _l(mLock); 1473 size_t size = mEffectChains.size(); 1474 for (size_t i = 0; i < size; i++) { 1475 mEffectChains[i]->setMode_l(mode); 1476 } 1477} 1478 1479// ---------------------------------------------------------------------------- 1480 1481AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1482 : PlaybackThread(audioFlinger, output, id, device), 1483 mAudioMixer(0) 1484{ 1485 mType = PlaybackThread::MIXER; 1486 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1487 1488 // FIXME - Current mixer implementation only supports stereo output 1489 if (mChannelCount == 1) { 1490 LOGE("Invalid audio hardware channel count"); 1491 } 1492} 1493 1494AudioFlinger::MixerThread::~MixerThread() 1495{ 1496 delete mAudioMixer; 1497} 1498 1499bool AudioFlinger::MixerThread::threadLoop() 1500{ 1501 Vector< sp<Track> > tracksToRemove; 1502 uint32_t mixerStatus = MIXER_IDLE; 1503 nsecs_t standbyTime = systemTime(); 1504 size_t mixBufferSize = mFrameCount * mFrameSize; 1505 // FIXME: Relaxed timing because of a certain device that can't meet latency 1506 // Should be reduced to 2x after the vendor fixes the driver issue 1507 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1508 nsecs_t lastWarning = 0; 1509 bool longStandbyExit = false; 1510 uint32_t activeSleepTime = activeSleepTimeUs(); 1511 uint32_t idleSleepTime = idleSleepTimeUs(); 1512 uint32_t sleepTime = idleSleepTime; 1513 Vector< sp<EffectChain> > effectChains; 1514 1515 while (!exitPending()) 1516 { 1517 processConfigEvents(); 1518 1519 mixerStatus = MIXER_IDLE; 1520 { // scope for mLock 1521 1522 Mutex::Autolock _l(mLock); 1523 1524 if (checkForNewParameters_l()) { 1525 mixBufferSize = mFrameCount * mFrameSize; 1526 // FIXME: Relaxed timing because of a certain device that can't meet latency 1527 // Should be reduced to 2x after the vendor fixes the driver issue 1528 maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1529 activeSleepTime = activeSleepTimeUs(); 1530 idleSleepTime = idleSleepTimeUs(); 1531 } 1532 1533 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1534 1535 // put audio hardware into standby after short delay 1536 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1537 mSuspended) { 1538 if (!mStandby) { 1539 LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1540 mOutput->standby(); 1541 mStandby = true; 1542 mBytesWritten = 0; 1543 } 1544 1545 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1546 // we're about to wait, flush the binder command buffer 1547 IPCThreadState::self()->flushCommands(); 1548 1549 if (exitPending()) break; 1550 1551 // wait until we have something to do... 1552 LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1553 mWaitWorkCV.wait(mLock); 1554 LOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1555 1556 if (mMasterMute == false) { 1557 char value[PROPERTY_VALUE_MAX]; 1558 property_get("ro.audio.silent", value, "0"); 1559 if (atoi(value)) { 1560 LOGD("Silence is golden"); 1561 setMasterMute(true); 1562 } 1563 } 1564 1565 standbyTime = systemTime() + kStandbyTimeInNsecs; 1566 sleepTime = idleSleepTime; 1567 continue; 1568 } 1569 } 1570 1571 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1572 1573 // prevent any changes in effect chain list and in each effect chain 1574 // during mixing and effect process as the audio buffers could be deleted 1575 // or modified if an effect is created or deleted 1576 lockEffectChains_l(effectChains); 1577 } 1578 1579 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1580 // mix buffers... 1581 mAudioMixer->process(); 1582 sleepTime = 0; 1583 standbyTime = systemTime() + kStandbyTimeInNsecs; 1584 //TODO: delay standby when effects have a tail 1585 } else { 1586 // If no tracks are ready, sleep once for the duration of an output 1587 // buffer size, then write 0s to the output 1588 if (sleepTime == 0) { 1589 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1590 sleepTime = activeSleepTime; 1591 } else { 1592 sleepTime = idleSleepTime; 1593 } 1594 } else if (mBytesWritten != 0 || 1595 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 1596 memset (mMixBuffer, 0, mixBufferSize); 1597 sleepTime = 0; 1598 LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 1599 } 1600 // TODO add standby time extension fct of effect tail 1601 } 1602 1603 if (mSuspended) { 1604 sleepTime = suspendSleepTimeUs(); 1605 } 1606 // sleepTime == 0 means we must write to audio hardware 1607 if (sleepTime == 0) { 1608 for (size_t i = 0; i < effectChains.size(); i ++) { 1609 effectChains[i]->process_l(); 1610 } 1611 // enable changes in effect chain 1612 unlockEffectChains(effectChains); 1613#ifdef LVMX 1614 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1615 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1616 LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize); 1617 } 1618#endif 1619 mLastWriteTime = systemTime(); 1620 mInWrite = true; 1621 mBytesWritten += mixBufferSize; 1622 1623 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 1624 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 1625 mNumWrites++; 1626 mInWrite = false; 1627 nsecs_t now = systemTime(); 1628 nsecs_t delta = now - mLastWriteTime; 1629 if (delta > maxPeriod) { 1630 mNumDelayedWrites++; 1631 if ((now - lastWarning) > kWarningThrottle) { 1632 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 1633 ns2ms(delta), mNumDelayedWrites, this); 1634 lastWarning = now; 1635 } 1636 if (mStandby) { 1637 longStandbyExit = true; 1638 } 1639 } 1640 mStandby = false; 1641 } else { 1642 // enable changes in effect chain 1643 unlockEffectChains(effectChains); 1644 usleep(sleepTime); 1645 } 1646 1647 // finally let go of all our tracks, without the lock held 1648 // since we can't guarantee the destructors won't acquire that 1649 // same lock. 1650 tracksToRemove.clear(); 1651 1652 // Effect chains will be actually deleted here if they were removed from 1653 // mEffectChains list during mixing or effects processing 1654 effectChains.clear(); 1655 } 1656 1657 if (!mStandby) { 1658 mOutput->standby(); 1659 } 1660 1661 LOGV("MixerThread %p exiting", this); 1662 return false; 1663} 1664 1665// prepareTracks_l() must be called with ThreadBase::mLock held 1666uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 1667{ 1668 1669 uint32_t mixerStatus = MIXER_IDLE; 1670 // find out which tracks need to be processed 1671 size_t count = activeTracks.size(); 1672 size_t mixedTracks = 0; 1673 size_t tracksWithEffect = 0; 1674 1675 float masterVolume = mMasterVolume; 1676 bool masterMute = mMasterMute; 1677 1678 if (masterMute) { 1679 masterVolume = 0; 1680 } 1681#ifdef LVMX 1682 bool tracksConnectedChanged = false; 1683 bool stateChanged = false; 1684 1685 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1686 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1687 { 1688 int activeTypes = 0; 1689 for (size_t i=0 ; i<count ; i++) { 1690 sp<Track> t = activeTracks[i].promote(); 1691 if (t == 0) continue; 1692 Track* const track = t.get(); 1693 int iTracktype=track->type(); 1694 activeTypes |= 1<<track->type(); 1695 } 1696 LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute); 1697 } 1698#endif 1699 // Delegate master volume control to effect in output mix effect chain if needed 1700 sp<EffectChain> chain = getEffectChain_l(AudioSystem::SESSION_OUTPUT_MIX); 1701 if (chain != 0) { 1702 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 1703 chain->setVolume_l(&v, &v); 1704 masterVolume = (float)((v + (1 << 23)) >> 24); 1705 chain.clear(); 1706 } 1707 1708 for (size_t i=0 ; i<count ; i++) { 1709 sp<Track> t = activeTracks[i].promote(); 1710 if (t == 0) continue; 1711 1712 Track* const track = t.get(); 1713 audio_track_cblk_t* cblk = track->cblk(); 1714 1715 // The first time a track is added we wait 1716 // for all its buffers to be filled before processing it 1717 mAudioMixer->setActiveTrack(track->name()); 1718 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 1719 !track->isPaused() && !track->isTerminated()) 1720 { 1721 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 1722 1723 mixedTracks++; 1724 1725 // track->mainBuffer() != mMixBuffer means there is an effect chain 1726 // connected to the track 1727 chain.clear(); 1728 if (track->mainBuffer() != mMixBuffer) { 1729 chain = getEffectChain_l(track->sessionId()); 1730 // Delegate volume control to effect in track effect chain if needed 1731 if (chain != 0) { 1732 tracksWithEffect++; 1733 } else { 1734 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 1735 track->name(), track->sessionId()); 1736 } 1737 } 1738 1739 1740 int param = AudioMixer::VOLUME; 1741 if (track->mFillingUpStatus == Track::FS_FILLED) { 1742 // no ramp for the first volume setting 1743 track->mFillingUpStatus = Track::FS_ACTIVE; 1744 if (track->mState == TrackBase::RESUMING) { 1745 track->mState = TrackBase::ACTIVE; 1746 param = AudioMixer::RAMP_VOLUME; 1747 } 1748 } else if (cblk->server != 0) { 1749 // If the track is stopped before the first frame was mixed, 1750 // do not apply ramp 1751 param = AudioMixer::RAMP_VOLUME; 1752 } 1753 1754 // compute volume for this track 1755 int16_t left, right, aux; 1756 if (track->isMuted() || track->isPausing() || 1757 mStreamTypes[track->type()].mute) { 1758 left = right = aux = 0; 1759 if (track->isPausing()) { 1760 track->setPaused(); 1761 } 1762 } else { 1763 // read original volumes with volume control 1764 float typeVolume = mStreamTypes[track->type()].volume; 1765#ifdef LVMX 1766 bool streamMute=false; 1767 // read the volume from the LivesVibes audio engine. 1768 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1769 { 1770 LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); 1771 if (streamMute) { 1772 typeVolume = 0; 1773 } 1774 } 1775#endif 1776 float v = masterVolume * typeVolume; 1777 uint32_t vl = (uint32_t)(v * cblk->volume[0]) << 12; 1778 uint32_t vr = (uint32_t)(v * cblk->volume[1]) << 12; 1779 1780 // Delegate volume control to effect in track effect chain if needed 1781 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1782 // Do not ramp volume is volume is controlled by effect 1783 param = AudioMixer::VOLUME; 1784 track->mHasVolumeController = true; 1785 } else { 1786 // force no volume ramp when volume controller was just disabled or removed 1787 // from effect chain to avoid volume spike 1788 if (track->mHasVolumeController) { 1789 param = AudioMixer::VOLUME; 1790 } 1791 track->mHasVolumeController = false; 1792 } 1793 1794 // Convert volumes from 8.24 to 4.12 format 1795 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1796 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1797 left = int16_t(v_clamped); 1798 v_clamped = (vr + (1 << 11)) >> 12; 1799 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1800 right = int16_t(v_clamped); 1801 1802 v_clamped = (uint32_t)(v * cblk->sendLevel); 1803 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1804 aux = int16_t(v_clamped); 1805 } 1806 1807#ifdef LVMX 1808 if ( tracksConnectedChanged || stateChanged ) 1809 { 1810 // only do the ramp when the volume is changed by the user / application 1811 param = AudioMixer::VOLUME; 1812 } 1813#endif 1814 1815 // XXX: these things DON'T need to be done each time 1816 mAudioMixer->setBufferProvider(track); 1817 mAudioMixer->enable(AudioMixer::MIXING); 1818 1819 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1820 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1821 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1822 mAudioMixer->setParameter( 1823 AudioMixer::TRACK, 1824 AudioMixer::FORMAT, (void *)track->format()); 1825 mAudioMixer->setParameter( 1826 AudioMixer::TRACK, 1827 AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); 1828 mAudioMixer->setParameter( 1829 AudioMixer::RESAMPLE, 1830 AudioMixer::SAMPLE_RATE, 1831 (void *)(cblk->sampleRate)); 1832 mAudioMixer->setParameter( 1833 AudioMixer::TRACK, 1834 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1835 mAudioMixer->setParameter( 1836 AudioMixer::TRACK, 1837 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1838 1839 // reset retry count 1840 track->mRetryCount = kMaxTrackRetries; 1841 mixerStatus = MIXER_TRACKS_READY; 1842 } else { 1843 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1844 if (track->isStopped()) { 1845 track->reset(); 1846 } 1847 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1848 // We have consumed all the buffers of this track. 1849 // Remove it from the list of active tracks. 1850 tracksToRemove->add(track); 1851 } else { 1852 // No buffers for this track. Give it a few chances to 1853 // fill a buffer, then remove it from active list. 1854 if (--(track->mRetryCount) <= 0) { 1855 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1856 tracksToRemove->add(track); 1857 } else if (mixerStatus != MIXER_TRACKS_READY) { 1858 mixerStatus = MIXER_TRACKS_ENABLED; 1859 } 1860 } 1861 mAudioMixer->disable(AudioMixer::MIXING); 1862 } 1863 } 1864 1865 // remove all the tracks that need to be... 1866 count = tracksToRemove->size(); 1867 if (UNLIKELY(count)) { 1868 for (size_t i=0 ; i<count ; i++) { 1869 const sp<Track>& track = tracksToRemove->itemAt(i); 1870 mActiveTracks.remove(track); 1871 if (track->mainBuffer() != mMixBuffer) { 1872 chain = getEffectChain_l(track->sessionId()); 1873 if (chain != 0) { 1874 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1875 chain->stopTrack(); 1876 } 1877 } 1878 if (track->isTerminated()) { 1879 mTracks.remove(track); 1880 deleteTrackName_l(track->mName); 1881 } 1882 } 1883 } 1884 1885 // mix buffer must be cleared if all tracks are connected to an 1886 // effect chain as in this case the mixer will not write to 1887 // mix buffer and track effects will accumulate into it 1888 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1889 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1890 } 1891 1892 return mixerStatus; 1893} 1894 1895void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1896{ 1897 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1898 this, streamType, mTracks.size()); 1899 Mutex::Autolock _l(mLock); 1900 1901 size_t size = mTracks.size(); 1902 for (size_t i = 0; i < size; i++) { 1903 sp<Track> t = mTracks[i]; 1904 if (t->type() == streamType) { 1905 t->mCblk->lock.lock(); 1906 t->mCblk->flags |= CBLK_INVALID_ON; 1907 t->mCblk->cv.signal(); 1908 t->mCblk->lock.unlock(); 1909 } 1910 } 1911} 1912 1913 1914// getTrackName_l() must be called with ThreadBase::mLock held 1915int AudioFlinger::MixerThread::getTrackName_l() 1916{ 1917 return mAudioMixer->getTrackName(); 1918} 1919 1920// deleteTrackName_l() must be called with ThreadBase::mLock held 1921void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1922{ 1923 LOGV("remove track (%d) and delete from mixer", name); 1924 mAudioMixer->deleteTrackName(name); 1925} 1926 1927// checkForNewParameters_l() must be called with ThreadBase::mLock held 1928bool AudioFlinger::MixerThread::checkForNewParameters_l() 1929{ 1930 bool reconfig = false; 1931 1932 while (!mNewParameters.isEmpty()) { 1933 status_t status = NO_ERROR; 1934 String8 keyValuePair = mNewParameters[0]; 1935 AudioParameter param = AudioParameter(keyValuePair); 1936 int value; 1937 1938 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1939 reconfig = true; 1940 } 1941 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1942 if (value != AudioSystem::PCM_16_BIT) { 1943 status = BAD_VALUE; 1944 } else { 1945 reconfig = true; 1946 } 1947 } 1948 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1949 if (value != AudioSystem::CHANNEL_OUT_STEREO) { 1950 status = BAD_VALUE; 1951 } else { 1952 reconfig = true; 1953 } 1954 } 1955 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1956 // do not accept frame count changes if tracks are open as the track buffer 1957 // size depends on frame count and correct behavior would not be garantied 1958 // if frame count is changed after track creation 1959 if (!mTracks.isEmpty()) { 1960 status = INVALID_OPERATION; 1961 } else { 1962 reconfig = true; 1963 } 1964 } 1965 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1966 // forward device change to effects that have requested to be 1967 // aware of attached audio device. 1968 mDevice = (uint32_t)value; 1969 for (size_t i = 0; i < mEffectChains.size(); i++) { 1970 mEffectChains[i]->setDevice_l(mDevice); 1971 } 1972 } 1973 1974 if (status == NO_ERROR) { 1975 status = mOutput->setParameters(keyValuePair); 1976 if (!mStandby && status == INVALID_OPERATION) { 1977 mOutput->standby(); 1978 mStandby = true; 1979 mBytesWritten = 0; 1980 status = mOutput->setParameters(keyValuePair); 1981 } 1982 if (status == NO_ERROR && reconfig) { 1983 delete mAudioMixer; 1984 readOutputParameters(); 1985 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1986 for (size_t i = 0; i < mTracks.size() ; i++) { 1987 int name = getTrackName_l(); 1988 if (name < 0) break; 1989 mTracks[i]->mName = name; 1990 // limit track sample rate to 2 x new output sample rate 1991 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 1992 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 1993 } 1994 } 1995 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 1996 } 1997 } 1998 1999 mNewParameters.removeAt(0); 2000 2001 mParamStatus = status; 2002 mParamCond.signal(); 2003 mWaitWorkCV.wait(mLock); 2004 } 2005 return reconfig; 2006} 2007 2008status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2009{ 2010 const size_t SIZE = 256; 2011 char buffer[SIZE]; 2012 String8 result; 2013 2014 PlaybackThread::dumpInternals(fd, args); 2015 2016 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2017 result.append(buffer); 2018 write(fd, result.string(), result.size()); 2019 return NO_ERROR; 2020} 2021 2022uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 2023{ 2024 return (uint32_t)(mOutput->latency() * 1000) / 2; 2025} 2026 2027uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2028{ 2029 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2030} 2031 2032uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2033{ 2034 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2035} 2036 2037// ---------------------------------------------------------------------------- 2038AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2039 : PlaybackThread(audioFlinger, output, id, device) 2040{ 2041 mType = PlaybackThread::DIRECT; 2042} 2043 2044AudioFlinger::DirectOutputThread::~DirectOutputThread() 2045{ 2046} 2047 2048 2049static inline int16_t clamp16(int32_t sample) 2050{ 2051 if ((sample>>15) ^ (sample>>31)) 2052 sample = 0x7FFF ^ (sample>>31); 2053 return sample; 2054} 2055 2056static inline 2057int32_t mul(int16_t in, int16_t v) 2058{ 2059#if defined(__arm__) && !defined(__thumb__) 2060 int32_t out; 2061 asm( "smulbb %[out], %[in], %[v] \n" 2062 : [out]"=r"(out) 2063 : [in]"%r"(in), [v]"r"(v) 2064 : ); 2065 return out; 2066#else 2067 return in * int32_t(v); 2068#endif 2069} 2070 2071void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2072{ 2073 // Do not apply volume on compressed audio 2074 if (!AudioSystem::isLinearPCM(mFormat)) { 2075 return; 2076 } 2077 2078 // convert to signed 16 bit before volume calculation 2079 if (mFormat == AudioSystem::PCM_8_BIT) { 2080 size_t count = mFrameCount * mChannelCount; 2081 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2082 int16_t *dst = mMixBuffer + count-1; 2083 while(count--) { 2084 *dst-- = (int16_t)(*src--^0x80) << 8; 2085 } 2086 } 2087 2088 size_t frameCount = mFrameCount; 2089 int16_t *out = mMixBuffer; 2090 if (ramp) { 2091 if (mChannelCount == 1) { 2092 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2093 int32_t vlInc = d / (int32_t)frameCount; 2094 int32_t vl = ((int32_t)mLeftVolShort << 16); 2095 do { 2096 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2097 out++; 2098 vl += vlInc; 2099 } while (--frameCount); 2100 2101 } else { 2102 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2103 int32_t vlInc = d / (int32_t)frameCount; 2104 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2105 int32_t vrInc = d / (int32_t)frameCount; 2106 int32_t vl = ((int32_t)mLeftVolShort << 16); 2107 int32_t vr = ((int32_t)mRightVolShort << 16); 2108 do { 2109 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2110 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2111 out += 2; 2112 vl += vlInc; 2113 vr += vrInc; 2114 } while (--frameCount); 2115 } 2116 } else { 2117 if (mChannelCount == 1) { 2118 do { 2119 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2120 out++; 2121 } while (--frameCount); 2122 } else { 2123 do { 2124 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2125 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2126 out += 2; 2127 } while (--frameCount); 2128 } 2129 } 2130 2131 // convert back to unsigned 8 bit after volume calculation 2132 if (mFormat == AudioSystem::PCM_8_BIT) { 2133 size_t count = mFrameCount * mChannelCount; 2134 int16_t *src = mMixBuffer; 2135 uint8_t *dst = (uint8_t *)mMixBuffer; 2136 while(count--) { 2137 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2138 } 2139 } 2140 2141 mLeftVolShort = leftVol; 2142 mRightVolShort = rightVol; 2143} 2144 2145bool AudioFlinger::DirectOutputThread::threadLoop() 2146{ 2147 uint32_t mixerStatus = MIXER_IDLE; 2148 sp<Track> trackToRemove; 2149 sp<Track> activeTrack; 2150 nsecs_t standbyTime = systemTime(); 2151 int8_t *curBuf; 2152 size_t mixBufferSize = mFrameCount*mFrameSize; 2153 uint32_t activeSleepTime = activeSleepTimeUs(); 2154 uint32_t idleSleepTime = idleSleepTimeUs(); 2155 uint32_t sleepTime = idleSleepTime; 2156 // use shorter standby delay as on normal output to release 2157 // hardware resources as soon as possible 2158 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2159 2160 while (!exitPending()) 2161 { 2162 bool rampVolume; 2163 uint16_t leftVol; 2164 uint16_t rightVol; 2165 Vector< sp<EffectChain> > effectChains; 2166 2167 processConfigEvents(); 2168 2169 mixerStatus = MIXER_IDLE; 2170 2171 { // scope for the mLock 2172 2173 Mutex::Autolock _l(mLock); 2174 2175 if (checkForNewParameters_l()) { 2176 mixBufferSize = mFrameCount*mFrameSize; 2177 activeSleepTime = activeSleepTimeUs(); 2178 idleSleepTime = idleSleepTimeUs(); 2179 standbyDelay = microseconds(activeSleepTime*2); 2180 } 2181 2182 // put audio hardware into standby after short delay 2183 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2184 mSuspended) { 2185 // wait until we have something to do... 2186 if (!mStandby) { 2187 LOGV("Audio hardware entering standby, mixer %p\n", this); 2188 mOutput->standby(); 2189 mStandby = true; 2190 mBytesWritten = 0; 2191 } 2192 2193 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2194 // we're about to wait, flush the binder command buffer 2195 IPCThreadState::self()->flushCommands(); 2196 2197 if (exitPending()) break; 2198 2199 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2200 mWaitWorkCV.wait(mLock); 2201 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2202 2203 if (mMasterMute == false) { 2204 char value[PROPERTY_VALUE_MAX]; 2205 property_get("ro.audio.silent", value, "0"); 2206 if (atoi(value)) { 2207 LOGD("Silence is golden"); 2208 setMasterMute(true); 2209 } 2210 } 2211 2212 standbyTime = systemTime() + standbyDelay; 2213 sleepTime = idleSleepTime; 2214 continue; 2215 } 2216 } 2217 2218 effectChains = mEffectChains; 2219 2220 // find out which tracks need to be processed 2221 if (mActiveTracks.size() != 0) { 2222 sp<Track> t = mActiveTracks[0].promote(); 2223 if (t == 0) continue; 2224 2225 Track* const track = t.get(); 2226 audio_track_cblk_t* cblk = track->cblk(); 2227 2228 // The first time a track is added we wait 2229 // for all its buffers to be filled before processing it 2230 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 2231 !track->isPaused() && !track->isTerminated()) 2232 { 2233 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2234 2235 if (track->mFillingUpStatus == Track::FS_FILLED) { 2236 track->mFillingUpStatus = Track::FS_ACTIVE; 2237 mLeftVolFloat = mRightVolFloat = 0; 2238 mLeftVolShort = mRightVolShort = 0; 2239 if (track->mState == TrackBase::RESUMING) { 2240 track->mState = TrackBase::ACTIVE; 2241 rampVolume = true; 2242 } 2243 } else if (cblk->server != 0) { 2244 // If the track is stopped before the first frame was mixed, 2245 // do not apply ramp 2246 rampVolume = true; 2247 } 2248 // compute volume for this track 2249 float left, right; 2250 if (track->isMuted() || mMasterMute || track->isPausing() || 2251 mStreamTypes[track->type()].mute) { 2252 left = right = 0; 2253 if (track->isPausing()) { 2254 track->setPaused(); 2255 } 2256 } else { 2257 float typeVolume = mStreamTypes[track->type()].volume; 2258 float v = mMasterVolume * typeVolume; 2259 float v_clamped = v * cblk->volume[0]; 2260 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2261 left = v_clamped/MAX_GAIN; 2262 v_clamped = v * cblk->volume[1]; 2263 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2264 right = v_clamped/MAX_GAIN; 2265 } 2266 2267 if (left != mLeftVolFloat || right != mRightVolFloat) { 2268 mLeftVolFloat = left; 2269 mRightVolFloat = right; 2270 2271 // If audio HAL implements volume control, 2272 // force software volume to nominal value 2273 if (mOutput->setVolume(left, right) == NO_ERROR) { 2274 left = 1.0f; 2275 right = 1.0f; 2276 } 2277 2278 // Convert volumes from float to 8.24 2279 uint32_t vl = (uint32_t)(left * (1 << 24)); 2280 uint32_t vr = (uint32_t)(right * (1 << 24)); 2281 2282 // Delegate volume control to effect in track effect chain if needed 2283 // only one effect chain can be present on DirectOutputThread, so if 2284 // there is one, the track is connected to it 2285 if (!effectChains.isEmpty()) { 2286 // Do not ramp volume is volume is controlled by effect 2287 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2288 rampVolume = false; 2289 } 2290 } 2291 2292 // Convert volumes from 8.24 to 4.12 format 2293 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2294 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2295 leftVol = (uint16_t)v_clamped; 2296 v_clamped = (vr + (1 << 11)) >> 12; 2297 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2298 rightVol = (uint16_t)v_clamped; 2299 } else { 2300 leftVol = mLeftVolShort; 2301 rightVol = mRightVolShort; 2302 rampVolume = false; 2303 } 2304 2305 // reset retry count 2306 track->mRetryCount = kMaxTrackRetriesDirect; 2307 activeTrack = t; 2308 mixerStatus = MIXER_TRACKS_READY; 2309 } else { 2310 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2311 if (track->isStopped()) { 2312 track->reset(); 2313 } 2314 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2315 // We have consumed all the buffers of this track. 2316 // Remove it from the list of active tracks. 2317 trackToRemove = track; 2318 } else { 2319 // No buffers for this track. Give it a few chances to 2320 // fill a buffer, then remove it from active list. 2321 if (--(track->mRetryCount) <= 0) { 2322 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2323 trackToRemove = track; 2324 } else { 2325 mixerStatus = MIXER_TRACKS_ENABLED; 2326 } 2327 } 2328 } 2329 } 2330 2331 // remove all the tracks that need to be... 2332 if (UNLIKELY(trackToRemove != 0)) { 2333 mActiveTracks.remove(trackToRemove); 2334 if (!effectChains.isEmpty()) { 2335 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2336 trackToRemove->sessionId()); 2337 effectChains[0]->stopTrack(); 2338 } 2339 if (trackToRemove->isTerminated()) { 2340 mTracks.remove(trackToRemove); 2341 deleteTrackName_l(trackToRemove->mName); 2342 } 2343 } 2344 2345 lockEffectChains_l(effectChains); 2346 } 2347 2348 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2349 AudioBufferProvider::Buffer buffer; 2350 size_t frameCount = mFrameCount; 2351 curBuf = (int8_t *)mMixBuffer; 2352 // output audio to hardware 2353 while (frameCount) { 2354 buffer.frameCount = frameCount; 2355 activeTrack->getNextBuffer(&buffer); 2356 if (UNLIKELY(buffer.raw == 0)) { 2357 memset(curBuf, 0, frameCount * mFrameSize); 2358 break; 2359 } 2360 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2361 frameCount -= buffer.frameCount; 2362 curBuf += buffer.frameCount * mFrameSize; 2363 activeTrack->releaseBuffer(&buffer); 2364 } 2365 sleepTime = 0; 2366 standbyTime = systemTime() + standbyDelay; 2367 } else { 2368 if (sleepTime == 0) { 2369 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2370 sleepTime = activeSleepTime; 2371 } else { 2372 sleepTime = idleSleepTime; 2373 } 2374 } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { 2375 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2376 sleepTime = 0; 2377 } 2378 } 2379 2380 if (mSuspended) { 2381 sleepTime = suspendSleepTimeUs(); 2382 } 2383 // sleepTime == 0 means we must write to audio hardware 2384 if (sleepTime == 0) { 2385 if (mixerStatus == MIXER_TRACKS_READY) { 2386 applyVolume(leftVol, rightVol, rampVolume); 2387 } 2388 for (size_t i = 0; i < effectChains.size(); i ++) { 2389 effectChains[i]->process_l(); 2390 } 2391 unlockEffectChains(effectChains); 2392 2393 mLastWriteTime = systemTime(); 2394 mInWrite = true; 2395 mBytesWritten += mixBufferSize; 2396 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 2397 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2398 mNumWrites++; 2399 mInWrite = false; 2400 mStandby = false; 2401 } else { 2402 unlockEffectChains(effectChains); 2403 usleep(sleepTime); 2404 } 2405 2406 // finally let go of removed track, without the lock held 2407 // since we can't guarantee the destructors won't acquire that 2408 // same lock. 2409 trackToRemove.clear(); 2410 activeTrack.clear(); 2411 2412 // Effect chains will be actually deleted here if they were removed from 2413 // mEffectChains list during mixing or effects processing 2414 effectChains.clear(); 2415 } 2416 2417 if (!mStandby) { 2418 mOutput->standby(); 2419 } 2420 2421 LOGV("DirectOutputThread %p exiting", this); 2422 return false; 2423} 2424 2425// getTrackName_l() must be called with ThreadBase::mLock held 2426int AudioFlinger::DirectOutputThread::getTrackName_l() 2427{ 2428 return 0; 2429} 2430 2431// deleteTrackName_l() must be called with ThreadBase::mLock held 2432void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2433{ 2434} 2435 2436// checkForNewParameters_l() must be called with ThreadBase::mLock held 2437bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2438{ 2439 bool reconfig = false; 2440 2441 while (!mNewParameters.isEmpty()) { 2442 status_t status = NO_ERROR; 2443 String8 keyValuePair = mNewParameters[0]; 2444 AudioParameter param = AudioParameter(keyValuePair); 2445 int value; 2446 2447 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2448 // do not accept frame count changes if tracks are open as the track buffer 2449 // size depends on frame count and correct behavior would not be garantied 2450 // if frame count is changed after track creation 2451 if (!mTracks.isEmpty()) { 2452 status = INVALID_OPERATION; 2453 } else { 2454 reconfig = true; 2455 } 2456 } 2457 if (status == NO_ERROR) { 2458 status = mOutput->setParameters(keyValuePair); 2459 if (!mStandby && status == INVALID_OPERATION) { 2460 mOutput->standby(); 2461 mStandby = true; 2462 mBytesWritten = 0; 2463 status = mOutput->setParameters(keyValuePair); 2464 } 2465 if (status == NO_ERROR && reconfig) { 2466 readOutputParameters(); 2467 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2468 } 2469 } 2470 2471 mNewParameters.removeAt(0); 2472 2473 mParamStatus = status; 2474 mParamCond.signal(); 2475 mWaitWorkCV.wait(mLock); 2476 } 2477 return reconfig; 2478} 2479 2480uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2481{ 2482 uint32_t time; 2483 if (AudioSystem::isLinearPCM(mFormat)) { 2484 time = (uint32_t)(mOutput->latency() * 1000) / 2; 2485 } else { 2486 time = 10000; 2487 } 2488 return time; 2489} 2490 2491uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2492{ 2493 uint32_t time; 2494 if (AudioSystem::isLinearPCM(mFormat)) { 2495 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2496 } else { 2497 time = 10000; 2498 } 2499 return time; 2500} 2501 2502uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2503{ 2504 uint32_t time; 2505 if (AudioSystem::isLinearPCM(mFormat)) { 2506 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2507 } else { 2508 time = 10000; 2509 } 2510 return time; 2511} 2512 2513 2514// ---------------------------------------------------------------------------- 2515 2516AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2517 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2518{ 2519 mType = PlaybackThread::DUPLICATING; 2520 addOutputTrack(mainThread); 2521} 2522 2523AudioFlinger::DuplicatingThread::~DuplicatingThread() 2524{ 2525 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2526 mOutputTracks[i]->destroy(); 2527 } 2528 mOutputTracks.clear(); 2529} 2530 2531bool AudioFlinger::DuplicatingThread::threadLoop() 2532{ 2533 Vector< sp<Track> > tracksToRemove; 2534 uint32_t mixerStatus = MIXER_IDLE; 2535 nsecs_t standbyTime = systemTime(); 2536 size_t mixBufferSize = mFrameCount*mFrameSize; 2537 SortedVector< sp<OutputTrack> > outputTracks; 2538 uint32_t writeFrames = 0; 2539 uint32_t activeSleepTime = activeSleepTimeUs(); 2540 uint32_t idleSleepTime = idleSleepTimeUs(); 2541 uint32_t sleepTime = idleSleepTime; 2542 Vector< sp<EffectChain> > effectChains; 2543 2544 while (!exitPending()) 2545 { 2546 processConfigEvents(); 2547 2548 mixerStatus = MIXER_IDLE; 2549 { // scope for the mLock 2550 2551 Mutex::Autolock _l(mLock); 2552 2553 if (checkForNewParameters_l()) { 2554 mixBufferSize = mFrameCount*mFrameSize; 2555 updateWaitTime(); 2556 activeSleepTime = activeSleepTimeUs(); 2557 idleSleepTime = idleSleepTimeUs(); 2558 } 2559 2560 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2561 2562 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2563 outputTracks.add(mOutputTracks[i]); 2564 } 2565 2566 // put audio hardware into standby after short delay 2567 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2568 mSuspended) { 2569 if (!mStandby) { 2570 for (size_t i = 0; i < outputTracks.size(); i++) { 2571 outputTracks[i]->stop(); 2572 } 2573 mStandby = true; 2574 mBytesWritten = 0; 2575 } 2576 2577 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2578 // we're about to wait, flush the binder command buffer 2579 IPCThreadState::self()->flushCommands(); 2580 outputTracks.clear(); 2581 2582 if (exitPending()) break; 2583 2584 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2585 mWaitWorkCV.wait(mLock); 2586 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2587 if (mMasterMute == false) { 2588 char value[PROPERTY_VALUE_MAX]; 2589 property_get("ro.audio.silent", value, "0"); 2590 if (atoi(value)) { 2591 LOGD("Silence is golden"); 2592 setMasterMute(true); 2593 } 2594 } 2595 2596 standbyTime = systemTime() + kStandbyTimeInNsecs; 2597 sleepTime = idleSleepTime; 2598 continue; 2599 } 2600 } 2601 2602 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2603 2604 // prevent any changes in effect chain list and in each effect chain 2605 // during mixing and effect process as the audio buffers could be deleted 2606 // or modified if an effect is created or deleted 2607 lockEffectChains_l(effectChains); 2608 } 2609 2610 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2611 // mix buffers... 2612 if (outputsReady(outputTracks)) { 2613 mAudioMixer->process(); 2614 } else { 2615 memset(mMixBuffer, 0, mixBufferSize); 2616 } 2617 sleepTime = 0; 2618 writeFrames = mFrameCount; 2619 } else { 2620 if (sleepTime == 0) { 2621 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2622 sleepTime = activeSleepTime; 2623 } else { 2624 sleepTime = idleSleepTime; 2625 } 2626 } else if (mBytesWritten != 0) { 2627 // flush remaining overflow buffers in output tracks 2628 for (size_t i = 0; i < outputTracks.size(); i++) { 2629 if (outputTracks[i]->isActive()) { 2630 sleepTime = 0; 2631 writeFrames = 0; 2632 memset(mMixBuffer, 0, mixBufferSize); 2633 break; 2634 } 2635 } 2636 } 2637 } 2638 2639 if (mSuspended) { 2640 sleepTime = suspendSleepTimeUs(); 2641 } 2642 // sleepTime == 0 means we must write to audio hardware 2643 if (sleepTime == 0) { 2644 for (size_t i = 0; i < effectChains.size(); i ++) { 2645 effectChains[i]->process_l(); 2646 } 2647 // enable changes in effect chain 2648 unlockEffectChains(effectChains); 2649 2650 standbyTime = systemTime() + kStandbyTimeInNsecs; 2651 for (size_t i = 0; i < outputTracks.size(); i++) { 2652 outputTracks[i]->write(mMixBuffer, writeFrames); 2653 } 2654 mStandby = false; 2655 mBytesWritten += mixBufferSize; 2656 } else { 2657 // enable changes in effect chain 2658 unlockEffectChains(effectChains); 2659 usleep(sleepTime); 2660 } 2661 2662 // finally let go of all our tracks, without the lock held 2663 // since we can't guarantee the destructors won't acquire that 2664 // same lock. 2665 tracksToRemove.clear(); 2666 outputTracks.clear(); 2667 2668 // Effect chains will be actually deleted here if they were removed from 2669 // mEffectChains list during mixing or effects processing 2670 effectChains.clear(); 2671 } 2672 2673 return false; 2674} 2675 2676void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2677{ 2678 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2679 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2680 this, 2681 mSampleRate, 2682 mFormat, 2683 mChannelCount, 2684 frameCount); 2685 if (outputTrack->cblk() != NULL) { 2686 thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); 2687 mOutputTracks.add(outputTrack); 2688 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2689 updateWaitTime(); 2690 } 2691} 2692 2693void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2694{ 2695 Mutex::Autolock _l(mLock); 2696 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2697 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2698 mOutputTracks[i]->destroy(); 2699 mOutputTracks.removeAt(i); 2700 updateWaitTime(); 2701 return; 2702 } 2703 } 2704 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2705} 2706 2707void AudioFlinger::DuplicatingThread::updateWaitTime() 2708{ 2709 mWaitTimeMs = UINT_MAX; 2710 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2711 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2712 if (strong != NULL) { 2713 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2714 if (waitTimeMs < mWaitTimeMs) { 2715 mWaitTimeMs = waitTimeMs; 2716 } 2717 } 2718 } 2719} 2720 2721 2722bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2723{ 2724 for (size_t i = 0; i < outputTracks.size(); i++) { 2725 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2726 if (thread == 0) { 2727 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2728 return false; 2729 } 2730 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2731 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2732 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2733 return false; 2734 } 2735 } 2736 return true; 2737} 2738 2739uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2740{ 2741 return (mWaitTimeMs * 1000) / 2; 2742} 2743 2744// ---------------------------------------------------------------------------- 2745 2746// TrackBase constructor must be called with AudioFlinger::mLock held 2747AudioFlinger::ThreadBase::TrackBase::TrackBase( 2748 const wp<ThreadBase>& thread, 2749 const sp<Client>& client, 2750 uint32_t sampleRate, 2751 int format, 2752 int channelCount, 2753 int frameCount, 2754 uint32_t flags, 2755 const sp<IMemory>& sharedBuffer, 2756 int sessionId) 2757 : RefBase(), 2758 mThread(thread), 2759 mClient(client), 2760 mCblk(0), 2761 mFrameCount(0), 2762 mState(IDLE), 2763 mClientTid(-1), 2764 mFormat(format), 2765 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2766 mSessionId(sessionId) 2767{ 2768 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2769 2770 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2771 size_t size = sizeof(audio_track_cblk_t); 2772 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2773 if (sharedBuffer == 0) { 2774 size += bufferSize; 2775 } 2776 2777 if (client != NULL) { 2778 mCblkMemory = client->heap()->allocate(size); 2779 if (mCblkMemory != 0) { 2780 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2781 if (mCblk) { // construct the shared structure in-place. 2782 new(mCblk) audio_track_cblk_t(); 2783 // clear all buffers 2784 mCblk->frameCount = frameCount; 2785 mCblk->sampleRate = sampleRate; 2786 mCblk->channelCount = (uint8_t)channelCount; 2787 if (sharedBuffer == 0) { 2788 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2789 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2790 // Force underrun condition to avoid false underrun callback until first data is 2791 // written to buffer 2792 mCblk->flags = CBLK_UNDERRUN_ON; 2793 } else { 2794 mBuffer = sharedBuffer->pointer(); 2795 } 2796 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2797 } 2798 } else { 2799 LOGE("not enough memory for AudioTrack size=%u", size); 2800 client->heap()->dump("AudioTrack"); 2801 return; 2802 } 2803 } else { 2804 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2805 if (mCblk) { // construct the shared structure in-place. 2806 new(mCblk) audio_track_cblk_t(); 2807 // clear all buffers 2808 mCblk->frameCount = frameCount; 2809 mCblk->sampleRate = sampleRate; 2810 mCblk->channelCount = (uint8_t)channelCount; 2811 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2812 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2813 // Force underrun condition to avoid false underrun callback until first data is 2814 // written to buffer 2815 mCblk->flags = CBLK_UNDERRUN_ON; 2816 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2817 } 2818 } 2819} 2820 2821AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2822{ 2823 if (mCblk) { 2824 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2825 if (mClient == NULL) { 2826 delete mCblk; 2827 } 2828 } 2829 mCblkMemory.clear(); // and free the shared memory 2830 if (mClient != NULL) { 2831 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2832 mClient.clear(); 2833 } 2834} 2835 2836void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2837{ 2838 buffer->raw = 0; 2839 mFrameCount = buffer->frameCount; 2840 step(); 2841 buffer->frameCount = 0; 2842} 2843 2844bool AudioFlinger::ThreadBase::TrackBase::step() { 2845 bool result; 2846 audio_track_cblk_t* cblk = this->cblk(); 2847 2848 result = cblk->stepServer(mFrameCount); 2849 if (!result) { 2850 LOGV("stepServer failed acquiring cblk mutex"); 2851 mFlags |= STEPSERVER_FAILED; 2852 } 2853 return result; 2854} 2855 2856void AudioFlinger::ThreadBase::TrackBase::reset() { 2857 audio_track_cblk_t* cblk = this->cblk(); 2858 2859 cblk->user = 0; 2860 cblk->server = 0; 2861 cblk->userBase = 0; 2862 cblk->serverBase = 0; 2863 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2864 LOGV("TrackBase::reset"); 2865} 2866 2867sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2868{ 2869 return mCblkMemory; 2870} 2871 2872int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2873 return (int)mCblk->sampleRate; 2874} 2875 2876int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2877 return (int)mCblk->channelCount; 2878} 2879 2880void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2881 audio_track_cblk_t* cblk = this->cblk(); 2882 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2883 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2884 2885 // Check validity of returned pointer in case the track control block would have been corrupted. 2886 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2887 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2888 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2889 server %d, serverBase %d, user %d, userBase %d, channelCount %d", 2890 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2891 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); 2892 return 0; 2893 } 2894 2895 return bufferStart; 2896} 2897 2898// ---------------------------------------------------------------------------- 2899 2900// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2901AudioFlinger::PlaybackThread::Track::Track( 2902 const wp<ThreadBase>& thread, 2903 const sp<Client>& client, 2904 int streamType, 2905 uint32_t sampleRate, 2906 int format, 2907 int channelCount, 2908 int frameCount, 2909 const sp<IMemory>& sharedBuffer, 2910 int sessionId) 2911 : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), 2912 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 2913 mAuxEffectId(0), mHasVolumeController(false) 2914{ 2915 if (mCblk != NULL) { 2916 sp<ThreadBase> baseThread = thread.promote(); 2917 if (baseThread != 0) { 2918 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2919 mName = playbackThread->getTrackName_l(); 2920 mMainBuffer = playbackThread->mixBuffer(); 2921 } 2922 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2923 if (mName < 0) { 2924 LOGE("no more track names available"); 2925 } 2926 mVolume[0] = 1.0f; 2927 mVolume[1] = 1.0f; 2928 mStreamType = streamType; 2929 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2930 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2931 mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); 2932 } 2933} 2934 2935AudioFlinger::PlaybackThread::Track::~Track() 2936{ 2937 LOGV("PlaybackThread::Track destructor"); 2938 sp<ThreadBase> thread = mThread.promote(); 2939 if (thread != 0) { 2940 Mutex::Autolock _l(thread->mLock); 2941 mState = TERMINATED; 2942 } 2943} 2944 2945void AudioFlinger::PlaybackThread::Track::destroy() 2946{ 2947 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2948 // by removing it from mTracks vector, so there is a risk that this Tracks's 2949 // desctructor is called. As the destructor needs to lock mLock, 2950 // we must acquire a strong reference on this Track before locking mLock 2951 // here so that the destructor is called only when exiting this function. 2952 // On the other hand, as long as Track::destroy() is only called by 2953 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2954 // this Track with its member mTrack. 2955 sp<Track> keep(this); 2956 { // scope for mLock 2957 sp<ThreadBase> thread = mThread.promote(); 2958 if (thread != 0) { 2959 if (!isOutputTrack()) { 2960 if (mState == ACTIVE || mState == RESUMING) { 2961 AudioSystem::stopOutput(thread->id(), 2962 (AudioSystem::stream_type)mStreamType, 2963 mSessionId); 2964 } 2965 AudioSystem::releaseOutput(thread->id()); 2966 } 2967 Mutex::Autolock _l(thread->mLock); 2968 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2969 playbackThread->destroyTrack_l(this); 2970 } 2971 } 2972} 2973 2974void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 2975{ 2976 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", 2977 mName - AudioMixer::TRACK0, 2978 (mClient == NULL) ? getpid() : mClient->pid(), 2979 mStreamType, 2980 mFormat, 2981 mCblk->channelCount, 2982 mSessionId, 2983 mFrameCount, 2984 mState, 2985 mMute, 2986 mFillingUpStatus, 2987 mCblk->sampleRate, 2988 mCblk->volume[0], 2989 mCblk->volume[1], 2990 mCblk->server, 2991 mCblk->user, 2992 (int)mMainBuffer, 2993 (int)mAuxBuffer); 2994} 2995 2996status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 2997{ 2998 audio_track_cblk_t* cblk = this->cblk(); 2999 uint32_t framesReady; 3000 uint32_t framesReq = buffer->frameCount; 3001 3002 // Check if last stepServer failed, try to step now 3003 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3004 if (!step()) goto getNextBuffer_exit; 3005 LOGV("stepServer recovered"); 3006 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3007 } 3008 3009 framesReady = cblk->framesReady(); 3010 3011 if (LIKELY(framesReady)) { 3012 uint32_t s = cblk->server; 3013 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3014 3015 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3016 if (framesReq > framesReady) { 3017 framesReq = framesReady; 3018 } 3019 if (s + framesReq > bufferEnd) { 3020 framesReq = bufferEnd - s; 3021 } 3022 3023 buffer->raw = getBuffer(s, framesReq); 3024 if (buffer->raw == 0) goto getNextBuffer_exit; 3025 3026 buffer->frameCount = framesReq; 3027 return NO_ERROR; 3028 } 3029 3030getNextBuffer_exit: 3031 buffer->raw = 0; 3032 buffer->frameCount = 0; 3033 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3034 return NOT_ENOUGH_DATA; 3035} 3036 3037bool AudioFlinger::PlaybackThread::Track::isReady() const { 3038 if (mFillingUpStatus != FS_FILLING) return true; 3039 3040 if (mCblk->framesReady() >= mCblk->frameCount || 3041 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3042 mFillingUpStatus = FS_FILLED; 3043 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3044 return true; 3045 } 3046 return false; 3047} 3048 3049status_t AudioFlinger::PlaybackThread::Track::start() 3050{ 3051 status_t status = NO_ERROR; 3052 LOGV("start(%d), calling thread %d session %d", 3053 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3054 sp<ThreadBase> thread = mThread.promote(); 3055 if (thread != 0) { 3056 Mutex::Autolock _l(thread->mLock); 3057 int state = mState; 3058 // here the track could be either new, or restarted 3059 // in both cases "unstop" the track 3060 if (mState == PAUSED) { 3061 mState = TrackBase::RESUMING; 3062 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3063 } else { 3064 mState = TrackBase::ACTIVE; 3065 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3066 } 3067 3068 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3069 thread->mLock.unlock(); 3070 status = AudioSystem::startOutput(thread->id(), 3071 (AudioSystem::stream_type)mStreamType, 3072 mSessionId); 3073 thread->mLock.lock(); 3074 } 3075 if (status == NO_ERROR) { 3076 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3077 playbackThread->addTrack_l(this); 3078 } else { 3079 mState = state; 3080 } 3081 } else { 3082 status = BAD_VALUE; 3083 } 3084 return status; 3085} 3086 3087void AudioFlinger::PlaybackThread::Track::stop() 3088{ 3089 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3090 sp<ThreadBase> thread = mThread.promote(); 3091 if (thread != 0) { 3092 Mutex::Autolock _l(thread->mLock); 3093 int state = mState; 3094 if (mState > STOPPED) { 3095 mState = STOPPED; 3096 // If the track is not active (PAUSED and buffers full), flush buffers 3097 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3098 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3099 reset(); 3100 } 3101 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3102 } 3103 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3104 thread->mLock.unlock(); 3105 AudioSystem::stopOutput(thread->id(), 3106 (AudioSystem::stream_type)mStreamType, 3107 mSessionId); 3108 thread->mLock.lock(); 3109 } 3110 } 3111} 3112 3113void AudioFlinger::PlaybackThread::Track::pause() 3114{ 3115 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3116 sp<ThreadBase> thread = mThread.promote(); 3117 if (thread != 0) { 3118 Mutex::Autolock _l(thread->mLock); 3119 if (mState == ACTIVE || mState == RESUMING) { 3120 mState = PAUSING; 3121 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3122 if (!isOutputTrack()) { 3123 thread->mLock.unlock(); 3124 AudioSystem::stopOutput(thread->id(), 3125 (AudioSystem::stream_type)mStreamType, 3126 mSessionId); 3127 thread->mLock.lock(); 3128 } 3129 } 3130 } 3131} 3132 3133void AudioFlinger::PlaybackThread::Track::flush() 3134{ 3135 LOGV("flush(%d)", mName); 3136 sp<ThreadBase> thread = mThread.promote(); 3137 if (thread != 0) { 3138 Mutex::Autolock _l(thread->mLock); 3139 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3140 return; 3141 } 3142 // No point remaining in PAUSED state after a flush => go to 3143 // STOPPED state 3144 mState = STOPPED; 3145 3146 mCblk->lock.lock(); 3147 // NOTE: reset() will reset cblk->user and cblk->server with 3148 // the risk that at the same time, the AudioMixer is trying to read 3149 // data. In this case, getNextBuffer() would return a NULL pointer 3150 // as audio buffer => the AudioMixer code MUST always test that pointer 3151 // returned by getNextBuffer() is not NULL! 3152 reset(); 3153 mCblk->lock.unlock(); 3154 } 3155} 3156 3157void AudioFlinger::PlaybackThread::Track::reset() 3158{ 3159 // Do not reset twice to avoid discarding data written just after a flush and before 3160 // the audioflinger thread detects the track is stopped. 3161 if (!mResetDone) { 3162 TrackBase::reset(); 3163 // Force underrun condition to avoid false underrun callback until first data is 3164 // written to buffer 3165 mCblk->flags |= CBLK_UNDERRUN_ON; 3166 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3167 mFillingUpStatus = FS_FILLING; 3168 mResetDone = true; 3169 } 3170} 3171 3172void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3173{ 3174 mMute = muted; 3175} 3176 3177void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3178{ 3179 mVolume[0] = left; 3180 mVolume[1] = right; 3181} 3182 3183status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3184{ 3185 status_t status = DEAD_OBJECT; 3186 sp<ThreadBase> thread = mThread.promote(); 3187 if (thread != 0) { 3188 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3189 status = playbackThread->attachAuxEffect(this, EffectId); 3190 } 3191 return status; 3192} 3193 3194void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3195{ 3196 mAuxEffectId = EffectId; 3197 mAuxBuffer = buffer; 3198} 3199 3200// ---------------------------------------------------------------------------- 3201 3202// RecordTrack constructor must be called with AudioFlinger::mLock held 3203AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3204 const wp<ThreadBase>& thread, 3205 const sp<Client>& client, 3206 uint32_t sampleRate, 3207 int format, 3208 int channelCount, 3209 int frameCount, 3210 uint32_t flags, 3211 int sessionId) 3212 : TrackBase(thread, client, sampleRate, format, 3213 channelCount, frameCount, flags, 0, sessionId), 3214 mOverflow(false) 3215{ 3216 if (mCblk != NULL) { 3217 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3218 if (format == AudioSystem::PCM_16_BIT) { 3219 mCblk->frameSize = channelCount * sizeof(int16_t); 3220 } else if (format == AudioSystem::PCM_8_BIT) { 3221 mCblk->frameSize = channelCount * sizeof(int8_t); 3222 } else { 3223 mCblk->frameSize = sizeof(int8_t); 3224 } 3225 } 3226} 3227 3228AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3229{ 3230 sp<ThreadBase> thread = mThread.promote(); 3231 if (thread != 0) { 3232 AudioSystem::releaseInput(thread->id()); 3233 } 3234} 3235 3236status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3237{ 3238 audio_track_cblk_t* cblk = this->cblk(); 3239 uint32_t framesAvail; 3240 uint32_t framesReq = buffer->frameCount; 3241 3242 // Check if last stepServer failed, try to step now 3243 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3244 if (!step()) goto getNextBuffer_exit; 3245 LOGV("stepServer recovered"); 3246 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3247 } 3248 3249 framesAvail = cblk->framesAvailable_l(); 3250 3251 if (LIKELY(framesAvail)) { 3252 uint32_t s = cblk->server; 3253 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3254 3255 if (framesReq > framesAvail) { 3256 framesReq = framesAvail; 3257 } 3258 if (s + framesReq > bufferEnd) { 3259 framesReq = bufferEnd - s; 3260 } 3261 3262 buffer->raw = getBuffer(s, framesReq); 3263 if (buffer->raw == 0) goto getNextBuffer_exit; 3264 3265 buffer->frameCount = framesReq; 3266 return NO_ERROR; 3267 } 3268 3269getNextBuffer_exit: 3270 buffer->raw = 0; 3271 buffer->frameCount = 0; 3272 return NOT_ENOUGH_DATA; 3273} 3274 3275status_t AudioFlinger::RecordThread::RecordTrack::start() 3276{ 3277 sp<ThreadBase> thread = mThread.promote(); 3278 if (thread != 0) { 3279 RecordThread *recordThread = (RecordThread *)thread.get(); 3280 return recordThread->start(this); 3281 } else { 3282 return BAD_VALUE; 3283 } 3284} 3285 3286void AudioFlinger::RecordThread::RecordTrack::stop() 3287{ 3288 sp<ThreadBase> thread = mThread.promote(); 3289 if (thread != 0) { 3290 RecordThread *recordThread = (RecordThread *)thread.get(); 3291 recordThread->stop(this); 3292 TrackBase::reset(); 3293 // Force overerrun condition to avoid false overrun callback until first data is 3294 // read from buffer 3295 mCblk->flags |= CBLK_UNDERRUN_ON; 3296 } 3297} 3298 3299void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3300{ 3301 snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", 3302 (mClient == NULL) ? getpid() : mClient->pid(), 3303 mFormat, 3304 mCblk->channelCount, 3305 mSessionId, 3306 mFrameCount, 3307 mState, 3308 mCblk->sampleRate, 3309 mCblk->server, 3310 mCblk->user); 3311} 3312 3313 3314// ---------------------------------------------------------------------------- 3315 3316AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3317 const wp<ThreadBase>& thread, 3318 DuplicatingThread *sourceThread, 3319 uint32_t sampleRate, 3320 int format, 3321 int channelCount, 3322 int frameCount) 3323 : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), 3324 mActive(false), mSourceThread(sourceThread) 3325{ 3326 3327 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3328 if (mCblk != NULL) { 3329 mCblk->flags |= CBLK_DIRECTION_OUT; 3330 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3331 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3332 mOutBuffer.frameCount = 0; 3333 playbackThread->mTracks.add(this); 3334 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", 3335 mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); 3336 } else { 3337 LOGW("Error creating output track on thread %p", playbackThread); 3338 } 3339} 3340 3341AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3342{ 3343 clearBufferQueue(); 3344} 3345 3346status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3347{ 3348 status_t status = Track::start(); 3349 if (status != NO_ERROR) { 3350 return status; 3351 } 3352 3353 mActive = true; 3354 mRetryCount = 127; 3355 return status; 3356} 3357 3358void AudioFlinger::PlaybackThread::OutputTrack::stop() 3359{ 3360 Track::stop(); 3361 clearBufferQueue(); 3362 mOutBuffer.frameCount = 0; 3363 mActive = false; 3364} 3365 3366bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3367{ 3368 Buffer *pInBuffer; 3369 Buffer inBuffer; 3370 uint32_t channelCount = mCblk->channelCount; 3371 bool outputBufferFull = false; 3372 inBuffer.frameCount = frames; 3373 inBuffer.i16 = data; 3374 3375 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3376 3377 if (!mActive && frames != 0) { 3378 start(); 3379 sp<ThreadBase> thread = mThread.promote(); 3380 if (thread != 0) { 3381 MixerThread *mixerThread = (MixerThread *)thread.get(); 3382 if (mCblk->frameCount > frames){ 3383 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3384 uint32_t startFrames = (mCblk->frameCount - frames); 3385 pInBuffer = new Buffer; 3386 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3387 pInBuffer->frameCount = startFrames; 3388 pInBuffer->i16 = pInBuffer->mBuffer; 3389 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3390 mBufferQueue.add(pInBuffer); 3391 } else { 3392 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3393 } 3394 } 3395 } 3396 } 3397 3398 while (waitTimeLeftMs) { 3399 // First write pending buffers, then new data 3400 if (mBufferQueue.size()) { 3401 pInBuffer = mBufferQueue.itemAt(0); 3402 } else { 3403 pInBuffer = &inBuffer; 3404 } 3405 3406 if (pInBuffer->frameCount == 0) { 3407 break; 3408 } 3409 3410 if (mOutBuffer.frameCount == 0) { 3411 mOutBuffer.frameCount = pInBuffer->frameCount; 3412 nsecs_t startTime = systemTime(); 3413 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3414 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3415 outputBufferFull = true; 3416 break; 3417 } 3418 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3419 if (waitTimeLeftMs >= waitTimeMs) { 3420 waitTimeLeftMs -= waitTimeMs; 3421 } else { 3422 waitTimeLeftMs = 0; 3423 } 3424 } 3425 3426 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3427 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3428 mCblk->stepUser(outFrames); 3429 pInBuffer->frameCount -= outFrames; 3430 pInBuffer->i16 += outFrames * channelCount; 3431 mOutBuffer.frameCount -= outFrames; 3432 mOutBuffer.i16 += outFrames * channelCount; 3433 3434 if (pInBuffer->frameCount == 0) { 3435 if (mBufferQueue.size()) { 3436 mBufferQueue.removeAt(0); 3437 delete [] pInBuffer->mBuffer; 3438 delete pInBuffer; 3439 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3440 } else { 3441 break; 3442 } 3443 } 3444 } 3445 3446 // If we could not write all frames, allocate a buffer and queue it for next time. 3447 if (inBuffer.frameCount) { 3448 sp<ThreadBase> thread = mThread.promote(); 3449 if (thread != 0 && !thread->standby()) { 3450 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3451 pInBuffer = new Buffer; 3452 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3453 pInBuffer->frameCount = inBuffer.frameCount; 3454 pInBuffer->i16 = pInBuffer->mBuffer; 3455 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3456 mBufferQueue.add(pInBuffer); 3457 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3458 } else { 3459 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3460 } 3461 } 3462 } 3463 3464 // Calling write() with a 0 length buffer, means that no more data will be written: 3465 // If no more buffers are pending, fill output track buffer to make sure it is started 3466 // by output mixer. 3467 if (frames == 0 && mBufferQueue.size() == 0) { 3468 if (mCblk->user < mCblk->frameCount) { 3469 frames = mCblk->frameCount - mCblk->user; 3470 pInBuffer = new Buffer; 3471 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3472 pInBuffer->frameCount = frames; 3473 pInBuffer->i16 = pInBuffer->mBuffer; 3474 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3475 mBufferQueue.add(pInBuffer); 3476 } else if (mActive) { 3477 stop(); 3478 } 3479 } 3480 3481 return outputBufferFull; 3482} 3483 3484status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3485{ 3486 int active; 3487 status_t result; 3488 audio_track_cblk_t* cblk = mCblk; 3489 uint32_t framesReq = buffer->frameCount; 3490 3491// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3492 buffer->frameCount = 0; 3493 3494 uint32_t framesAvail = cblk->framesAvailable(); 3495 3496 3497 if (framesAvail == 0) { 3498 Mutex::Autolock _l(cblk->lock); 3499 goto start_loop_here; 3500 while (framesAvail == 0) { 3501 active = mActive; 3502 if (UNLIKELY(!active)) { 3503 LOGV("Not active and NO_MORE_BUFFERS"); 3504 return AudioTrack::NO_MORE_BUFFERS; 3505 } 3506 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3507 if (result != NO_ERROR) { 3508 return AudioTrack::NO_MORE_BUFFERS; 3509 } 3510 // read the server count again 3511 start_loop_here: 3512 framesAvail = cblk->framesAvailable_l(); 3513 } 3514 } 3515 3516// if (framesAvail < framesReq) { 3517// return AudioTrack::NO_MORE_BUFFERS; 3518// } 3519 3520 if (framesReq > framesAvail) { 3521 framesReq = framesAvail; 3522 } 3523 3524 uint32_t u = cblk->user; 3525 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3526 3527 if (u + framesReq > bufferEnd) { 3528 framesReq = bufferEnd - u; 3529 } 3530 3531 buffer->frameCount = framesReq; 3532 buffer->raw = (void *)cblk->buffer(u); 3533 return NO_ERROR; 3534} 3535 3536 3537void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3538{ 3539 size_t size = mBufferQueue.size(); 3540 Buffer *pBuffer; 3541 3542 for (size_t i = 0; i < size; i++) { 3543 pBuffer = mBufferQueue.itemAt(i); 3544 delete [] pBuffer->mBuffer; 3545 delete pBuffer; 3546 } 3547 mBufferQueue.clear(); 3548} 3549 3550// ---------------------------------------------------------------------------- 3551 3552AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3553 : RefBase(), 3554 mAudioFlinger(audioFlinger), 3555 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3556 mPid(pid) 3557{ 3558 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3559} 3560 3561// Client destructor must be called with AudioFlinger::mLock held 3562AudioFlinger::Client::~Client() 3563{ 3564 mAudioFlinger->removeClient_l(mPid); 3565} 3566 3567const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3568{ 3569 return mMemoryDealer; 3570} 3571 3572// ---------------------------------------------------------------------------- 3573 3574AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3575 const sp<IAudioFlingerClient>& client, 3576 pid_t pid) 3577 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3578{ 3579} 3580 3581AudioFlinger::NotificationClient::~NotificationClient() 3582{ 3583 mClient.clear(); 3584} 3585 3586void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3587{ 3588 sp<NotificationClient> keep(this); 3589 { 3590 mAudioFlinger->removeNotificationClient(mPid); 3591 } 3592} 3593 3594// ---------------------------------------------------------------------------- 3595 3596AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3597 : BnAudioTrack(), 3598 mTrack(track) 3599{ 3600} 3601 3602AudioFlinger::TrackHandle::~TrackHandle() { 3603 // just stop the track on deletion, associated resources 3604 // will be freed from the main thread once all pending buffers have 3605 // been played. Unless it's not in the active track list, in which 3606 // case we free everything now... 3607 mTrack->destroy(); 3608} 3609 3610status_t AudioFlinger::TrackHandle::start() { 3611 return mTrack->start(); 3612} 3613 3614void AudioFlinger::TrackHandle::stop() { 3615 mTrack->stop(); 3616} 3617 3618void AudioFlinger::TrackHandle::flush() { 3619 mTrack->flush(); 3620} 3621 3622void AudioFlinger::TrackHandle::mute(bool e) { 3623 mTrack->mute(e); 3624} 3625 3626void AudioFlinger::TrackHandle::pause() { 3627 mTrack->pause(); 3628} 3629 3630void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3631 mTrack->setVolume(left, right); 3632} 3633 3634sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3635 return mTrack->getCblk(); 3636} 3637 3638status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3639{ 3640 return mTrack->attachAuxEffect(EffectId); 3641} 3642 3643status_t AudioFlinger::TrackHandle::onTransact( 3644 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3645{ 3646 return BnAudioTrack::onTransact(code, data, reply, flags); 3647} 3648 3649// ---------------------------------------------------------------------------- 3650 3651sp<IAudioRecord> AudioFlinger::openRecord( 3652 pid_t pid, 3653 int input, 3654 uint32_t sampleRate, 3655 int format, 3656 int channelCount, 3657 int frameCount, 3658 uint32_t flags, 3659 int *sessionId, 3660 status_t *status) 3661{ 3662 sp<RecordThread::RecordTrack> recordTrack; 3663 sp<RecordHandle> recordHandle; 3664 sp<Client> client; 3665 wp<Client> wclient; 3666 status_t lStatus; 3667 RecordThread *thread; 3668 size_t inFrameCount; 3669 int lSessionId; 3670 3671 // check calling permissions 3672 if (!recordingAllowed()) { 3673 lStatus = PERMISSION_DENIED; 3674 goto Exit; 3675 } 3676 3677 // add client to list 3678 { // scope for mLock 3679 Mutex::Autolock _l(mLock); 3680 thread = checkRecordThread_l(input); 3681 if (thread == NULL) { 3682 lStatus = BAD_VALUE; 3683 goto Exit; 3684 } 3685 3686 wclient = mClients.valueFor(pid); 3687 if (wclient != NULL) { 3688 client = wclient.promote(); 3689 } else { 3690 client = new Client(this, pid); 3691 mClients.add(pid, client); 3692 } 3693 3694 // If no audio session id is provided, create one here 3695 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 3696 lSessionId = *sessionId; 3697 } else { 3698 lSessionId = nextUniqueId(); 3699 if (sessionId != NULL) { 3700 *sessionId = lSessionId; 3701 } 3702 } 3703 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3704 recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, 3705 format, channelCount, frameCount, flags, lSessionId); 3706 } 3707 if (recordTrack->getCblk() == NULL) { 3708 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3709 // destructor is called by the TrackBase destructor with mLock held 3710 client.clear(); 3711 recordTrack.clear(); 3712 lStatus = NO_MEMORY; 3713 goto Exit; 3714 } 3715 3716 // return to handle to client 3717 recordHandle = new RecordHandle(recordTrack); 3718 lStatus = NO_ERROR; 3719 3720Exit: 3721 if (status) { 3722 *status = lStatus; 3723 } 3724 return recordHandle; 3725} 3726 3727// ---------------------------------------------------------------------------- 3728 3729AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3730 : BnAudioRecord(), 3731 mRecordTrack(recordTrack) 3732{ 3733} 3734 3735AudioFlinger::RecordHandle::~RecordHandle() { 3736 stop(); 3737} 3738 3739status_t AudioFlinger::RecordHandle::start() { 3740 LOGV("RecordHandle::start()"); 3741 return mRecordTrack->start(); 3742} 3743 3744void AudioFlinger::RecordHandle::stop() { 3745 LOGV("RecordHandle::stop()"); 3746 mRecordTrack->stop(); 3747} 3748 3749sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3750 return mRecordTrack->getCblk(); 3751} 3752 3753status_t AudioFlinger::RecordHandle::onTransact( 3754 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3755{ 3756 return BnAudioRecord::onTransact(code, data, reply, flags); 3757} 3758 3759// ---------------------------------------------------------------------------- 3760 3761AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : 3762 ThreadBase(audioFlinger, id), 3763 mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3764{ 3765 mReqChannelCount = AudioSystem::popCount(channels); 3766 mReqSampleRate = sampleRate; 3767 readInputParameters(); 3768} 3769 3770 3771AudioFlinger::RecordThread::~RecordThread() 3772{ 3773 delete[] mRsmpInBuffer; 3774 if (mResampler != 0) { 3775 delete mResampler; 3776 delete[] mRsmpOutBuffer; 3777 } 3778} 3779 3780void AudioFlinger::RecordThread::onFirstRef() 3781{ 3782 const size_t SIZE = 256; 3783 char buffer[SIZE]; 3784 3785 snprintf(buffer, SIZE, "Record Thread %p", this); 3786 3787 run(buffer, PRIORITY_URGENT_AUDIO); 3788} 3789 3790bool AudioFlinger::RecordThread::threadLoop() 3791{ 3792 AudioBufferProvider::Buffer buffer; 3793 sp<RecordTrack> activeTrack; 3794 3795 // start recording 3796 while (!exitPending()) { 3797 3798 processConfigEvents(); 3799 3800 { // scope for mLock 3801 Mutex::Autolock _l(mLock); 3802 checkForNewParameters_l(); 3803 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3804 if (!mStandby) { 3805 mInput->standby(); 3806 mStandby = true; 3807 } 3808 3809 if (exitPending()) break; 3810 3811 LOGV("RecordThread: loop stopping"); 3812 // go to sleep 3813 mWaitWorkCV.wait(mLock); 3814 LOGV("RecordThread: loop starting"); 3815 continue; 3816 } 3817 if (mActiveTrack != 0) { 3818 if (mActiveTrack->mState == TrackBase::PAUSING) { 3819 if (!mStandby) { 3820 mInput->standby(); 3821 mStandby = true; 3822 } 3823 mActiveTrack.clear(); 3824 mStartStopCond.broadcast(); 3825 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3826 if (mReqChannelCount != mActiveTrack->channelCount()) { 3827 mActiveTrack.clear(); 3828 mStartStopCond.broadcast(); 3829 } else if (mBytesRead != 0) { 3830 // record start succeeds only if first read from audio input 3831 // succeeds 3832 if (mBytesRead > 0) { 3833 mActiveTrack->mState = TrackBase::ACTIVE; 3834 } else { 3835 mActiveTrack.clear(); 3836 } 3837 mStartStopCond.broadcast(); 3838 } 3839 mStandby = false; 3840 } 3841 } 3842 } 3843 3844 if (mActiveTrack != 0) { 3845 if (mActiveTrack->mState != TrackBase::ACTIVE && 3846 mActiveTrack->mState != TrackBase::RESUMING) { 3847 usleep(5000); 3848 continue; 3849 } 3850 buffer.frameCount = mFrameCount; 3851 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3852 size_t framesOut = buffer.frameCount; 3853 if (mResampler == 0) { 3854 // no resampling 3855 while (framesOut) { 3856 size_t framesIn = mFrameCount - mRsmpInIndex; 3857 if (framesIn) { 3858 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3859 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3860 if (framesIn > framesOut) 3861 framesIn = framesOut; 3862 mRsmpInIndex += framesIn; 3863 framesOut -= framesIn; 3864 if ((int)mChannelCount == mReqChannelCount || 3865 mFormat != AudioSystem::PCM_16_BIT) { 3866 memcpy(dst, src, framesIn * mFrameSize); 3867 } else { 3868 int16_t *src16 = (int16_t *)src; 3869 int16_t *dst16 = (int16_t *)dst; 3870 if (mChannelCount == 1) { 3871 while (framesIn--) { 3872 *dst16++ = *src16; 3873 *dst16++ = *src16++; 3874 } 3875 } else { 3876 while (framesIn--) { 3877 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3878 src16 += 2; 3879 } 3880 } 3881 } 3882 } 3883 if (framesOut && mFrameCount == mRsmpInIndex) { 3884 if (framesOut == mFrameCount && 3885 ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { 3886 mBytesRead = mInput->read(buffer.raw, mInputBytes); 3887 framesOut = 0; 3888 } else { 3889 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 3890 mRsmpInIndex = 0; 3891 } 3892 if (mBytesRead < 0) { 3893 LOGE("Error reading audio input"); 3894 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3895 // Force input into standby so that it tries to 3896 // recover at next read attempt 3897 mInput->standby(); 3898 usleep(5000); 3899 } 3900 mRsmpInIndex = mFrameCount; 3901 framesOut = 0; 3902 buffer.frameCount = 0; 3903 } 3904 } 3905 } 3906 } else { 3907 // resampling 3908 3909 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3910 // alter output frame count as if we were expecting stereo samples 3911 if (mChannelCount == 1 && mReqChannelCount == 1) { 3912 framesOut >>= 1; 3913 } 3914 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3915 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3916 // are 32 bit aligned which should be always true. 3917 if (mChannelCount == 2 && mReqChannelCount == 1) { 3918 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3919 // the resampler always outputs stereo samples: do post stereo to mono conversion 3920 int16_t *src = (int16_t *)mRsmpOutBuffer; 3921 int16_t *dst = buffer.i16; 3922 while (framesOut--) { 3923 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3924 src += 2; 3925 } 3926 } else { 3927 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3928 } 3929 3930 } 3931 mActiveTrack->releaseBuffer(&buffer); 3932 mActiveTrack->overflow(); 3933 } 3934 // client isn't retrieving buffers fast enough 3935 else { 3936 if (!mActiveTrack->setOverflow()) 3937 LOGW("RecordThread: buffer overflow"); 3938 // Release the processor for a while before asking for a new buffer. 3939 // This will give the application more chance to read from the buffer and 3940 // clear the overflow. 3941 usleep(5000); 3942 } 3943 } 3944 } 3945 3946 if (!mStandby) { 3947 mInput->standby(); 3948 } 3949 mActiveTrack.clear(); 3950 3951 mStartStopCond.broadcast(); 3952 3953 LOGV("RecordThread %p exiting", this); 3954 return false; 3955} 3956 3957status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 3958{ 3959 LOGV("RecordThread::start"); 3960 sp <ThreadBase> strongMe = this; 3961 status_t status = NO_ERROR; 3962 { 3963 AutoMutex lock(&mLock); 3964 if (mActiveTrack != 0) { 3965 if (recordTrack != mActiveTrack.get()) { 3966 status = -EBUSY; 3967 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3968 mActiveTrack->mState = TrackBase::ACTIVE; 3969 } 3970 return status; 3971 } 3972 3973 recordTrack->mState = TrackBase::IDLE; 3974 mActiveTrack = recordTrack; 3975 mLock.unlock(); 3976 status_t status = AudioSystem::startInput(mId); 3977 mLock.lock(); 3978 if (status != NO_ERROR) { 3979 mActiveTrack.clear(); 3980 return status; 3981 } 3982 mActiveTrack->mState = TrackBase::RESUMING; 3983 mRsmpInIndex = mFrameCount; 3984 mBytesRead = 0; 3985 // signal thread to start 3986 LOGV("Signal record thread"); 3987 mWaitWorkCV.signal(); 3988 // do not wait for mStartStopCond if exiting 3989 if (mExiting) { 3990 mActiveTrack.clear(); 3991 status = INVALID_OPERATION; 3992 goto startError; 3993 } 3994 mStartStopCond.wait(mLock); 3995 if (mActiveTrack == 0) { 3996 LOGV("Record failed to start"); 3997 status = BAD_VALUE; 3998 goto startError; 3999 } 4000 LOGV("Record started OK"); 4001 return status; 4002 } 4003startError: 4004 AudioSystem::stopInput(mId); 4005 return status; 4006} 4007 4008void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4009 LOGV("RecordThread::stop"); 4010 sp <ThreadBase> strongMe = this; 4011 { 4012 AutoMutex lock(&mLock); 4013 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4014 mActiveTrack->mState = TrackBase::PAUSING; 4015 // do not wait for mStartStopCond if exiting 4016 if (mExiting) { 4017 return; 4018 } 4019 mStartStopCond.wait(mLock); 4020 // if we have been restarted, recordTrack == mActiveTrack.get() here 4021 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4022 mLock.unlock(); 4023 AudioSystem::stopInput(mId); 4024 mLock.lock(); 4025 LOGV("Record stopped OK"); 4026 } 4027 } 4028 } 4029} 4030 4031status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4032{ 4033 const size_t SIZE = 256; 4034 char buffer[SIZE]; 4035 String8 result; 4036 pid_t pid = 0; 4037 4038 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4039 result.append(buffer); 4040 4041 if (mActiveTrack != 0) { 4042 result.append("Active Track:\n"); 4043 result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); 4044 mActiveTrack->dump(buffer, SIZE); 4045 result.append(buffer); 4046 4047 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4048 result.append(buffer); 4049 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4050 result.append(buffer); 4051 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4052 result.append(buffer); 4053 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4054 result.append(buffer); 4055 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4056 result.append(buffer); 4057 4058 4059 } else { 4060 result.append("No record client\n"); 4061 } 4062 write(fd, result.string(), result.size()); 4063 4064 dumpBase(fd, args); 4065 4066 return NO_ERROR; 4067} 4068 4069status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4070{ 4071 size_t framesReq = buffer->frameCount; 4072 size_t framesReady = mFrameCount - mRsmpInIndex; 4073 int channelCount; 4074 4075 if (framesReady == 0) { 4076 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 4077 if (mBytesRead < 0) { 4078 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4079 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4080 // Force input into standby so that it tries to 4081 // recover at next read attempt 4082 mInput->standby(); 4083 usleep(5000); 4084 } 4085 buffer->raw = 0; 4086 buffer->frameCount = 0; 4087 return NOT_ENOUGH_DATA; 4088 } 4089 mRsmpInIndex = 0; 4090 framesReady = mFrameCount; 4091 } 4092 4093 if (framesReq > framesReady) { 4094 framesReq = framesReady; 4095 } 4096 4097 if (mChannelCount == 1 && mReqChannelCount == 2) { 4098 channelCount = 1; 4099 } else { 4100 channelCount = 2; 4101 } 4102 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4103 buffer->frameCount = framesReq; 4104 return NO_ERROR; 4105} 4106 4107void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4108{ 4109 mRsmpInIndex += buffer->frameCount; 4110 buffer->frameCount = 0; 4111} 4112 4113bool AudioFlinger::RecordThread::checkForNewParameters_l() 4114{ 4115 bool reconfig = false; 4116 4117 while (!mNewParameters.isEmpty()) { 4118 status_t status = NO_ERROR; 4119 String8 keyValuePair = mNewParameters[0]; 4120 AudioParameter param = AudioParameter(keyValuePair); 4121 int value; 4122 int reqFormat = mFormat; 4123 int reqSamplingRate = mReqSampleRate; 4124 int reqChannelCount = mReqChannelCount; 4125 4126 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4127 reqSamplingRate = value; 4128 reconfig = true; 4129 } 4130 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4131 reqFormat = value; 4132 reconfig = true; 4133 } 4134 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4135 reqChannelCount = AudioSystem::popCount(value); 4136 reconfig = true; 4137 } 4138 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4139 // do not accept frame count changes if tracks are open as the track buffer 4140 // size depends on frame count and correct behavior would not be garantied 4141 // if frame count is changed after track creation 4142 if (mActiveTrack != 0) { 4143 status = INVALID_OPERATION; 4144 } else { 4145 reconfig = true; 4146 } 4147 } 4148 if (status == NO_ERROR) { 4149 status = mInput->setParameters(keyValuePair); 4150 if (status == INVALID_OPERATION) { 4151 mInput->standby(); 4152 status = mInput->setParameters(keyValuePair); 4153 } 4154 if (reconfig) { 4155 if (status == BAD_VALUE && 4156 reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && 4157 ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && 4158 (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { 4159 status = NO_ERROR; 4160 } 4161 if (status == NO_ERROR) { 4162 readInputParameters(); 4163 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4164 } 4165 } 4166 } 4167 4168 mNewParameters.removeAt(0); 4169 4170 mParamStatus = status; 4171 mParamCond.signal(); 4172 mWaitWorkCV.wait(mLock); 4173 } 4174 return reconfig; 4175} 4176 4177String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4178{ 4179 return mInput->getParameters(keys); 4180} 4181 4182void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4183 AudioSystem::OutputDescriptor desc; 4184 void *param2 = 0; 4185 4186 switch (event) { 4187 case AudioSystem::INPUT_OPENED: 4188 case AudioSystem::INPUT_CONFIG_CHANGED: 4189 desc.channels = mChannels; 4190 desc.samplingRate = mSampleRate; 4191 desc.format = mFormat; 4192 desc.frameCount = mFrameCount; 4193 desc.latency = 0; 4194 param2 = &desc; 4195 break; 4196 4197 case AudioSystem::INPUT_CLOSED: 4198 default: 4199 break; 4200 } 4201 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4202} 4203 4204void AudioFlinger::RecordThread::readInputParameters() 4205{ 4206 if (mRsmpInBuffer) delete mRsmpInBuffer; 4207 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4208 if (mResampler) delete mResampler; 4209 mResampler = 0; 4210 4211 mSampleRate = mInput->sampleRate(); 4212 mChannels = mInput->channels(); 4213 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 4214 mFormat = mInput->format(); 4215 mFrameSize = (uint16_t)mInput->frameSize(); 4216 mInputBytes = mInput->bufferSize(); 4217 mFrameCount = mInputBytes / mFrameSize; 4218 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4219 4220 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4221 { 4222 int channelCount; 4223 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4224 // stereo to mono post process as the resampler always outputs stereo. 4225 if (mChannelCount == 1 && mReqChannelCount == 2) { 4226 channelCount = 1; 4227 } else { 4228 channelCount = 2; 4229 } 4230 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4231 mResampler->setSampleRate(mSampleRate); 4232 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4233 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4234 4235 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4236 if (mChannelCount == 1 && mReqChannelCount == 1) { 4237 mFrameCount >>= 1; 4238 } 4239 4240 } 4241 mRsmpInIndex = mFrameCount; 4242} 4243 4244unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4245{ 4246 return mInput->getInputFramesLost(); 4247} 4248 4249// ---------------------------------------------------------------------------- 4250 4251int AudioFlinger::openOutput(uint32_t *pDevices, 4252 uint32_t *pSamplingRate, 4253 uint32_t *pFormat, 4254 uint32_t *pChannels, 4255 uint32_t *pLatencyMs, 4256 uint32_t flags) 4257{ 4258 status_t status; 4259 PlaybackThread *thread = NULL; 4260 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4261 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4262 uint32_t format = pFormat ? *pFormat : 0; 4263 uint32_t channels = pChannels ? *pChannels : 0; 4264 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4265 4266 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4267 pDevices ? *pDevices : 0, 4268 samplingRate, 4269 format, 4270 channels, 4271 flags); 4272 4273 if (pDevices == NULL || *pDevices == 0) { 4274 return 0; 4275 } 4276 Mutex::Autolock _l(mLock); 4277 4278 AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, 4279 (int *)&format, 4280 &channels, 4281 &samplingRate, 4282 &status); 4283 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4284 output, 4285 samplingRate, 4286 format, 4287 channels, 4288 status); 4289 4290 mHardwareStatus = AUDIO_HW_IDLE; 4291 if (output != 0) { 4292 int id = nextUniqueId(); 4293 if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || 4294 (format != AudioSystem::PCM_16_BIT) || 4295 (channels != AudioSystem::CHANNEL_OUT_STEREO)) { 4296 thread = new DirectOutputThread(this, output, id, *pDevices); 4297 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4298 } else { 4299 thread = new MixerThread(this, output, id, *pDevices); 4300 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4301 4302#ifdef LVMX 4303 unsigned bitsPerSample = 4304 (format == AudioSystem::PCM_16_BIT) ? 16 : 4305 ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); 4306 unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; 4307 int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); 4308 4309 LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); 4310 LifeVibes::setDevice(audioOutputType, *pDevices); 4311#endif 4312 4313 } 4314 mPlaybackThreads.add(id, thread); 4315 4316 if (pSamplingRate) *pSamplingRate = samplingRate; 4317 if (pFormat) *pFormat = format; 4318 if (pChannels) *pChannels = channels; 4319 if (pLatencyMs) *pLatencyMs = thread->latency(); 4320 4321 // notify client processes of the new output creation 4322 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4323 return id; 4324 } 4325 4326 return 0; 4327} 4328 4329int AudioFlinger::openDuplicateOutput(int output1, int output2) 4330{ 4331 Mutex::Autolock _l(mLock); 4332 MixerThread *thread1 = checkMixerThread_l(output1); 4333 MixerThread *thread2 = checkMixerThread_l(output2); 4334 4335 if (thread1 == NULL || thread2 == NULL) { 4336 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4337 return 0; 4338 } 4339 4340 int id = nextUniqueId(); 4341 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4342 thread->addOutputTrack(thread2); 4343 mPlaybackThreads.add(id, thread); 4344 // notify client processes of the new output creation 4345 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4346 return id; 4347} 4348 4349status_t AudioFlinger::closeOutput(int output) 4350{ 4351 // keep strong reference on the playback thread so that 4352 // it is not destroyed while exit() is executed 4353 sp <PlaybackThread> thread; 4354 { 4355 Mutex::Autolock _l(mLock); 4356 thread = checkPlaybackThread_l(output); 4357 if (thread == NULL) { 4358 return BAD_VALUE; 4359 } 4360 4361 LOGV("closeOutput() %d", output); 4362 4363 if (thread->type() == PlaybackThread::MIXER) { 4364 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4365 if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { 4366 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4367 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4368 } 4369 } 4370 } 4371 void *param2 = 0; 4372 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4373 mPlaybackThreads.removeItem(output); 4374 } 4375 thread->exit(); 4376 4377 if (thread->type() != PlaybackThread::DUPLICATING) { 4378 mAudioHardware->closeOutputStream(thread->getOutput()); 4379 } 4380 return NO_ERROR; 4381} 4382 4383status_t AudioFlinger::suspendOutput(int output) 4384{ 4385 Mutex::Autolock _l(mLock); 4386 PlaybackThread *thread = checkPlaybackThread_l(output); 4387 4388 if (thread == NULL) { 4389 return BAD_VALUE; 4390 } 4391 4392 LOGV("suspendOutput() %d", output); 4393 thread->suspend(); 4394 4395 return NO_ERROR; 4396} 4397 4398status_t AudioFlinger::restoreOutput(int output) 4399{ 4400 Mutex::Autolock _l(mLock); 4401 PlaybackThread *thread = checkPlaybackThread_l(output); 4402 4403 if (thread == NULL) { 4404 return BAD_VALUE; 4405 } 4406 4407 LOGV("restoreOutput() %d", output); 4408 4409 thread->restore(); 4410 4411 return NO_ERROR; 4412} 4413 4414int AudioFlinger::openInput(uint32_t *pDevices, 4415 uint32_t *pSamplingRate, 4416 uint32_t *pFormat, 4417 uint32_t *pChannels, 4418 uint32_t acoustics) 4419{ 4420 status_t status; 4421 RecordThread *thread = NULL; 4422 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4423 uint32_t format = pFormat ? *pFormat : 0; 4424 uint32_t channels = pChannels ? *pChannels : 0; 4425 uint32_t reqSamplingRate = samplingRate; 4426 uint32_t reqFormat = format; 4427 uint32_t reqChannels = channels; 4428 4429 if (pDevices == NULL || *pDevices == 0) { 4430 return 0; 4431 } 4432 Mutex::Autolock _l(mLock); 4433 4434 AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, 4435 (int *)&format, 4436 &channels, 4437 &samplingRate, 4438 &status, 4439 (AudioSystem::audio_in_acoustics)acoustics); 4440 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4441 input, 4442 samplingRate, 4443 format, 4444 channels, 4445 acoustics, 4446 status); 4447 4448 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4449 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4450 // or stereo to mono conversions on 16 bit PCM inputs. 4451 if (input == 0 && status == BAD_VALUE && 4452 reqFormat == format && format == AudioSystem::PCM_16_BIT && 4453 (samplingRate <= 2 * reqSamplingRate) && 4454 (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) { 4455 LOGV("openInput() reopening with proposed sampling rate and channels"); 4456 input = mAudioHardware->openInputStream(*pDevices, 4457 (int *)&format, 4458 &channels, 4459 &samplingRate, 4460 &status, 4461 (AudioSystem::audio_in_acoustics)acoustics); 4462 } 4463 4464 if (input != 0) { 4465 int id = nextUniqueId(); 4466 // Start record thread 4467 thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); 4468 mRecordThreads.add(id, thread); 4469 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4470 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4471 if (pFormat) *pFormat = format; 4472 if (pChannels) *pChannels = reqChannels; 4473 4474 input->standby(); 4475 4476 // notify client processes of the new input creation 4477 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4478 return id; 4479 } 4480 4481 return 0; 4482} 4483 4484status_t AudioFlinger::closeInput(int input) 4485{ 4486 // keep strong reference on the record thread so that 4487 // it is not destroyed while exit() is executed 4488 sp <RecordThread> thread; 4489 { 4490 Mutex::Autolock _l(mLock); 4491 thread = checkRecordThread_l(input); 4492 if (thread == NULL) { 4493 return BAD_VALUE; 4494 } 4495 4496 LOGV("closeInput() %d", input); 4497 void *param2 = 0; 4498 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4499 mRecordThreads.removeItem(input); 4500 } 4501 thread->exit(); 4502 4503 mAudioHardware->closeInputStream(thread->getInput()); 4504 4505 return NO_ERROR; 4506} 4507 4508status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4509{ 4510 Mutex::Autolock _l(mLock); 4511 MixerThread *dstThread = checkMixerThread_l(output); 4512 if (dstThread == NULL) { 4513 LOGW("setStreamOutput() bad output id %d", output); 4514 return BAD_VALUE; 4515 } 4516 4517 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4518 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4519 4520 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4521 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4522 if (thread != dstThread && 4523 thread->type() != PlaybackThread::DIRECT) { 4524 MixerThread *srcThread = (MixerThread *)thread; 4525 srcThread->invalidateTracks(stream); 4526 } 4527 } 4528 4529 return NO_ERROR; 4530} 4531 4532 4533int AudioFlinger::newAudioSessionId() 4534{ 4535 return nextUniqueId(); 4536} 4537 4538// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4539AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4540{ 4541 PlaybackThread *thread = NULL; 4542 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4543 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4544 } 4545 return thread; 4546} 4547 4548// checkMixerThread_l() must be called with AudioFlinger::mLock held 4549AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4550{ 4551 PlaybackThread *thread = checkPlaybackThread_l(output); 4552 if (thread != NULL) { 4553 if (thread->type() == PlaybackThread::DIRECT) { 4554 thread = NULL; 4555 } 4556 } 4557 return (MixerThread *)thread; 4558} 4559 4560// checkRecordThread_l() must be called with AudioFlinger::mLock held 4561AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4562{ 4563 RecordThread *thread = NULL; 4564 if (mRecordThreads.indexOfKey(input) >= 0) { 4565 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4566 } 4567 return thread; 4568} 4569 4570int AudioFlinger::nextUniqueId() 4571{ 4572 return android_atomic_inc(&mNextUniqueId); 4573} 4574 4575// ---------------------------------------------------------------------------- 4576// Effect management 4577// ---------------------------------------------------------------------------- 4578 4579 4580status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) 4581{ 4582 // check calling permissions 4583 if (!settingsAllowed()) { 4584 return PERMISSION_DENIED; 4585 } 4586 // only allow libraries loaded from /system/lib/soundfx for now 4587 if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) { 4588 return PERMISSION_DENIED; 4589 } 4590 4591 Mutex::Autolock _l(mLock); 4592 return EffectLoadLibrary(libPath, handle); 4593} 4594 4595status_t AudioFlinger::unloadEffectLibrary(int handle) 4596{ 4597 // check calling permissions 4598 if (!settingsAllowed()) { 4599 return PERMISSION_DENIED; 4600 } 4601 4602 Mutex::Autolock _l(mLock); 4603 return EffectUnloadLibrary(handle); 4604} 4605 4606status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4607{ 4608 Mutex::Autolock _l(mLock); 4609 return EffectQueryNumberEffects(numEffects); 4610} 4611 4612status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4613{ 4614 Mutex::Autolock _l(mLock); 4615 return EffectQueryEffect(index, descriptor); 4616} 4617 4618status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4619{ 4620 Mutex::Autolock _l(mLock); 4621 return EffectGetDescriptor(pUuid, descriptor); 4622} 4623 4624 4625// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4626static const effect_uuid_t VISUALIZATION_UUID_ = 4627 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4628 4629sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4630 effect_descriptor_t *pDesc, 4631 const sp<IEffectClient>& effectClient, 4632 int32_t priority, 4633 int output, 4634 int sessionId, 4635 status_t *status, 4636 int *id, 4637 int *enabled) 4638{ 4639 status_t lStatus = NO_ERROR; 4640 sp<EffectHandle> handle; 4641 effect_interface_t itfe; 4642 effect_descriptor_t desc; 4643 sp<Client> client; 4644 wp<Client> wclient; 4645 4646 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", 4647 pid, effectClient.get(), priority, sessionId, output); 4648 4649 if (pDesc == NULL) { 4650 lStatus = BAD_VALUE; 4651 goto Exit; 4652 } 4653 4654 { 4655 Mutex::Autolock _l(mLock); 4656 4657 // check recording permission for visualizer 4658 if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4659 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) { 4660 if (!recordingAllowed()) { 4661 lStatus = PERMISSION_DENIED; 4662 goto Exit; 4663 } 4664 } 4665 4666 if (!EffectIsNullUuid(&pDesc->uuid)) { 4667 // if uuid is specified, request effect descriptor 4668 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4669 if (lStatus < 0) { 4670 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4671 goto Exit; 4672 } 4673 } else { 4674 // if uuid is not specified, look for an available implementation 4675 // of the required type in effect factory 4676 if (EffectIsNullUuid(&pDesc->type)) { 4677 LOGW("createEffect() no effect type"); 4678 lStatus = BAD_VALUE; 4679 goto Exit; 4680 } 4681 uint32_t numEffects = 0; 4682 effect_descriptor_t d; 4683 bool found = false; 4684 4685 lStatus = EffectQueryNumberEffects(&numEffects); 4686 if (lStatus < 0) { 4687 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4688 goto Exit; 4689 } 4690 for (uint32_t i = 0; i < numEffects; i++) { 4691 lStatus = EffectQueryEffect(i, &desc); 4692 if (lStatus < 0) { 4693 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4694 continue; 4695 } 4696 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4697 // If matching type found save effect descriptor. If the session is 4698 // 0 and the effect is not auxiliary, continue enumeration in case 4699 // an auxiliary version of this effect type is available 4700 found = true; 4701 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4702 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX || 4703 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4704 break; 4705 } 4706 } 4707 } 4708 if (!found) { 4709 lStatus = BAD_VALUE; 4710 LOGW("createEffect() effect not found"); 4711 goto Exit; 4712 } 4713 // For same effect type, chose auxiliary version over insert version if 4714 // connect to output mix (Compliance to OpenSL ES) 4715 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && 4716 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4717 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4718 } 4719 } 4720 4721 // Do not allow auxiliary effects on a session different from 0 (output mix) 4722 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX && 4723 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4724 lStatus = INVALID_OPERATION; 4725 goto Exit; 4726 } 4727 4728 // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects 4729 // that can only be created by audio policy manager (running in same process) 4730 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && 4731 getpid() != IPCThreadState::self()->getCallingPid()) { 4732 lStatus = INVALID_OPERATION; 4733 goto Exit; 4734 } 4735 4736 // return effect descriptor 4737 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4738 4739 // If output is not specified try to find a matching audio session ID in one of the 4740 // output threads. 4741 // TODO: allow attachment of effect to inputs 4742 if (output == 0) { 4743 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) { 4744 // output must be specified by AudioPolicyManager when using session 4745 // AudioSystem::SESSION_OUTPUT_STAGE 4746 lStatus = BAD_VALUE; 4747 goto Exit; 4748 } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 4749 output = AudioSystem::getOutputForEffect(&desc); 4750 LOGV("createEffect() got output %d for effect %s", output, desc.name); 4751 } else { 4752 // look for the thread where the specified audio session is present 4753 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4754 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4755 output = mPlaybackThreads.keyAt(i); 4756 break; 4757 } 4758 } 4759 // If no output thread contains the requested session ID, default to 4760 // first output. The effect chain will be moved to the correct output 4761 // thread when a track with the same session ID is created 4762 if (output == 0 && mPlaybackThreads.size()) { 4763 output = mPlaybackThreads.keyAt(0); 4764 } 4765 } 4766 } 4767 PlaybackThread *thread = checkPlaybackThread_l(output); 4768 if (thread == NULL) { 4769 LOGE("createEffect() unknown output thread"); 4770 lStatus = BAD_VALUE; 4771 goto Exit; 4772 } 4773 4774 wclient = mClients.valueFor(pid); 4775 4776 if (wclient != NULL) { 4777 client = wclient.promote(); 4778 } else { 4779 client = new Client(this, pid); 4780 mClients.add(pid, client); 4781 } 4782 4783 // create effect on selected output trhead 4784 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4785 &desc, enabled, &lStatus); 4786 if (handle != 0 && id != NULL) { 4787 *id = handle->id(); 4788 } 4789 } 4790 4791Exit: 4792 if(status) { 4793 *status = lStatus; 4794 } 4795 return handle; 4796} 4797 4798status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4799{ 4800 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4801 session, srcOutput, dstOutput); 4802 Mutex::Autolock _l(mLock); 4803 if (srcOutput == dstOutput) { 4804 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4805 return NO_ERROR; 4806 } 4807 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4808 if (srcThread == NULL) { 4809 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4810 return BAD_VALUE; 4811 } 4812 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 4813 if (dstThread == NULL) { 4814 LOGW("moveEffects() bad dstOutput %d", dstOutput); 4815 return BAD_VALUE; 4816 } 4817 4818 Mutex::Autolock _dl(dstThread->mLock); 4819 Mutex::Autolock _sl(srcThread->mLock); 4820 moveEffectChain_l(session, srcThread, dstThread, false); 4821 4822 return NO_ERROR; 4823} 4824 4825// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 4826status_t AudioFlinger::moveEffectChain_l(int session, 4827 AudioFlinger::PlaybackThread *srcThread, 4828 AudioFlinger::PlaybackThread *dstThread, 4829 bool reRegister) 4830{ 4831 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 4832 session, srcThread, dstThread); 4833 4834 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 4835 if (chain == 0) { 4836 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 4837 session, srcThread); 4838 return INVALID_OPERATION; 4839 } 4840 4841 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 4842 // so that a new chain is created with correct parameters when first effect is added. This is 4843 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 4844 // removed. 4845 srcThread->removeEffectChain_l(chain); 4846 4847 // transfer all effects one by one so that new effect chain is created on new thread with 4848 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 4849 int dstOutput = dstThread->id(); 4850 sp<EffectChain> dstChain; 4851 uint32_t strategy; 4852 sp<EffectModule> effect = chain->getEffectFromId_l(0); 4853 while (effect != 0) { 4854 srcThread->removeEffect_l(effect); 4855 dstThread->addEffect_l(effect); 4856 // if the move request is not received from audio policy manager, the effect must be 4857 // re-registered with the new strategy and output 4858 if (dstChain == 0) { 4859 dstChain = effect->chain().promote(); 4860 if (dstChain == 0) { 4861 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 4862 srcThread->addEffect_l(effect); 4863 return NO_INIT; 4864 } 4865 strategy = dstChain->strategy(); 4866 } 4867 if (reRegister) { 4868 AudioSystem::unregisterEffect(effect->id()); 4869 AudioSystem::registerEffect(&effect->desc(), 4870 dstOutput, 4871 strategy, 4872 session, 4873 effect->id()); 4874 } 4875 effect = chain->getEffectFromId_l(0); 4876 } 4877 4878 return NO_ERROR; 4879} 4880 4881// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 4882sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( 4883 const sp<AudioFlinger::Client>& client, 4884 const sp<IEffectClient>& effectClient, 4885 int32_t priority, 4886 int sessionId, 4887 effect_descriptor_t *desc, 4888 int *enabled, 4889 status_t *status 4890 ) 4891{ 4892 sp<EffectModule> effect; 4893 sp<EffectHandle> handle; 4894 status_t lStatus; 4895 sp<Track> track; 4896 sp<EffectChain> chain; 4897 bool chainCreated = false; 4898 bool effectCreated = false; 4899 bool effectRegistered = false; 4900 4901 if (mOutput == 0) { 4902 LOGW("createEffect_l() Audio driver not initialized."); 4903 lStatus = NO_INIT; 4904 goto Exit; 4905 } 4906 4907 // Do not allow auxiliary effect on session other than 0 4908 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && 4909 sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 4910 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4911 desc->name, sessionId); 4912 lStatus = BAD_VALUE; 4913 goto Exit; 4914 } 4915 4916 // Do not allow effects with session ID 0 on direct output or duplicating threads 4917 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 4918 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) { 4919 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4920 desc->name, sessionId); 4921 lStatus = BAD_VALUE; 4922 goto Exit; 4923 } 4924 4925 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 4926 4927 { // scope for mLock 4928 Mutex::Autolock _l(mLock); 4929 4930 // check for existing effect chain with the requested audio session 4931 chain = getEffectChain_l(sessionId); 4932 if (chain == 0) { 4933 // create a new chain for this session 4934 LOGV("createEffect_l() new effect chain for session %d", sessionId); 4935 chain = new EffectChain(this, sessionId); 4936 addEffectChain_l(chain); 4937 chain->setStrategy(getStrategyForSession_l(sessionId)); 4938 chainCreated = true; 4939 } else { 4940 effect = chain->getEffectFromDesc_l(desc); 4941 } 4942 4943 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 4944 4945 if (effect == 0) { 4946 int id = mAudioFlinger->nextUniqueId(); 4947 // Check CPU and memory usage 4948 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 4949 if (lStatus != NO_ERROR) { 4950 goto Exit; 4951 } 4952 effectRegistered = true; 4953 // create a new effect module if none present in the chain 4954 effect = new EffectModule(this, chain, desc, id, sessionId); 4955 lStatus = effect->status(); 4956 if (lStatus != NO_ERROR) { 4957 goto Exit; 4958 } 4959 lStatus = chain->addEffect_l(effect); 4960 if (lStatus != NO_ERROR) { 4961 goto Exit; 4962 } 4963 effectCreated = true; 4964 4965 effect->setDevice(mDevice); 4966 effect->setMode(mAudioFlinger->getMode()); 4967 } 4968 // create effect handle and connect it to effect module 4969 handle = new EffectHandle(effect, client, effectClient, priority); 4970 lStatus = effect->addHandle(handle); 4971 if (enabled) { 4972 *enabled = (int)effect->isEnabled(); 4973 } 4974 } 4975 4976Exit: 4977 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 4978 Mutex::Autolock _l(mLock); 4979 if (effectCreated) { 4980 chain->removeEffect_l(effect); 4981 } 4982 if (effectRegistered) { 4983 AudioSystem::unregisterEffect(effect->id()); 4984 } 4985 if (chainCreated) { 4986 removeEffectChain_l(chain); 4987 } 4988 handle.clear(); 4989 } 4990 4991 if(status) { 4992 *status = lStatus; 4993 } 4994 return handle; 4995} 4996 4997// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 4998// PlaybackThread::mLock held 4999status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect) 5000{ 5001 // check for existing effect chain with the requested audio session 5002 int sessionId = effect->sessionId(); 5003 sp<EffectChain> chain = getEffectChain_l(sessionId); 5004 bool chainCreated = false; 5005 5006 if (chain == 0) { 5007 // create a new chain for this session 5008 LOGV("addEffect_l() new effect chain for session %d", sessionId); 5009 chain = new EffectChain(this, sessionId); 5010 addEffectChain_l(chain); 5011 chain->setStrategy(getStrategyForSession_l(sessionId)); 5012 chainCreated = true; 5013 } 5014 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5015 5016 if (chain->getEffectFromId_l(effect->id()) != 0) { 5017 LOGW("addEffect_l() %p effect %s already present in chain %p", 5018 this, effect->desc().name, chain.get()); 5019 return BAD_VALUE; 5020 } 5021 5022 status_t status = chain->addEffect_l(effect); 5023 if (status != NO_ERROR) { 5024 if (chainCreated) { 5025 removeEffectChain_l(chain); 5026 } 5027 return status; 5028 } 5029 5030 effect->setDevice(mDevice); 5031 effect->setMode(mAudioFlinger->getMode()); 5032 return NO_ERROR; 5033} 5034 5035void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) { 5036 5037 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5038 effect_descriptor_t desc = effect->desc(); 5039 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5040 detachAuxEffect_l(effect->id()); 5041 } 5042 5043 sp<EffectChain> chain = effect->chain().promote(); 5044 if (chain != 0) { 5045 // remove effect chain if removing last effect 5046 if (chain->removeEffect_l(effect) == 0) { 5047 removeEffectChain_l(chain); 5048 } 5049 } else { 5050 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5051 } 5052} 5053 5054void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect, 5055 const wp<EffectHandle>& handle) { 5056 Mutex::Autolock _l(mLock); 5057 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5058 // delete the effect module if removing last handle on it 5059 if (effect->removeHandle(handle) == 0) { 5060 removeEffect_l(effect); 5061 AudioSystem::unregisterEffect(effect->id()); 5062 } 5063} 5064 5065status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5066{ 5067 int session = chain->sessionId(); 5068 int16_t *buffer = mMixBuffer; 5069 bool ownsBuffer = false; 5070 5071 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5072 if (session > 0) { 5073 // Only one effect chain can be present in direct output thread and it uses 5074 // the mix buffer as input 5075 if (mType != DIRECT) { 5076 size_t numSamples = mFrameCount * mChannelCount; 5077 buffer = new int16_t[numSamples]; 5078 memset(buffer, 0, numSamples * sizeof(int16_t)); 5079 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5080 ownsBuffer = true; 5081 } 5082 5083 // Attach all tracks with same session ID to this chain. 5084 for (size_t i = 0; i < mTracks.size(); ++i) { 5085 sp<Track> track = mTracks[i]; 5086 if (session == track->sessionId()) { 5087 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5088 track->setMainBuffer(buffer); 5089 } 5090 } 5091 5092 // indicate all active tracks in the chain 5093 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5094 sp<Track> track = mActiveTracks[i].promote(); 5095 if (track == 0) continue; 5096 if (session == track->sessionId()) { 5097 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5098 chain->startTrack(); 5099 } 5100 } 5101 } 5102 5103 chain->setInBuffer(buffer, ownsBuffer); 5104 chain->setOutBuffer(mMixBuffer); 5105 // Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect 5106 // chains list in order to be processed last as it contains output stage effects 5107 // Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before 5108 // session AudioSystem::SESSION_OUTPUT_STAGE to be processed 5109 // after track specific effects and before output stage 5110 // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and 5111 // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX 5112 // Effect chain for other sessions are inserted at beginning of effect 5113 // chains list to be processed before output mix effects. Relative order between other 5114 // sessions is not important 5115 size_t size = mEffectChains.size(); 5116 size_t i = 0; 5117 for (i = 0; i < size; i++) { 5118 if (mEffectChains[i]->sessionId() < session) break; 5119 } 5120 mEffectChains.insertAt(chain, i); 5121 5122 return NO_ERROR; 5123} 5124 5125size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5126{ 5127 int session = chain->sessionId(); 5128 5129 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5130 5131 for (size_t i = 0; i < mEffectChains.size(); i++) { 5132 if (chain == mEffectChains[i]) { 5133 mEffectChains.removeAt(i); 5134 // detach all tracks with same session ID from this chain 5135 for (size_t i = 0; i < mTracks.size(); ++i) { 5136 sp<Track> track = mTracks[i]; 5137 if (session == track->sessionId()) { 5138 track->setMainBuffer(mMixBuffer); 5139 } 5140 } 5141 break; 5142 } 5143 } 5144 return mEffectChains.size(); 5145} 5146 5147void AudioFlinger::PlaybackThread::lockEffectChains_l( 5148 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5149{ 5150 effectChains = mEffectChains; 5151 for (size_t i = 0; i < mEffectChains.size(); i++) { 5152 mEffectChains[i]->lock(); 5153 } 5154} 5155 5156void AudioFlinger::PlaybackThread::unlockEffectChains( 5157 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5158{ 5159 for (size_t i = 0; i < effectChains.size(); i++) { 5160 effectChains[i]->unlock(); 5161 } 5162} 5163 5164 5165sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) 5166{ 5167 sp<EffectModule> effect; 5168 5169 sp<EffectChain> chain = getEffectChain_l(sessionId); 5170 if (chain != 0) { 5171 effect = chain->getEffectFromId_l(effectId); 5172 } 5173 return effect; 5174} 5175 5176status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5177 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5178{ 5179 Mutex::Autolock _l(mLock); 5180 return attachAuxEffect_l(track, EffectId); 5181} 5182 5183status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5184 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5185{ 5186 status_t status = NO_ERROR; 5187 5188 if (EffectId == 0) { 5189 track->setAuxBuffer(0, NULL); 5190 } else { 5191 // Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX 5192 sp<EffectModule> effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId); 5193 if (effect != 0) { 5194 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5195 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5196 } else { 5197 status = INVALID_OPERATION; 5198 } 5199 } else { 5200 status = BAD_VALUE; 5201 } 5202 } 5203 return status; 5204} 5205 5206void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5207{ 5208 for (size_t i = 0; i < mTracks.size(); ++i) { 5209 sp<Track> track = mTracks[i]; 5210 if (track->auxEffectId() == effectId) { 5211 attachAuxEffect_l(track, 0); 5212 } 5213 } 5214} 5215 5216// ---------------------------------------------------------------------------- 5217// EffectModule implementation 5218// ---------------------------------------------------------------------------- 5219 5220#undef LOG_TAG 5221#define LOG_TAG "AudioFlinger::EffectModule" 5222 5223AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5224 const wp<AudioFlinger::EffectChain>& chain, 5225 effect_descriptor_t *desc, 5226 int id, 5227 int sessionId) 5228 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5229 mStatus(NO_INIT), mState(IDLE) 5230{ 5231 LOGV("Constructor %p", this); 5232 int lStatus; 5233 sp<ThreadBase> thread = mThread.promote(); 5234 if (thread == 0) { 5235 return; 5236 } 5237 PlaybackThread *p = (PlaybackThread *)thread.get(); 5238 5239 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5240 5241 // create effect engine from effect factory 5242 mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); 5243 5244 if (mStatus != NO_ERROR) { 5245 return; 5246 } 5247 lStatus = init(); 5248 if (lStatus < 0) { 5249 mStatus = lStatus; 5250 goto Error; 5251 } 5252 5253 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5254 return; 5255Error: 5256 EffectRelease(mEffectInterface); 5257 mEffectInterface = NULL; 5258 LOGV("Constructor Error %d", mStatus); 5259} 5260 5261AudioFlinger::EffectModule::~EffectModule() 5262{ 5263 LOGV("Destructor %p", this); 5264 if (mEffectInterface != NULL) { 5265 // release effect engine 5266 EffectRelease(mEffectInterface); 5267 } 5268} 5269 5270status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5271{ 5272 status_t status; 5273 5274 Mutex::Autolock _l(mLock); 5275 // First handle in mHandles has highest priority and controls the effect module 5276 int priority = handle->priority(); 5277 size_t size = mHandles.size(); 5278 sp<EffectHandle> h; 5279 size_t i; 5280 for (i = 0; i < size; i++) { 5281 h = mHandles[i].promote(); 5282 if (h == 0) continue; 5283 if (h->priority() <= priority) break; 5284 } 5285 // if inserted in first place, move effect control from previous owner to this handle 5286 if (i == 0) { 5287 if (h != 0) { 5288 h->setControl(false, true); 5289 } 5290 handle->setControl(true, false); 5291 status = NO_ERROR; 5292 } else { 5293 status = ALREADY_EXISTS; 5294 } 5295 mHandles.insertAt(handle, i); 5296 return status; 5297} 5298 5299size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5300{ 5301 Mutex::Autolock _l(mLock); 5302 size_t size = mHandles.size(); 5303 size_t i; 5304 for (i = 0; i < size; i++) { 5305 if (mHandles[i] == handle) break; 5306 } 5307 if (i == size) { 5308 return size; 5309 } 5310 mHandles.removeAt(i); 5311 size = mHandles.size(); 5312 // if removed from first place, move effect control from this handle to next in line 5313 if (i == 0 && size != 0) { 5314 sp<EffectHandle> h = mHandles[0].promote(); 5315 if (h != 0) { 5316 h->setControl(true, true); 5317 } 5318 } 5319 5320 return size; 5321} 5322 5323void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5324{ 5325 // keep a strong reference on this EffectModule to avoid calling the 5326 // destructor before we exit 5327 sp<EffectModule> keep(this); 5328 { 5329 sp<ThreadBase> thread = mThread.promote(); 5330 if (thread != 0) { 5331 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5332 playbackThread->disconnectEffect(keep, handle); 5333 } 5334 } 5335} 5336 5337void AudioFlinger::EffectModule::updateState() { 5338 Mutex::Autolock _l(mLock); 5339 5340 switch (mState) { 5341 case RESTART: 5342 reset_l(); 5343 // FALL THROUGH 5344 5345 case STARTING: 5346 // clear auxiliary effect input buffer for next accumulation 5347 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5348 memset(mConfig.inputCfg.buffer.raw, 5349 0, 5350 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5351 } 5352 start_l(); 5353 mState = ACTIVE; 5354 break; 5355 case STOPPING: 5356 stop_l(); 5357 mDisableWaitCnt = mMaxDisableWaitCnt; 5358 mState = STOPPED; 5359 break; 5360 case STOPPED: 5361 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5362 // turn off sequence. 5363 if (--mDisableWaitCnt == 0) { 5364 reset_l(); 5365 mState = IDLE; 5366 } 5367 break; 5368 default: //IDLE , ACTIVE 5369 break; 5370 } 5371} 5372 5373void AudioFlinger::EffectModule::process() 5374{ 5375 Mutex::Autolock _l(mLock); 5376 5377 if (mEffectInterface == NULL || 5378 mConfig.inputCfg.buffer.raw == NULL || 5379 mConfig.outputCfg.buffer.raw == NULL) { 5380 return; 5381 } 5382 5383 if (isProcessEnabled()) { 5384 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5385 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5386 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5387 mConfig.inputCfg.buffer.s32, 5388 mConfig.inputCfg.buffer.frameCount/2); 5389 } 5390 5391 // do the actual processing in the effect engine 5392 int ret = (*mEffectInterface)->process(mEffectInterface, 5393 &mConfig.inputCfg.buffer, 5394 &mConfig.outputCfg.buffer); 5395 5396 // force transition to IDLE state when engine is ready 5397 if (mState == STOPPED && ret == -ENODATA) { 5398 mDisableWaitCnt = 1; 5399 } 5400 5401 // clear auxiliary effect input buffer for next accumulation 5402 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5403 memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5404 } 5405 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5406 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){ 5407 // If an insert effect is idle and input buffer is different from output buffer, copy input to 5408 // output 5409 sp<EffectChain> chain = mChain.promote(); 5410 if (chain != 0 && chain->activeTracks() != 0) { 5411 size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t); 5412 if (mConfig.inputCfg.channels == CHANNEL_STEREO) { 5413 size *= 2; 5414 } 5415 memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size); 5416 } 5417 } 5418} 5419 5420void AudioFlinger::EffectModule::reset_l() 5421{ 5422 if (mEffectInterface == NULL) { 5423 return; 5424 } 5425 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5426} 5427 5428status_t AudioFlinger::EffectModule::configure() 5429{ 5430 uint32_t channels; 5431 if (mEffectInterface == NULL) { 5432 return NO_INIT; 5433 } 5434 5435 sp<ThreadBase> thread = mThread.promote(); 5436 if (thread == 0) { 5437 return DEAD_OBJECT; 5438 } 5439 5440 // TODO: handle configuration of effects replacing track process 5441 if (thread->channelCount() == 1) { 5442 channels = CHANNEL_MONO; 5443 } else { 5444 channels = CHANNEL_STEREO; 5445 } 5446 5447 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5448 mConfig.inputCfg.channels = CHANNEL_MONO; 5449 } else { 5450 mConfig.inputCfg.channels = channels; 5451 } 5452 mConfig.outputCfg.channels = channels; 5453 mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; 5454 mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; 5455 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5456 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5457 mConfig.inputCfg.bufferProvider.cookie = NULL; 5458 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5459 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5460 mConfig.outputCfg.bufferProvider.cookie = NULL; 5461 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5462 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5463 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5464 // Insert effect: 5465 // - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE, 5466 // always overwrites output buffer: input buffer == output buffer 5467 // - in other sessions: 5468 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5469 // other effect: overwrites output buffer: input buffer == output buffer 5470 // Auxiliary effect: 5471 // accumulates in output buffer: input buffer != output buffer 5472 // Therefore: accumulate <=> input buffer != output buffer 5473 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5474 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5475 } else { 5476 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5477 } 5478 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5479 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5480 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5481 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5482 5483 LOGV("configure() %p thread %p buffer %p framecount %d", 5484 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5485 5486 status_t cmdStatus; 5487 uint32_t size = sizeof(int); 5488 status_t status = (*mEffectInterface)->command(mEffectInterface, 5489 EFFECT_CMD_CONFIGURE, 5490 sizeof(effect_config_t), 5491 &mConfig, 5492 &size, 5493 &cmdStatus); 5494 if (status == 0) { 5495 status = cmdStatus; 5496 } 5497 5498 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5499 (1000 * mConfig.outputCfg.buffer.frameCount); 5500 5501 return status; 5502} 5503 5504status_t AudioFlinger::EffectModule::init() 5505{ 5506 Mutex::Autolock _l(mLock); 5507 if (mEffectInterface == NULL) { 5508 return NO_INIT; 5509 } 5510 status_t cmdStatus; 5511 uint32_t size = sizeof(status_t); 5512 status_t status = (*mEffectInterface)->command(mEffectInterface, 5513 EFFECT_CMD_INIT, 5514 0, 5515 NULL, 5516 &size, 5517 &cmdStatus); 5518 if (status == 0) { 5519 status = cmdStatus; 5520 } 5521 return status; 5522} 5523 5524status_t AudioFlinger::EffectModule::start_l() 5525{ 5526 if (mEffectInterface == NULL) { 5527 return NO_INIT; 5528 } 5529 status_t cmdStatus; 5530 uint32_t size = sizeof(status_t); 5531 status_t status = (*mEffectInterface)->command(mEffectInterface, 5532 EFFECT_CMD_ENABLE, 5533 0, 5534 NULL, 5535 &size, 5536 &cmdStatus); 5537 if (status == 0) { 5538 status = cmdStatus; 5539 } 5540 return status; 5541} 5542 5543status_t AudioFlinger::EffectModule::stop_l() 5544{ 5545 if (mEffectInterface == NULL) { 5546 return NO_INIT; 5547 } 5548 status_t cmdStatus; 5549 uint32_t size = sizeof(status_t); 5550 status_t status = (*mEffectInterface)->command(mEffectInterface, 5551 EFFECT_CMD_DISABLE, 5552 0, 5553 NULL, 5554 &size, 5555 &cmdStatus); 5556 if (status == 0) { 5557 status = cmdStatus; 5558 } 5559 return status; 5560} 5561 5562status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5563 uint32_t cmdSize, 5564 void *pCmdData, 5565 uint32_t *replySize, 5566 void *pReplyData) 5567{ 5568 Mutex::Autolock _l(mLock); 5569// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5570 5571 if (mEffectInterface == NULL) { 5572 return NO_INIT; 5573 } 5574 status_t status = (*mEffectInterface)->command(mEffectInterface, 5575 cmdCode, 5576 cmdSize, 5577 pCmdData, 5578 replySize, 5579 pReplyData); 5580 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5581 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5582 for (size_t i = 1; i < mHandles.size(); i++) { 5583 sp<EffectHandle> h = mHandles[i].promote(); 5584 if (h != 0) { 5585 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5586 } 5587 } 5588 } 5589 return status; 5590} 5591 5592status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5593{ 5594 Mutex::Autolock _l(mLock); 5595 LOGV("setEnabled %p enabled %d", this, enabled); 5596 5597 if (enabled != isEnabled()) { 5598 switch (mState) { 5599 // going from disabled to enabled 5600 case IDLE: 5601 mState = STARTING; 5602 break; 5603 case STOPPED: 5604 mState = RESTART; 5605 break; 5606 case STOPPING: 5607 mState = ACTIVE; 5608 break; 5609 5610 // going from enabled to disabled 5611 case RESTART: 5612 mState = STOPPED; 5613 break; 5614 case STARTING: 5615 mState = IDLE; 5616 break; 5617 case ACTIVE: 5618 mState = STOPPING; 5619 break; 5620 } 5621 for (size_t i = 1; i < mHandles.size(); i++) { 5622 sp<EffectHandle> h = mHandles[i].promote(); 5623 if (h != 0) { 5624 h->setEnabled(enabled); 5625 } 5626 } 5627 } 5628 return NO_ERROR; 5629} 5630 5631bool AudioFlinger::EffectModule::isEnabled() 5632{ 5633 switch (mState) { 5634 case RESTART: 5635 case STARTING: 5636 case ACTIVE: 5637 return true; 5638 case IDLE: 5639 case STOPPING: 5640 case STOPPED: 5641 default: 5642 return false; 5643 } 5644} 5645 5646bool AudioFlinger::EffectModule::isProcessEnabled() 5647{ 5648 switch (mState) { 5649 case RESTART: 5650 case ACTIVE: 5651 case STOPPING: 5652 case STOPPED: 5653 return true; 5654 case IDLE: 5655 case STARTING: 5656 default: 5657 return false; 5658 } 5659} 5660 5661status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5662{ 5663 Mutex::Autolock _l(mLock); 5664 status_t status = NO_ERROR; 5665 5666 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5667 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5668 if (isProcessEnabled() && 5669 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5670 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5671 status_t cmdStatus; 5672 uint32_t volume[2]; 5673 uint32_t *pVolume = NULL; 5674 uint32_t size = sizeof(volume); 5675 volume[0] = *left; 5676 volume[1] = *right; 5677 if (controller) { 5678 pVolume = volume; 5679 } 5680 status = (*mEffectInterface)->command(mEffectInterface, 5681 EFFECT_CMD_SET_VOLUME, 5682 size, 5683 volume, 5684 &size, 5685 pVolume); 5686 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5687 *left = volume[0]; 5688 *right = volume[1]; 5689 } 5690 } 5691 return status; 5692} 5693 5694status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5695{ 5696 Mutex::Autolock _l(mLock); 5697 status_t status = NO_ERROR; 5698 if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5699 // convert device bit field from AudioSystem to EffectApi format. 5700 device = deviceAudioSystemToEffectApi(device); 5701 if (device == 0) { 5702 return BAD_VALUE; 5703 } 5704 status_t cmdStatus; 5705 uint32_t size = sizeof(status_t); 5706 status = (*mEffectInterface)->command(mEffectInterface, 5707 EFFECT_CMD_SET_DEVICE, 5708 sizeof(uint32_t), 5709 &device, 5710 &size, 5711 &cmdStatus); 5712 if (status == NO_ERROR) { 5713 status = cmdStatus; 5714 } 5715 } 5716 return status; 5717} 5718 5719status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 5720{ 5721 Mutex::Autolock _l(mLock); 5722 status_t status = NO_ERROR; 5723 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 5724 // convert audio mode from AudioSystem to EffectApi format. 5725 int effectMode = modeAudioSystemToEffectApi(mode); 5726 if (effectMode < 0) { 5727 return BAD_VALUE; 5728 } 5729 status_t cmdStatus; 5730 uint32_t size = sizeof(status_t); 5731 status = (*mEffectInterface)->command(mEffectInterface, 5732 EFFECT_CMD_SET_AUDIO_MODE, 5733 sizeof(int), 5734 &effectMode, 5735 &size, 5736 &cmdStatus); 5737 if (status == NO_ERROR) { 5738 status = cmdStatus; 5739 } 5740 } 5741 return status; 5742} 5743 5744// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified 5745const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { 5746 DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE 5747 DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER 5748 DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET 5749 DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE 5750 DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO 5751 DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET 5752 DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT 5753 DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP 5754 DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES 5755 DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER 5756 DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL 5757}; 5758 5759uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) 5760{ 5761 uint32_t deviceOut = 0; 5762 while (device) { 5763 const uint32_t i = 31 - __builtin_clz(device); 5764 device &= ~(1 << i); 5765 if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { 5766 LOGE("device convertion error for AudioSystem device 0x%08x", device); 5767 return 0; 5768 } 5769 deviceOut |= (uint32_t)sDeviceConvTable[i]; 5770 } 5771 return deviceOut; 5772} 5773 5774// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified 5775const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { 5776 AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL 5777 AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE 5778 AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL 5779}; 5780 5781int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) 5782{ 5783 int modeOut = -1; 5784 if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { 5785 modeOut = (int)sModeConvTable[mode]; 5786 } 5787 return modeOut; 5788} 5789 5790status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 5791{ 5792 const size_t SIZE = 256; 5793 char buffer[SIZE]; 5794 String8 result; 5795 5796 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 5797 result.append(buffer); 5798 5799 bool locked = tryLock(mLock); 5800 // failed to lock - AudioFlinger is probably deadlocked 5801 if (!locked) { 5802 result.append("\t\tCould not lock Fx mutex:\n"); 5803 } 5804 5805 result.append("\t\tSession Status State Engine:\n"); 5806 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 5807 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 5808 result.append(buffer); 5809 5810 result.append("\t\tDescriptor:\n"); 5811 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5812 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 5813 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 5814 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 5815 result.append(buffer); 5816 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5817 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 5818 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 5819 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 5820 result.append(buffer); 5821 snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", 5822 mDescriptor.apiVersion, 5823 mDescriptor.flags); 5824 result.append(buffer); 5825 snprintf(buffer, SIZE, "\t\t- name: %s\n", 5826 mDescriptor.name); 5827 result.append(buffer); 5828 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 5829 mDescriptor.implementor); 5830 result.append(buffer); 5831 5832 result.append("\t\t- Input configuration:\n"); 5833 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5834 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5835 (uint32_t)mConfig.inputCfg.buffer.raw, 5836 mConfig.inputCfg.buffer.frameCount, 5837 mConfig.inputCfg.samplingRate, 5838 mConfig.inputCfg.channels, 5839 mConfig.inputCfg.format); 5840 result.append(buffer); 5841 5842 result.append("\t\t- Output configuration:\n"); 5843 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5844 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5845 (uint32_t)mConfig.outputCfg.buffer.raw, 5846 mConfig.outputCfg.buffer.frameCount, 5847 mConfig.outputCfg.samplingRate, 5848 mConfig.outputCfg.channels, 5849 mConfig.outputCfg.format); 5850 result.append(buffer); 5851 5852 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 5853 result.append(buffer); 5854 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 5855 for (size_t i = 0; i < mHandles.size(); ++i) { 5856 sp<EffectHandle> handle = mHandles[i].promote(); 5857 if (handle != 0) { 5858 handle->dump(buffer, SIZE); 5859 result.append(buffer); 5860 } 5861 } 5862 5863 result.append("\n"); 5864 5865 write(fd, result.string(), result.length()); 5866 5867 if (locked) { 5868 mLock.unlock(); 5869 } 5870 5871 return NO_ERROR; 5872} 5873 5874// ---------------------------------------------------------------------------- 5875// EffectHandle implementation 5876// ---------------------------------------------------------------------------- 5877 5878#undef LOG_TAG 5879#define LOG_TAG "AudioFlinger::EffectHandle" 5880 5881AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 5882 const sp<AudioFlinger::Client>& client, 5883 const sp<IEffectClient>& effectClient, 5884 int32_t priority) 5885 : BnEffect(), 5886 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 5887{ 5888 LOGV("constructor %p", this); 5889 5890 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 5891 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 5892 if (mCblkMemory != 0) { 5893 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 5894 5895 if (mCblk) { 5896 new(mCblk) effect_param_cblk_t(); 5897 mBuffer = (uint8_t *)mCblk + bufOffset; 5898 } 5899 } else { 5900 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 5901 return; 5902 } 5903} 5904 5905AudioFlinger::EffectHandle::~EffectHandle() 5906{ 5907 LOGV("Destructor %p", this); 5908 disconnect(); 5909} 5910 5911status_t AudioFlinger::EffectHandle::enable() 5912{ 5913 if (!mHasControl) return INVALID_OPERATION; 5914 if (mEffect == 0) return DEAD_OBJECT; 5915 5916 return mEffect->setEnabled(true); 5917} 5918 5919status_t AudioFlinger::EffectHandle::disable() 5920{ 5921 if (!mHasControl) return INVALID_OPERATION; 5922 if (mEffect == NULL) return DEAD_OBJECT; 5923 5924 return mEffect->setEnabled(false); 5925} 5926 5927void AudioFlinger::EffectHandle::disconnect() 5928{ 5929 if (mEffect == 0) { 5930 return; 5931 } 5932 mEffect->disconnect(this); 5933 // release sp on module => module destructor can be called now 5934 mEffect.clear(); 5935 if (mCblk) { 5936 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 5937 } 5938 mCblkMemory.clear(); // and free the shared memory 5939 if (mClient != 0) { 5940 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 5941 mClient.clear(); 5942 } 5943} 5944 5945status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 5946 uint32_t cmdSize, 5947 void *pCmdData, 5948 uint32_t *replySize, 5949 void *pReplyData) 5950{ 5951// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 5952// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 5953 5954 // only get parameter command is permitted for applications not controlling the effect 5955 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 5956 return INVALID_OPERATION; 5957 } 5958 if (mEffect == 0) return DEAD_OBJECT; 5959 5960 // handle commands that are not forwarded transparently to effect engine 5961 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 5962 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 5963 // no risk to block the whole media server process or mixer threads is we are stuck here 5964 Mutex::Autolock _l(mCblk->lock); 5965 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 5966 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 5967 mCblk->serverIndex = 0; 5968 mCblk->clientIndex = 0; 5969 return BAD_VALUE; 5970 } 5971 status_t status = NO_ERROR; 5972 while (mCblk->serverIndex < mCblk->clientIndex) { 5973 int reply; 5974 uint32_t rsize = sizeof(int); 5975 int *p = (int *)(mBuffer + mCblk->serverIndex); 5976 int size = *p++; 5977 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 5978 LOGW("command(): invalid parameter block size"); 5979 break; 5980 } 5981 effect_param_t *param = (effect_param_t *)p; 5982 if (param->psize == 0 || param->vsize == 0) { 5983 LOGW("command(): null parameter or value size"); 5984 mCblk->serverIndex += size; 5985 continue; 5986 } 5987 uint32_t psize = sizeof(effect_param_t) + 5988 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 5989 param->vsize; 5990 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 5991 psize, 5992 p, 5993 &rsize, 5994 &reply); 5995 // stop at first error encountered 5996 if (ret != NO_ERROR) { 5997 status = ret; 5998 *(int *)pReplyData = reply; 5999 break; 6000 } else if (reply != NO_ERROR) { 6001 *(int *)pReplyData = reply; 6002 break; 6003 } 6004 mCblk->serverIndex += size; 6005 } 6006 mCblk->serverIndex = 0; 6007 mCblk->clientIndex = 0; 6008 return status; 6009 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6010 *(int *)pReplyData = NO_ERROR; 6011 return enable(); 6012 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6013 *(int *)pReplyData = NO_ERROR; 6014 return disable(); 6015 } 6016 6017 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6018} 6019 6020sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 6021 return mCblkMemory; 6022} 6023 6024void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 6025{ 6026 LOGV("setControl %p control %d", this, hasControl); 6027 6028 mHasControl = hasControl; 6029 if (signal && mEffectClient != 0) { 6030 mEffectClient->controlStatusChanged(hasControl); 6031 } 6032} 6033 6034void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6035 uint32_t cmdSize, 6036 void *pCmdData, 6037 uint32_t replySize, 6038 void *pReplyData) 6039{ 6040 if (mEffectClient != 0) { 6041 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6042 } 6043} 6044 6045 6046 6047void AudioFlinger::EffectHandle::setEnabled(bool enabled) 6048{ 6049 if (mEffectClient != 0) { 6050 mEffectClient->enableStatusChanged(enabled); 6051 } 6052} 6053 6054status_t AudioFlinger::EffectHandle::onTransact( 6055 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6056{ 6057 return BnEffect::onTransact(code, data, reply, flags); 6058} 6059 6060 6061void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 6062{ 6063 bool locked = tryLock(mCblk->lock); 6064 6065 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 6066 (mClient == NULL) ? getpid() : mClient->pid(), 6067 mPriority, 6068 mHasControl, 6069 !locked, 6070 mCblk->clientIndex, 6071 mCblk->serverIndex 6072 ); 6073 6074 if (locked) { 6075 mCblk->lock.unlock(); 6076 } 6077} 6078 6079#undef LOG_TAG 6080#define LOG_TAG "AudioFlinger::EffectChain" 6081 6082AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 6083 int sessionId) 6084 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false), 6085 mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 6086 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 6087{ 6088 mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 6089} 6090 6091AudioFlinger::EffectChain::~EffectChain() 6092{ 6093 if (mOwnInBuffer) { 6094 delete mInBuffer; 6095 } 6096 6097} 6098 6099// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6100sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6101{ 6102 sp<EffectModule> effect; 6103 size_t size = mEffects.size(); 6104 6105 for (size_t i = 0; i < size; i++) { 6106 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6107 effect = mEffects[i]; 6108 break; 6109 } 6110 } 6111 return effect; 6112} 6113 6114// getEffectFromId_l() must be called with PlaybackThread::mLock held 6115sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6116{ 6117 sp<EffectModule> effect; 6118 size_t size = mEffects.size(); 6119 6120 for (size_t i = 0; i < size; i++) { 6121 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6122 if (id == 0 || mEffects[i]->id() == id) { 6123 effect = mEffects[i]; 6124 break; 6125 } 6126 } 6127 return effect; 6128} 6129 6130// Must be called with EffectChain::mLock locked 6131void AudioFlinger::EffectChain::process_l() 6132{ 6133 size_t size = mEffects.size(); 6134 for (size_t i = 0; i < size; i++) { 6135 mEffects[i]->process(); 6136 } 6137 for (size_t i = 0; i < size; i++) { 6138 mEffects[i]->updateState(); 6139 } 6140 // if no track is active, input buffer must be cleared here as the mixer process 6141 // will not do it 6142 if (mSessionId > 0 && activeTracks() == 0) { 6143 sp<ThreadBase> thread = mThread.promote(); 6144 if (thread != 0) { 6145 size_t numSamples = thread->frameCount() * thread->channelCount(); 6146 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6147 } 6148 } 6149} 6150 6151// addEffect_l() must be called with PlaybackThread::mLock held 6152status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6153{ 6154 effect_descriptor_t desc = effect->desc(); 6155 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6156 6157 Mutex::Autolock _l(mLock); 6158 effect->setChain(this); 6159 sp<ThreadBase> thread = mThread.promote(); 6160 if (thread == 0) { 6161 return NO_INIT; 6162 } 6163 effect->setThread(thread); 6164 6165 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6166 // Auxiliary effects are inserted at the beginning of mEffects vector as 6167 // they are processed first and accumulated in chain input buffer 6168 mEffects.insertAt(effect, 0); 6169 6170 // the input buffer for auxiliary effect contains mono samples in 6171 // 32 bit format. This is to avoid saturation in AudoMixer 6172 // accumulation stage. Saturation is done in EffectModule::process() before 6173 // calling the process in effect engine 6174 size_t numSamples = thread->frameCount(); 6175 int32_t *buffer = new int32_t[numSamples]; 6176 memset(buffer, 0, numSamples * sizeof(int32_t)); 6177 effect->setInBuffer((int16_t *)buffer); 6178 // auxiliary effects output samples to chain input buffer for further processing 6179 // by insert effects 6180 effect->setOutBuffer(mInBuffer); 6181 } else { 6182 // Insert effects are inserted at the end of mEffects vector as they are processed 6183 // after track and auxiliary effects. 6184 // Insert effect order as a function of indicated preference: 6185 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6186 // another effect is present 6187 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6188 // last effect claiming first position 6189 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6190 // first effect claiming last position 6191 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6192 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6193 // already present 6194 6195 int size = (int)mEffects.size(); 6196 int idx_insert = size; 6197 int idx_insert_first = -1; 6198 int idx_insert_last = -1; 6199 6200 for (int i = 0; i < size; i++) { 6201 effect_descriptor_t d = mEffects[i]->desc(); 6202 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6203 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6204 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6205 // check invalid effect chaining combinations 6206 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6207 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6208 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6209 return INVALID_OPERATION; 6210 } 6211 // remember position of first insert effect and by default 6212 // select this as insert position for new effect 6213 if (idx_insert == size) { 6214 idx_insert = i; 6215 } 6216 // remember position of last insert effect claiming 6217 // first position 6218 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6219 idx_insert_first = i; 6220 } 6221 // remember position of first insert effect claiming 6222 // last position 6223 if (iPref == EFFECT_FLAG_INSERT_LAST && 6224 idx_insert_last == -1) { 6225 idx_insert_last = i; 6226 } 6227 } 6228 } 6229 6230 // modify idx_insert from first position if needed 6231 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6232 if (idx_insert_last != -1) { 6233 idx_insert = idx_insert_last; 6234 } else { 6235 idx_insert = size; 6236 } 6237 } else { 6238 if (idx_insert_first != -1) { 6239 idx_insert = idx_insert_first + 1; 6240 } 6241 } 6242 6243 // always read samples from chain input buffer 6244 effect->setInBuffer(mInBuffer); 6245 6246 // if last effect in the chain, output samples to chain 6247 // output buffer, otherwise to chain input buffer 6248 if (idx_insert == size) { 6249 if (idx_insert != 0) { 6250 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6251 mEffects[idx_insert-1]->configure(); 6252 } 6253 effect->setOutBuffer(mOutBuffer); 6254 } else { 6255 effect->setOutBuffer(mInBuffer); 6256 } 6257 mEffects.insertAt(effect, idx_insert); 6258 6259 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6260 } 6261 effect->configure(); 6262 return NO_ERROR; 6263} 6264 6265// removeEffect_l() must be called with PlaybackThread::mLock held 6266size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6267{ 6268 Mutex::Autolock _l(mLock); 6269 int size = (int)mEffects.size(); 6270 int i; 6271 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6272 6273 for (i = 0; i < size; i++) { 6274 if (effect == mEffects[i]) { 6275 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6276 delete[] effect->inBuffer(); 6277 } else { 6278 if (i == size - 1 && i != 0) { 6279 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6280 mEffects[i - 1]->configure(); 6281 } 6282 } 6283 mEffects.removeAt(i); 6284 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6285 break; 6286 } 6287 } 6288 6289 return mEffects.size(); 6290} 6291 6292// setDevice_l() must be called with PlaybackThread::mLock held 6293void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6294{ 6295 size_t size = mEffects.size(); 6296 for (size_t i = 0; i < size; i++) { 6297 mEffects[i]->setDevice(device); 6298 } 6299} 6300 6301// setMode_l() must be called with PlaybackThread::mLock held 6302void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6303{ 6304 size_t size = mEffects.size(); 6305 for (size_t i = 0; i < size; i++) { 6306 mEffects[i]->setMode(mode); 6307 } 6308} 6309 6310// setVolume_l() must be called with PlaybackThread::mLock held 6311bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6312{ 6313 uint32_t newLeft = *left; 6314 uint32_t newRight = *right; 6315 bool hasControl = false; 6316 int ctrlIdx = -1; 6317 size_t size = mEffects.size(); 6318 6319 // first update volume controller 6320 for (size_t i = size; i > 0; i--) { 6321 if (mEffects[i - 1]->isProcessEnabled() && 6322 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6323 ctrlIdx = i - 1; 6324 hasControl = true; 6325 break; 6326 } 6327 } 6328 6329 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6330 if (hasControl) { 6331 *left = mNewLeftVolume; 6332 *right = mNewRightVolume; 6333 } 6334 return hasControl; 6335 } 6336 6337 mVolumeCtrlIdx = ctrlIdx; 6338 mLeftVolume = newLeft; 6339 mRightVolume = newRight; 6340 6341 // second get volume update from volume controller 6342 if (ctrlIdx >= 0) { 6343 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6344 mNewLeftVolume = newLeft; 6345 mNewRightVolume = newRight; 6346 } 6347 // then indicate volume to all other effects in chain. 6348 // Pass altered volume to effects before volume controller 6349 // and requested volume to effects after controller 6350 uint32_t lVol = newLeft; 6351 uint32_t rVol = newRight; 6352 6353 for (size_t i = 0; i < size; i++) { 6354 if ((int)i == ctrlIdx) continue; 6355 // this also works for ctrlIdx == -1 when there is no volume controller 6356 if ((int)i > ctrlIdx) { 6357 lVol = *left; 6358 rVol = *right; 6359 } 6360 mEffects[i]->setVolume(&lVol, &rVol, false); 6361 } 6362 *left = newLeft; 6363 *right = newRight; 6364 6365 return hasControl; 6366} 6367 6368status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6369{ 6370 const size_t SIZE = 256; 6371 char buffer[SIZE]; 6372 String8 result; 6373 6374 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6375 result.append(buffer); 6376 6377 bool locked = tryLock(mLock); 6378 // failed to lock - AudioFlinger is probably deadlocked 6379 if (!locked) { 6380 result.append("\tCould not lock mutex:\n"); 6381 } 6382 6383 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6384 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6385 mEffects.size(), 6386 (uint32_t)mInBuffer, 6387 (uint32_t)mOutBuffer, 6388 mActiveTrackCnt); 6389 result.append(buffer); 6390 write(fd, result.string(), result.size()); 6391 6392 for (size_t i = 0; i < mEffects.size(); ++i) { 6393 sp<EffectModule> effect = mEffects[i]; 6394 if (effect != 0) { 6395 effect->dump(fd, args); 6396 } 6397 } 6398 6399 if (locked) { 6400 mLock.unlock(); 6401 } 6402 6403 return NO_ERROR; 6404} 6405 6406#undef LOG_TAG 6407#define LOG_TAG "AudioFlinger" 6408 6409// ---------------------------------------------------------------------------- 6410 6411status_t AudioFlinger::onTransact( 6412 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6413{ 6414 return BnAudioFlinger::onTransact(code, data, reply, flags); 6415} 6416 6417}; // namespace android 6418