AudioFlinger.cpp revision 42968939dfce0954d6540011199045ec4ed7de80
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/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38 39#include <media/AudioTrack.h> 40#include <media/AudioRecord.h> 41#include <media/IMediaPlayerService.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <audio_effects/effect_visualizer.h> 54#include <audio_effects/effect_ns.h> 55#include <audio_effects/effect_aec.h> 56 57#include <cpustats/ThreadCpuUsage.h> 58#include <powermanager/PowerManager.h> 59// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 60 61// ---------------------------------------------------------------------------- 62 63 64namespace android { 65 66static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 67static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 68 69//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 70static const float MAX_GAIN = 4096.0f; 71static const float MAX_GAIN_INT = 0x1000; 72 73// retry counts for buffer fill timeout 74// 50 * ~20msecs = 1 second 75static const int8_t kMaxTrackRetries = 50; 76static const int8_t kMaxTrackStartupRetries = 50; 77// allow less retry attempts on direct output thread. 78// direct outputs can be a scarce resource in audio hardware and should 79// be released as quickly as possible. 80static const int8_t kMaxTrackRetriesDirect = 2; 81 82static const int kDumpLockRetries = 50; 83static const int kDumpLockSleepUs = 20000; 84 85// don't warn about blocked writes or record buffer overflows more often than this 86static const nsecs_t kWarningThrottleNs = seconds(5); 87 88// RecordThread loop sleep time upon application overrun or audio HAL read error 89static const int kRecordThreadSleepUs = 5000; 90 91// maximum time to wait for setParameters to complete 92static const nsecs_t kSetParametersTimeoutNs = seconds(2); 93 94// minimum sleep time for the mixer thread loop when tracks are active but in underrun 95static const uint32_t kMinThreadSleepTimeUs = 5000; 96// maximum divider applied to the active sleep time in the mixer thread loop 97static const uint32_t kMaxThreadSleepTimeShift = 2; 98 99 100// ---------------------------------------------------------------------------- 101 102static bool recordingAllowed() { 103 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 104 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 105 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 106 return ok; 107} 108 109static bool settingsAllowed() { 110 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 111 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 112 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 113 return ok; 114} 115 116// To collect the amplifier usage 117static void addBatteryData(uint32_t params) { 118 sp<IBinder> binder = 119 defaultServiceManager()->getService(String16("media.player")); 120 sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder); 121 if (service.get() == NULL) { 122 LOGW("Cannot connect to the MediaPlayerService for battery tracking"); 123 return; 124 } 125 126 service->addBatteryData(params); 127} 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 164 mBtNrecIsOff(false) 165{ 166} 167 168void AudioFlinger::onFirstRef() 169{ 170 int rc = 0; 171 172 Mutex::Autolock _l(mLock); 173 174 /* TODO: move all this work into an Init() function */ 175 mHardwareStatus = AUDIO_HW_IDLE; 176 177 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 178 const hw_module_t *mod; 179 audio_hw_device_t *dev; 180 181 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 182 if (rc) 183 continue; 184 185 LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 186 mod->name, mod->id); 187 mAudioHwDevs.push(dev); 188 189 if (!mPrimaryHardwareDev) { 190 mPrimaryHardwareDev = dev; 191 LOGI("Using '%s' (%s.%s) as the primary audio interface", 192 mod->name, mod->id, audio_interfaces[i]); 193 } 194 } 195 196 mHardwareStatus = AUDIO_HW_INIT; 197 198 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 199 LOGE("Primary audio interface not found"); 200 return; 201 } 202 203 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 204 audio_hw_device_t *dev = mAudioHwDevs[i]; 205 206 mHardwareStatus = AUDIO_HW_INIT; 207 rc = dev->init_check(dev); 208 if (rc == 0) { 209 AutoMutex lock(mHardwareLock); 210 211 mMode = AUDIO_MODE_NORMAL; 212 mHardwareStatus = AUDIO_HW_SET_MODE; 213 dev->set_mode(dev, mMode); 214 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 215 dev->set_master_volume(dev, 1.0f); 216 mHardwareStatus = AUDIO_HW_IDLE; 217 } 218 } 219} 220 221status_t AudioFlinger::initCheck() const 222{ 223 Mutex::Autolock _l(mLock); 224 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 225 return NO_INIT; 226 return NO_ERROR; 227} 228 229AudioFlinger::~AudioFlinger() 230{ 231 int num_devs = mAudioHwDevs.size(); 232 233 while (!mRecordThreads.isEmpty()) { 234 // closeInput() will remove first entry from mRecordThreads 235 closeInput(mRecordThreads.keyAt(0)); 236 } 237 while (!mPlaybackThreads.isEmpty()) { 238 // closeOutput() will remove first entry from mPlaybackThreads 239 closeOutput(mPlaybackThreads.keyAt(0)); 240 } 241 242 for (int i = 0; i < num_devs; i++) { 243 audio_hw_device_t *dev = mAudioHwDevs[i]; 244 audio_hw_device_close(dev); 245 } 246 mAudioHwDevs.clear(); 247} 248 249audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 250{ 251 /* first matching HW device is returned */ 252 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 253 audio_hw_device_t *dev = mAudioHwDevs[i]; 254 if ((dev->get_supported_devices(dev) & devices) == devices) 255 return dev; 256 } 257 return NULL; 258} 259 260status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 261{ 262 const size_t SIZE = 256; 263 char buffer[SIZE]; 264 String8 result; 265 266 result.append("Clients:\n"); 267 for (size_t i = 0; i < mClients.size(); ++i) { 268 wp<Client> wClient = mClients.valueAt(i); 269 if (wClient != 0) { 270 sp<Client> client = wClient.promote(); 271 if (client != 0) { 272 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 273 result.append(buffer); 274 } 275 } 276 } 277 278 result.append("Global session refs:\n"); 279 result.append(" session pid cnt\n"); 280 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 281 AudioSessionRef *r = mAudioSessionRefs[i]; 282 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 283 result.append(buffer); 284 } 285 write(fd, result.string(), result.size()); 286 return NO_ERROR; 287} 288 289 290status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 291{ 292 const size_t SIZE = 256; 293 char buffer[SIZE]; 294 String8 result; 295 int hardwareStatus = mHardwareStatus; 296 297 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 298 result.append(buffer); 299 write(fd, result.string(), result.size()); 300 return NO_ERROR; 301} 302 303status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 304{ 305 const size_t SIZE = 256; 306 char buffer[SIZE]; 307 String8 result; 308 snprintf(buffer, SIZE, "Permission Denial: " 309 "can't dump AudioFlinger from pid=%d, uid=%d\n", 310 IPCThreadState::self()->getCallingPid(), 311 IPCThreadState::self()->getCallingUid()); 312 result.append(buffer); 313 write(fd, result.string(), result.size()); 314 return NO_ERROR; 315} 316 317static bool tryLock(Mutex& mutex) 318{ 319 bool locked = false; 320 for (int i = 0; i < kDumpLockRetries; ++i) { 321 if (mutex.tryLock() == NO_ERROR) { 322 locked = true; 323 break; 324 } 325 usleep(kDumpLockSleepUs); 326 } 327 return locked; 328} 329 330status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 331{ 332 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 333 dumpPermissionDenial(fd, args); 334 } else { 335 // get state of hardware lock 336 bool hardwareLocked = tryLock(mHardwareLock); 337 if (!hardwareLocked) { 338 String8 result(kHardwareLockedString); 339 write(fd, result.string(), result.size()); 340 } else { 341 mHardwareLock.unlock(); 342 } 343 344 bool locked = tryLock(mLock); 345 346 // failed to lock - AudioFlinger is probably deadlocked 347 if (!locked) { 348 String8 result(kDeadlockedString); 349 write(fd, result.string(), result.size()); 350 } 351 352 dumpClients(fd, args); 353 dumpInternals(fd, args); 354 355 // dump playback threads 356 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 357 mPlaybackThreads.valueAt(i)->dump(fd, args); 358 } 359 360 // dump record threads 361 for (size_t i = 0; i < mRecordThreads.size(); i++) { 362 mRecordThreads.valueAt(i)->dump(fd, args); 363 } 364 365 // dump all hardware devs 366 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 367 audio_hw_device_t *dev = mAudioHwDevs[i]; 368 dev->dump(dev, fd); 369 } 370 if (locked) mLock.unlock(); 371 } 372 return NO_ERROR; 373} 374 375 376// IAudioFlinger interface 377 378 379sp<IAudioTrack> AudioFlinger::createTrack( 380 pid_t pid, 381 int streamType, 382 uint32_t sampleRate, 383 uint32_t format, 384 uint32_t channelMask, 385 int frameCount, 386 uint32_t flags, 387 const sp<IMemory>& sharedBuffer, 388 int output, 389 int *sessionId, 390 status_t *status) 391{ 392 sp<PlaybackThread::Track> track; 393 sp<TrackHandle> trackHandle; 394 sp<Client> client; 395 wp<Client> wclient; 396 status_t lStatus; 397 int lSessionId; 398 399 if (streamType >= AUDIO_STREAM_CNT) { 400 LOGE("createTrack() invalid stream type %d", streamType); 401 lStatus = BAD_VALUE; 402 goto Exit; 403 } 404 405 { 406 Mutex::Autolock _l(mLock); 407 PlaybackThread *thread = checkPlaybackThread_l(output); 408 PlaybackThread *effectThread = NULL; 409 if (thread == NULL) { 410 LOGE("unknown output thread"); 411 lStatus = BAD_VALUE; 412 goto Exit; 413 } 414 415 wclient = mClients.valueFor(pid); 416 417 if (wclient != NULL) { 418 client = wclient.promote(); 419 } else { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 425 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 426 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 427 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 428 if (mPlaybackThreads.keyAt(i) != output) { 429 // prevent same audio session on different output threads 430 uint32_t sessions = t->hasAudioSession(*sessionId); 431 if (sessions & PlaybackThread::TRACK_SESSION) { 432 LOGE("createTrack() session ID %d already in use", *sessionId); 433 lStatus = BAD_VALUE; 434 goto Exit; 435 } 436 // check if an effect with same session ID is waiting for a track to be created 437 if (sessions & PlaybackThread::EFFECT_SESSION) { 438 effectThread = t.get(); 439 } 440 } 441 } 442 lSessionId = *sessionId; 443 } else { 444 // if no audio session id is provided, create one here 445 lSessionId = nextUniqueId(); 446 if (sessionId != NULL) { 447 *sessionId = lSessionId; 448 } 449 } 450 ALOGV("createTrack() lSessionId: %d", lSessionId); 451 452 track = thread->createTrack_l(client, streamType, sampleRate, format, 453 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 454 455 // move effect chain to this output thread if an effect on same session was waiting 456 // for a track to be created 457 if (lStatus == NO_ERROR && effectThread != NULL) { 458 Mutex::Autolock _dl(thread->mLock); 459 Mutex::Autolock _sl(effectThread->mLock); 460 moveEffectChain_l(lSessionId, effectThread, thread, true); 461 } 462 } 463 if (lStatus == NO_ERROR) { 464 trackHandle = new TrackHandle(track); 465 } else { 466 // remove local strong reference to Client before deleting the Track so that the Client 467 // destructor is called by the TrackBase destructor with mLock held 468 client.clear(); 469 track.clear(); 470 } 471 472Exit: 473 if(status) { 474 *status = lStatus; 475 } 476 return trackHandle; 477} 478 479uint32_t AudioFlinger::sampleRate(int output) const 480{ 481 Mutex::Autolock _l(mLock); 482 PlaybackThread *thread = checkPlaybackThread_l(output); 483 if (thread == NULL) { 484 LOGW("sampleRate() unknown thread %d", output); 485 return 0; 486 } 487 return thread->sampleRate(); 488} 489 490int AudioFlinger::channelCount(int output) const 491{ 492 Mutex::Autolock _l(mLock); 493 PlaybackThread *thread = checkPlaybackThread_l(output); 494 if (thread == NULL) { 495 LOGW("channelCount() unknown thread %d", output); 496 return 0; 497 } 498 return thread->channelCount(); 499} 500 501uint32_t AudioFlinger::format(int output) const 502{ 503 Mutex::Autolock _l(mLock); 504 PlaybackThread *thread = checkPlaybackThread_l(output); 505 if (thread == NULL) { 506 LOGW("format() unknown thread %d", output); 507 return 0; 508 } 509 return thread->format(); 510} 511 512size_t AudioFlinger::frameCount(int output) const 513{ 514 Mutex::Autolock _l(mLock); 515 PlaybackThread *thread = checkPlaybackThread_l(output); 516 if (thread == NULL) { 517 LOGW("frameCount() unknown thread %d", output); 518 return 0; 519 } 520 return thread->frameCount(); 521} 522 523uint32_t AudioFlinger::latency(int output) const 524{ 525 Mutex::Autolock _l(mLock); 526 PlaybackThread *thread = checkPlaybackThread_l(output); 527 if (thread == NULL) { 528 LOGW("latency() unknown thread %d", output); 529 return 0; 530 } 531 return thread->latency(); 532} 533 534status_t AudioFlinger::setMasterVolume(float value) 535{ 536 status_t ret = initCheck(); 537 if (ret != NO_ERROR) { 538 return ret; 539 } 540 541 // check calling permissions 542 if (!settingsAllowed()) { 543 return PERMISSION_DENIED; 544 } 545 546 // when hw supports master volume, don't scale in sw mixer 547 { // scope for the lock 548 AutoMutex lock(mHardwareLock); 549 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 550 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 551 value = 1.0f; 552 } 553 mHardwareStatus = AUDIO_HW_IDLE; 554 } 555 556 Mutex::Autolock _l(mLock); 557 mMasterVolume = value; 558 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 559 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 560 561 return NO_ERROR; 562} 563 564status_t AudioFlinger::setMode(int mode) 565{ 566 status_t ret = initCheck(); 567 if (ret != NO_ERROR) { 568 return ret; 569 } 570 571 // check calling permissions 572 if (!settingsAllowed()) { 573 return PERMISSION_DENIED; 574 } 575 if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) { 576 LOGW("Illegal value: setMode(%d)", mode); 577 return BAD_VALUE; 578 } 579 580 { // scope for the lock 581 AutoMutex lock(mHardwareLock); 582 mHardwareStatus = AUDIO_HW_SET_MODE; 583 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 584 mHardwareStatus = AUDIO_HW_IDLE; 585 } 586 587 if (NO_ERROR == ret) { 588 Mutex::Autolock _l(mLock); 589 mMode = mode; 590 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 591 mPlaybackThreads.valueAt(i)->setMode(mode); 592 } 593 594 return ret; 595} 596 597status_t AudioFlinger::setMicMute(bool state) 598{ 599 status_t ret = initCheck(); 600 if (ret != NO_ERROR) { 601 return ret; 602 } 603 604 // check calling permissions 605 if (!settingsAllowed()) { 606 return PERMISSION_DENIED; 607 } 608 609 AutoMutex lock(mHardwareLock); 610 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 611 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 612 mHardwareStatus = AUDIO_HW_IDLE; 613 return ret; 614} 615 616bool AudioFlinger::getMicMute() const 617{ 618 status_t ret = initCheck(); 619 if (ret != NO_ERROR) { 620 return false; 621 } 622 623 bool state = AUDIO_MODE_INVALID; 624 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 625 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 626 mHardwareStatus = AUDIO_HW_IDLE; 627 return state; 628} 629 630status_t AudioFlinger::setMasterMute(bool muted) 631{ 632 // check calling permissions 633 if (!settingsAllowed()) { 634 return PERMISSION_DENIED; 635 } 636 637 Mutex::Autolock _l(mLock); 638 mMasterMute = muted; 639 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 640 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 641 642 return NO_ERROR; 643} 644 645float AudioFlinger::masterVolume() const 646{ 647 return mMasterVolume; 648} 649 650bool AudioFlinger::masterMute() const 651{ 652 return mMasterMute; 653} 654 655status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 656{ 657 // check calling permissions 658 if (!settingsAllowed()) { 659 return PERMISSION_DENIED; 660 } 661 662 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 663 LOGE("setStreamVolume() invalid stream %d", stream); 664 return BAD_VALUE; 665 } 666 667 AutoMutex lock(mLock); 668 PlaybackThread *thread = NULL; 669 if (output) { 670 thread = checkPlaybackThread_l(output); 671 if (thread == NULL) { 672 return BAD_VALUE; 673 } 674 } 675 676 mStreamTypes[stream].volume = value; 677 678 if (thread == NULL) { 679 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 680 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 681 } 682 } else { 683 thread->setStreamVolume(stream, value); 684 } 685 686 return NO_ERROR; 687} 688 689status_t AudioFlinger::setStreamMute(int stream, bool muted) 690{ 691 // check calling permissions 692 if (!settingsAllowed()) { 693 return PERMISSION_DENIED; 694 } 695 696 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || 697 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 698 LOGE("setStreamMute() invalid stream %d", stream); 699 return BAD_VALUE; 700 } 701 702 AutoMutex lock(mLock); 703 mStreamTypes[stream].mute = muted; 704 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 705 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 706 707 return NO_ERROR; 708} 709 710float AudioFlinger::streamVolume(int stream, int output) const 711{ 712 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 713 return 0.0f; 714 } 715 716 AutoMutex lock(mLock); 717 float volume; 718 if (output) { 719 PlaybackThread *thread = checkPlaybackThread_l(output); 720 if (thread == NULL) { 721 return 0.0f; 722 } 723 volume = thread->streamVolume(stream); 724 } else { 725 volume = mStreamTypes[stream].volume; 726 } 727 728 return volume; 729} 730 731bool AudioFlinger::streamMute(int stream) const 732{ 733 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 734 return true; 735 } 736 737 return mStreamTypes[stream].mute; 738} 739 740status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 741{ 742 status_t result; 743 744 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 745 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 746 // check calling permissions 747 if (!settingsAllowed()) { 748 return PERMISSION_DENIED; 749 } 750 751 // ioHandle == 0 means the parameters are global to the audio hardware interface 752 if (ioHandle == 0) { 753 AutoMutex lock(mHardwareLock); 754 mHardwareStatus = AUDIO_SET_PARAMETER; 755 status_t final_result = NO_ERROR; 756 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 757 audio_hw_device_t *dev = mAudioHwDevs[i]; 758 result = dev->set_parameters(dev, keyValuePairs.string()); 759 final_result = result ?: final_result; 760 } 761 mHardwareStatus = AUDIO_HW_IDLE; 762 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 763 AudioParameter param = AudioParameter(keyValuePairs); 764 String8 value; 765 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 766 Mutex::Autolock _l(mLock); 767 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 768 if (mBtNrecIsOff != btNrecIsOff) { 769 for (size_t i = 0; i < mRecordThreads.size(); i++) { 770 sp<RecordThread> thread = mRecordThreads.valueAt(i); 771 RecordThread::RecordTrack *track = thread->track(); 772 if (track != NULL) { 773 audio_devices_t device = (audio_devices_t)( 774 thread->device() & AUDIO_DEVICE_IN_ALL); 775 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 776 thread->setEffectSuspended(FX_IID_AEC, 777 suspend, 778 track->sessionId()); 779 thread->setEffectSuspended(FX_IID_NS, 780 suspend, 781 track->sessionId()); 782 } 783 } 784 mBtNrecIsOff = btNrecIsOff; 785 } 786 } 787 return final_result; 788 } 789 790 // hold a strong ref on thread in case closeOutput() or closeInput() is called 791 // and the thread is exited once the lock is released 792 sp<ThreadBase> thread; 793 { 794 Mutex::Autolock _l(mLock); 795 thread = checkPlaybackThread_l(ioHandle); 796 if (thread == NULL) { 797 thread = checkRecordThread_l(ioHandle); 798 } else if (thread.get() == primaryPlaybackThread_l()) { 799 // indicate output device change to all input threads for pre processing 800 AudioParameter param = AudioParameter(keyValuePairs); 801 int value; 802 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 803 for (size_t i = 0; i < mRecordThreads.size(); i++) { 804 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 805 } 806 } 807 } 808 } 809 if (thread != NULL) { 810 result = thread->setParameters(keyValuePairs); 811 return result; 812 } 813 return BAD_VALUE; 814} 815 816String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 817{ 818// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 819// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 820 821 if (ioHandle == 0) { 822 String8 out_s8; 823 824 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 825 audio_hw_device_t *dev = mAudioHwDevs[i]; 826 char *s = dev->get_parameters(dev, keys.string()); 827 out_s8 += String8(s); 828 free(s); 829 } 830 return out_s8; 831 } 832 833 Mutex::Autolock _l(mLock); 834 835 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 836 if (playbackThread != NULL) { 837 return playbackThread->getParameters(keys); 838 } 839 RecordThread *recordThread = checkRecordThread_l(ioHandle); 840 if (recordThread != NULL) { 841 return recordThread->getParameters(keys); 842 } 843 return String8(""); 844} 845 846size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 847{ 848 status_t ret = initCheck(); 849 if (ret != NO_ERROR) { 850 return 0; 851 } 852 853 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 854} 855 856unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 857{ 858 if (ioHandle == 0) { 859 return 0; 860 } 861 862 Mutex::Autolock _l(mLock); 863 864 RecordThread *recordThread = checkRecordThread_l(ioHandle); 865 if (recordThread != NULL) { 866 return recordThread->getInputFramesLost(); 867 } 868 return 0; 869} 870 871status_t AudioFlinger::setVoiceVolume(float value) 872{ 873 status_t ret = initCheck(); 874 if (ret != NO_ERROR) { 875 return ret; 876 } 877 878 // check calling permissions 879 if (!settingsAllowed()) { 880 return PERMISSION_DENIED; 881 } 882 883 AutoMutex lock(mHardwareLock); 884 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 885 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 886 mHardwareStatus = AUDIO_HW_IDLE; 887 888 return ret; 889} 890 891status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 892{ 893 status_t status; 894 895 Mutex::Autolock _l(mLock); 896 897 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 898 if (playbackThread != NULL) { 899 return playbackThread->getRenderPosition(halFrames, dspFrames); 900 } 901 902 return BAD_VALUE; 903} 904 905void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 906{ 907 908 Mutex::Autolock _l(mLock); 909 910 int pid = IPCThreadState::self()->getCallingPid(); 911 if (mNotificationClients.indexOfKey(pid) < 0) { 912 sp<NotificationClient> notificationClient = new NotificationClient(this, 913 client, 914 pid); 915 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 916 917 mNotificationClients.add(pid, notificationClient); 918 919 sp<IBinder> binder = client->asBinder(); 920 binder->linkToDeath(notificationClient); 921 922 // the config change is always sent from playback or record threads to avoid deadlock 923 // with AudioSystem::gLock 924 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 925 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 926 } 927 928 for (size_t i = 0; i < mRecordThreads.size(); i++) { 929 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 930 } 931 } 932} 933 934void AudioFlinger::removeNotificationClient(pid_t pid) 935{ 936 Mutex::Autolock _l(mLock); 937 938 int index = mNotificationClients.indexOfKey(pid); 939 if (index >= 0) { 940 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 941 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 942 mNotificationClients.removeItem(pid); 943 } 944 945 ALOGV("%d died, releasing its sessions", pid); 946 int num = mAudioSessionRefs.size(); 947 bool removed = false; 948 for (int i = 0; i< num; i++) { 949 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 950 ALOGV(" pid %d @ %d", ref->pid, i); 951 if (ref->pid == pid) { 952 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 953 mAudioSessionRefs.removeAt(i); 954 delete ref; 955 removed = true; 956 i--; 957 num--; 958 } 959 } 960 if (removed) { 961 purgeStaleEffects_l(); 962 } 963} 964 965// audioConfigChanged_l() must be called with AudioFlinger::mLock held 966void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 967{ 968 size_t size = mNotificationClients.size(); 969 for (size_t i = 0; i < size; i++) { 970 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 971 } 972} 973 974// removeClient_l() must be called with AudioFlinger::mLock held 975void AudioFlinger::removeClient_l(pid_t pid) 976{ 977 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 978 mClients.removeItem(pid); 979} 980 981 982// ---------------------------------------------------------------------------- 983 984AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 985 : Thread(false), 986 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 987 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), 988 mDevice(device) 989{ 990 mDeathRecipient = new PMDeathRecipient(this); 991} 992 993AudioFlinger::ThreadBase::~ThreadBase() 994{ 995 mParamCond.broadcast(); 996 // do not lock the mutex in destructor 997 releaseWakeLock_l(); 998 if (mPowerManager != 0) { 999 sp<IBinder> binder = mPowerManager->asBinder(); 1000 binder->unlinkToDeath(mDeathRecipient); 1001 } 1002} 1003 1004void AudioFlinger::ThreadBase::exit() 1005{ 1006 // keep a strong ref on ourself so that we won't get 1007 // destroyed in the middle of requestExitAndWait() 1008 sp <ThreadBase> strongMe = this; 1009 1010 ALOGV("ThreadBase::exit"); 1011 { 1012 AutoMutex lock(&mLock); 1013 mExiting = true; 1014 requestExit(); 1015 mWaitWorkCV.signal(); 1016 } 1017 requestExitAndWait(); 1018} 1019 1020uint32_t AudioFlinger::ThreadBase::sampleRate() const 1021{ 1022 return mSampleRate; 1023} 1024 1025int AudioFlinger::ThreadBase::channelCount() const 1026{ 1027 return (int)mChannelCount; 1028} 1029 1030uint32_t AudioFlinger::ThreadBase::format() const 1031{ 1032 return mFormat; 1033} 1034 1035size_t AudioFlinger::ThreadBase::frameCount() const 1036{ 1037 return mFrameCount; 1038} 1039 1040status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1041{ 1042 status_t status; 1043 1044 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1045 Mutex::Autolock _l(mLock); 1046 1047 mNewParameters.add(keyValuePairs); 1048 mWaitWorkCV.signal(); 1049 // wait condition with timeout in case the thread loop has exited 1050 // before the request could be processed 1051 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1052 status = mParamStatus; 1053 mWaitWorkCV.signal(); 1054 } else { 1055 status = TIMED_OUT; 1056 } 1057 return status; 1058} 1059 1060void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1061{ 1062 Mutex::Autolock _l(mLock); 1063 sendConfigEvent_l(event, param); 1064} 1065 1066// sendConfigEvent_l() must be called with ThreadBase::mLock held 1067void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1068{ 1069 ConfigEvent *configEvent = new ConfigEvent(); 1070 configEvent->mEvent = event; 1071 configEvent->mParam = param; 1072 mConfigEvents.add(configEvent); 1073 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1074 mWaitWorkCV.signal(); 1075} 1076 1077void AudioFlinger::ThreadBase::processConfigEvents() 1078{ 1079 mLock.lock(); 1080 while(!mConfigEvents.isEmpty()) { 1081 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1082 ConfigEvent *configEvent = mConfigEvents[0]; 1083 mConfigEvents.removeAt(0); 1084 // release mLock before locking AudioFlinger mLock: lock order is always 1085 // AudioFlinger then ThreadBase to avoid cross deadlock 1086 mLock.unlock(); 1087 mAudioFlinger->mLock.lock(); 1088 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 1089 mAudioFlinger->mLock.unlock(); 1090 delete configEvent; 1091 mLock.lock(); 1092 } 1093 mLock.unlock(); 1094} 1095 1096status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1097{ 1098 const size_t SIZE = 256; 1099 char buffer[SIZE]; 1100 String8 result; 1101 1102 bool locked = tryLock(mLock); 1103 if (!locked) { 1104 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1105 write(fd, buffer, strlen(buffer)); 1106 } 1107 1108 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1109 result.append(buffer); 1110 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1111 result.append(buffer); 1112 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1113 result.append(buffer); 1114 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1115 result.append(buffer); 1116 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1117 result.append(buffer); 1118 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1119 result.append(buffer); 1120 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 1121 result.append(buffer); 1122 1123 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1124 result.append(buffer); 1125 result.append(" Index Command"); 1126 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1127 snprintf(buffer, SIZE, "\n %02d ", i); 1128 result.append(buffer); 1129 result.append(mNewParameters[i]); 1130 } 1131 1132 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1133 result.append(buffer); 1134 snprintf(buffer, SIZE, " Index event param\n"); 1135 result.append(buffer); 1136 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1137 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 1138 result.append(buffer); 1139 } 1140 result.append("\n"); 1141 1142 write(fd, result.string(), result.size()); 1143 1144 if (locked) { 1145 mLock.unlock(); 1146 } 1147 return NO_ERROR; 1148} 1149 1150status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1151{ 1152 const size_t SIZE = 256; 1153 char buffer[SIZE]; 1154 String8 result; 1155 1156 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1157 write(fd, buffer, strlen(buffer)); 1158 1159 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1160 sp<EffectChain> chain = mEffectChains[i]; 1161 if (chain != 0) { 1162 chain->dump(fd, args); 1163 } 1164 } 1165 return NO_ERROR; 1166} 1167 1168void AudioFlinger::ThreadBase::acquireWakeLock() 1169{ 1170 Mutex::Autolock _l(mLock); 1171 acquireWakeLock_l(); 1172} 1173 1174void AudioFlinger::ThreadBase::acquireWakeLock_l() 1175{ 1176 if (mPowerManager == 0) { 1177 // use checkService() to avoid blocking if power service is not up yet 1178 sp<IBinder> binder = 1179 defaultServiceManager()->checkService(String16("power")); 1180 if (binder == 0) { 1181 LOGW("Thread %s cannot connect to the power manager service", mName); 1182 } else { 1183 mPowerManager = interface_cast<IPowerManager>(binder); 1184 binder->linkToDeath(mDeathRecipient); 1185 } 1186 } 1187 if (mPowerManager != 0) { 1188 sp<IBinder> binder = new BBinder(); 1189 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1190 binder, 1191 String16(mName)); 1192 if (status == NO_ERROR) { 1193 mWakeLockToken = binder; 1194 } 1195 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1196 } 1197} 1198 1199void AudioFlinger::ThreadBase::releaseWakeLock() 1200{ 1201 Mutex::Autolock _l(mLock); 1202 releaseWakeLock_l(); 1203} 1204 1205void AudioFlinger::ThreadBase::releaseWakeLock_l() 1206{ 1207 if (mWakeLockToken != 0) { 1208 ALOGV("releaseWakeLock_l() %s", mName); 1209 if (mPowerManager != 0) { 1210 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1211 } 1212 mWakeLockToken.clear(); 1213 } 1214} 1215 1216void AudioFlinger::ThreadBase::clearPowerManager() 1217{ 1218 Mutex::Autolock _l(mLock); 1219 releaseWakeLock_l(); 1220 mPowerManager.clear(); 1221} 1222 1223void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1224{ 1225 sp<ThreadBase> thread = mThread.promote(); 1226 if (thread != 0) { 1227 thread->clearPowerManager(); 1228 } 1229 LOGW("power manager service died !!!"); 1230} 1231 1232void AudioFlinger::ThreadBase::setEffectSuspended( 1233 const effect_uuid_t *type, bool suspend, int sessionId) 1234{ 1235 Mutex::Autolock _l(mLock); 1236 setEffectSuspended_l(type, suspend, sessionId); 1237} 1238 1239void AudioFlinger::ThreadBase::setEffectSuspended_l( 1240 const effect_uuid_t *type, bool suspend, int sessionId) 1241{ 1242 sp<EffectChain> chain; 1243 chain = getEffectChain_l(sessionId); 1244 if (chain != 0) { 1245 if (type != NULL) { 1246 chain->setEffectSuspended_l(type, suspend); 1247 } else { 1248 chain->setEffectSuspendedAll_l(suspend); 1249 } 1250 } 1251 1252 updateSuspendedSessions_l(type, suspend, sessionId); 1253} 1254 1255void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1256{ 1257 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1258 if (index < 0) { 1259 return; 1260 } 1261 1262 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1263 mSuspendedSessions.editValueAt(index); 1264 1265 for (size_t i = 0; i < sessionEffects.size(); i++) { 1266 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1267 for (int j = 0; j < desc->mRefCount; j++) { 1268 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1269 chain->setEffectSuspendedAll_l(true); 1270 } else { 1271 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1272 desc->mType.timeLow); 1273 chain->setEffectSuspended_l(&desc->mType, true); 1274 } 1275 } 1276 } 1277} 1278 1279void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1280 bool suspend, 1281 int sessionId) 1282{ 1283 int index = mSuspendedSessions.indexOfKey(sessionId); 1284 1285 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1286 1287 if (suspend) { 1288 if (index >= 0) { 1289 sessionEffects = mSuspendedSessions.editValueAt(index); 1290 } else { 1291 mSuspendedSessions.add(sessionId, sessionEffects); 1292 } 1293 } else { 1294 if (index < 0) { 1295 return; 1296 } 1297 sessionEffects = mSuspendedSessions.editValueAt(index); 1298 } 1299 1300 1301 int key = EffectChain::kKeyForSuspendAll; 1302 if (type != NULL) { 1303 key = type->timeLow; 1304 } 1305 index = sessionEffects.indexOfKey(key); 1306 1307 sp <SuspendedSessionDesc> desc; 1308 if (suspend) { 1309 if (index >= 0) { 1310 desc = sessionEffects.valueAt(index); 1311 } else { 1312 desc = new SuspendedSessionDesc(); 1313 if (type != NULL) { 1314 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1315 } 1316 sessionEffects.add(key, desc); 1317 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1318 } 1319 desc->mRefCount++; 1320 } else { 1321 if (index < 0) { 1322 return; 1323 } 1324 desc = sessionEffects.valueAt(index); 1325 if (--desc->mRefCount == 0) { 1326 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1327 sessionEffects.removeItemsAt(index); 1328 if (sessionEffects.isEmpty()) { 1329 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1330 sessionId); 1331 mSuspendedSessions.removeItem(sessionId); 1332 } 1333 } 1334 } 1335 if (!sessionEffects.isEmpty()) { 1336 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1337 } 1338} 1339 1340void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1341 bool enabled, 1342 int sessionId) 1343{ 1344 Mutex::Autolock _l(mLock); 1345 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1346} 1347 1348void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1349 bool enabled, 1350 int sessionId) 1351{ 1352 if (mType != RECORD) { 1353 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1354 // another session. This gives the priority to well behaved effect control panels 1355 // and applications not using global effects. 1356 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1357 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1358 } 1359 } 1360 1361 sp<EffectChain> chain = getEffectChain_l(sessionId); 1362 if (chain != 0) { 1363 chain->checkSuspendOnEffectEnabled(effect, enabled); 1364 } 1365} 1366 1367// ---------------------------------------------------------------------------- 1368 1369AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1370 AudioStreamOut* output, 1371 int id, 1372 uint32_t device) 1373 : ThreadBase(audioFlinger, id, device), 1374 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 1375 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1376{ 1377 snprintf(mName, kNameLength, "AudioOut_%d", id); 1378 1379 readOutputParameters(); 1380 1381 mMasterVolume = mAudioFlinger->masterVolume(); 1382 mMasterMute = mAudioFlinger->masterMute(); 1383 1384 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 1385 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1386 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1387 mStreamTypes[stream].valid = true; 1388 } 1389} 1390 1391AudioFlinger::PlaybackThread::~PlaybackThread() 1392{ 1393 delete [] mMixBuffer; 1394} 1395 1396status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1397{ 1398 dumpInternals(fd, args); 1399 dumpTracks(fd, args); 1400 dumpEffectChains(fd, args); 1401 return NO_ERROR; 1402} 1403 1404status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1405{ 1406 const size_t SIZE = 256; 1407 char buffer[SIZE]; 1408 String8 result; 1409 1410 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1411 result.append(buffer); 1412 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1413 for (size_t i = 0; i < mTracks.size(); ++i) { 1414 sp<Track> track = mTracks[i]; 1415 if (track != 0) { 1416 track->dump(buffer, SIZE); 1417 result.append(buffer); 1418 } 1419 } 1420 1421 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1422 result.append(buffer); 1423 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1424 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1425 wp<Track> wTrack = mActiveTracks[i]; 1426 if (wTrack != 0) { 1427 sp<Track> track = wTrack.promote(); 1428 if (track != 0) { 1429 track->dump(buffer, SIZE); 1430 result.append(buffer); 1431 } 1432 } 1433 } 1434 write(fd, result.string(), result.size()); 1435 return NO_ERROR; 1436} 1437 1438status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1439{ 1440 const size_t SIZE = 256; 1441 char buffer[SIZE]; 1442 String8 result; 1443 1444 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1445 result.append(buffer); 1446 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1447 result.append(buffer); 1448 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1449 result.append(buffer); 1450 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1451 result.append(buffer); 1452 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1453 result.append(buffer); 1454 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1455 result.append(buffer); 1456 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1457 result.append(buffer); 1458 write(fd, result.string(), result.size()); 1459 1460 dumpBase(fd, args); 1461 1462 return NO_ERROR; 1463} 1464 1465// Thread virtuals 1466status_t AudioFlinger::PlaybackThread::readyToRun() 1467{ 1468 status_t status = initCheck(); 1469 if (status == NO_ERROR) { 1470 LOGI("AudioFlinger's thread %p ready to run", this); 1471 } else { 1472 LOGE("No working audio driver found."); 1473 } 1474 return status; 1475} 1476 1477void AudioFlinger::PlaybackThread::onFirstRef() 1478{ 1479 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1480} 1481 1482// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1483sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1484 const sp<AudioFlinger::Client>& client, 1485 int streamType, 1486 uint32_t sampleRate, 1487 uint32_t format, 1488 uint32_t channelMask, 1489 int frameCount, 1490 const sp<IMemory>& sharedBuffer, 1491 int sessionId, 1492 status_t *status) 1493{ 1494 sp<Track> track; 1495 status_t lStatus; 1496 1497 if (mType == DIRECT) { 1498 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1499 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1500 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1501 "for output %p with format %d", 1502 sampleRate, format, channelMask, mOutput, mFormat); 1503 lStatus = BAD_VALUE; 1504 goto Exit; 1505 } 1506 } 1507 } else { 1508 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1509 if (sampleRate > mSampleRate*2) { 1510 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1511 lStatus = BAD_VALUE; 1512 goto Exit; 1513 } 1514 } 1515 1516 lStatus = initCheck(); 1517 if (lStatus != NO_ERROR) { 1518 LOGE("Audio driver not initialized."); 1519 goto Exit; 1520 } 1521 1522 { // scope for mLock 1523 Mutex::Autolock _l(mLock); 1524 1525 // all tracks in same audio session must share the same routing strategy otherwise 1526 // conflicts will happen when tracks are moved from one output to another by audio policy 1527 // manager 1528 uint32_t strategy = 1529 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1530 for (size_t i = 0; i < mTracks.size(); ++i) { 1531 sp<Track> t = mTracks[i]; 1532 if (t != 0) { 1533 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1534 if (sessionId == t->sessionId() && strategy != actual) { 1535 LOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1536 strategy, actual); 1537 lStatus = BAD_VALUE; 1538 goto Exit; 1539 } 1540 } 1541 } 1542 1543 track = new Track(this, client, streamType, sampleRate, format, 1544 channelMask, frameCount, sharedBuffer, sessionId); 1545 if (track->getCblk() == NULL || track->name() < 0) { 1546 lStatus = NO_MEMORY; 1547 goto Exit; 1548 } 1549 mTracks.add(track); 1550 1551 sp<EffectChain> chain = getEffectChain_l(sessionId); 1552 if (chain != 0) { 1553 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1554 track->setMainBuffer(chain->inBuffer()); 1555 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1556 chain->incTrackCnt(); 1557 } 1558 1559 // invalidate track immediately if the stream type was moved to another thread since 1560 // createTrack() was called by the client process. 1561 if (!mStreamTypes[streamType].valid) { 1562 LOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1563 this, streamType); 1564 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1565 } 1566 } 1567 lStatus = NO_ERROR; 1568 1569Exit: 1570 if(status) { 1571 *status = lStatus; 1572 } 1573 return track; 1574} 1575 1576uint32_t AudioFlinger::PlaybackThread::latency() const 1577{ 1578 Mutex::Autolock _l(mLock); 1579 if (initCheck() == NO_ERROR) { 1580 return mOutput->stream->get_latency(mOutput->stream); 1581 } else { 1582 return 0; 1583 } 1584} 1585 1586status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1587{ 1588 mMasterVolume = value; 1589 return NO_ERROR; 1590} 1591 1592status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1593{ 1594 mMasterMute = muted; 1595 return NO_ERROR; 1596} 1597 1598float AudioFlinger::PlaybackThread::masterVolume() const 1599{ 1600 return mMasterVolume; 1601} 1602 1603bool AudioFlinger::PlaybackThread::masterMute() const 1604{ 1605 return mMasterMute; 1606} 1607 1608status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1609{ 1610 mStreamTypes[stream].volume = value; 1611 return NO_ERROR; 1612} 1613 1614status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1615{ 1616 mStreamTypes[stream].mute = muted; 1617 return NO_ERROR; 1618} 1619 1620float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1621{ 1622 return mStreamTypes[stream].volume; 1623} 1624 1625bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1626{ 1627 return mStreamTypes[stream].mute; 1628} 1629 1630// addTrack_l() must be called with ThreadBase::mLock held 1631status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1632{ 1633 status_t status = ALREADY_EXISTS; 1634 1635 // set retry count for buffer fill 1636 track->mRetryCount = kMaxTrackStartupRetries; 1637 if (mActiveTracks.indexOf(track) < 0) { 1638 // the track is newly added, make sure it fills up all its 1639 // buffers before playing. This is to ensure the client will 1640 // effectively get the latency it requested. 1641 track->mFillingUpStatus = Track::FS_FILLING; 1642 track->mResetDone = false; 1643 mActiveTracks.add(track); 1644 if (track->mainBuffer() != mMixBuffer) { 1645 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1646 if (chain != 0) { 1647 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1648 chain->incActiveTrackCnt(); 1649 } 1650 } 1651 1652 status = NO_ERROR; 1653 } 1654 1655 ALOGV("mWaitWorkCV.broadcast"); 1656 mWaitWorkCV.broadcast(); 1657 1658 return status; 1659} 1660 1661// destroyTrack_l() must be called with ThreadBase::mLock held 1662void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1663{ 1664 track->mState = TrackBase::TERMINATED; 1665 if (mActiveTracks.indexOf(track) < 0) { 1666 removeTrack_l(track); 1667 } 1668} 1669 1670void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1671{ 1672 mTracks.remove(track); 1673 deleteTrackName_l(track->name()); 1674 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1675 if (chain != 0) { 1676 chain->decTrackCnt(); 1677 } 1678} 1679 1680String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1681{ 1682 String8 out_s8 = String8(""); 1683 char *s; 1684 1685 Mutex::Autolock _l(mLock); 1686 if (initCheck() != NO_ERROR) { 1687 return out_s8; 1688 } 1689 1690 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1691 out_s8 = String8(s); 1692 free(s); 1693 return out_s8; 1694} 1695 1696// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1697void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1698 AudioSystem::OutputDescriptor desc; 1699 void *param2 = 0; 1700 1701 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1702 1703 switch (event) { 1704 case AudioSystem::OUTPUT_OPENED: 1705 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1706 desc.channels = mChannelMask; 1707 desc.samplingRate = mSampleRate; 1708 desc.format = mFormat; 1709 desc.frameCount = mFrameCount; 1710 desc.latency = latency(); 1711 param2 = &desc; 1712 break; 1713 1714 case AudioSystem::STREAM_CONFIG_CHANGED: 1715 param2 = ¶m; 1716 case AudioSystem::OUTPUT_CLOSED: 1717 default: 1718 break; 1719 } 1720 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1721} 1722 1723void AudioFlinger::PlaybackThread::readOutputParameters() 1724{ 1725 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1726 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1727 mChannelCount = (uint16_t)popcount(mChannelMask); 1728 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1729 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); 1730 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1731 1732 // FIXME - Current mixer implementation only supports stereo output: Always 1733 // Allocate a stereo buffer even if HW output is mono. 1734 if (mMixBuffer != NULL) delete[] mMixBuffer; 1735 mMixBuffer = new int16_t[mFrameCount * 2]; 1736 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1737 1738 // force reconfiguration of effect chains and engines to take new buffer size and audio 1739 // parameters into account 1740 // Note that mLock is not held when readOutputParameters() is called from the constructor 1741 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1742 // matter. 1743 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1744 Vector< sp<EffectChain> > effectChains = mEffectChains; 1745 for (size_t i = 0; i < effectChains.size(); i ++) { 1746 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1747 } 1748} 1749 1750status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1751{ 1752 if (halFrames == 0 || dspFrames == 0) { 1753 return BAD_VALUE; 1754 } 1755 Mutex::Autolock _l(mLock); 1756 if (initCheck() != NO_ERROR) { 1757 return INVALID_OPERATION; 1758 } 1759 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1760 1761 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1762} 1763 1764uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1765{ 1766 Mutex::Autolock _l(mLock); 1767 uint32_t result = 0; 1768 if (getEffectChain_l(sessionId) != 0) { 1769 result = EFFECT_SESSION; 1770 } 1771 1772 for (size_t i = 0; i < mTracks.size(); ++i) { 1773 sp<Track> track = mTracks[i]; 1774 if (sessionId == track->sessionId() && 1775 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1776 result |= TRACK_SESSION; 1777 break; 1778 } 1779 } 1780 1781 return result; 1782} 1783 1784uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1785{ 1786 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1787 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1788 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1789 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1790 } 1791 for (size_t i = 0; i < mTracks.size(); i++) { 1792 sp<Track> track = mTracks[i]; 1793 if (sessionId == track->sessionId() && 1794 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1795 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1796 } 1797 } 1798 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1799} 1800 1801 1802AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() 1803{ 1804 Mutex::Autolock _l(mLock); 1805 return mOutput; 1806} 1807 1808AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1809{ 1810 Mutex::Autolock _l(mLock); 1811 AudioStreamOut *output = mOutput; 1812 mOutput = NULL; 1813 return output; 1814} 1815 1816// this method must always be called either with ThreadBase mLock held or inside the thread loop 1817audio_stream_t* AudioFlinger::PlaybackThread::stream() 1818{ 1819 if (mOutput == NULL) { 1820 return NULL; 1821 } 1822 return &mOutput->stream->common; 1823} 1824 1825uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1826{ 1827 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1828 // decoding and transfer time. So sleeping for half of the latency would likely cause 1829 // underruns 1830 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1831 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1832 } else { 1833 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1834 } 1835} 1836 1837// ---------------------------------------------------------------------------- 1838 1839AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1840 : PlaybackThread(audioFlinger, output, id, device), 1841 mAudioMixer(0) 1842{ 1843 mType = ThreadBase::MIXER; 1844 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1845 1846 // FIXME - Current mixer implementation only supports stereo output 1847 if (mChannelCount == 1) { 1848 LOGE("Invalid audio hardware channel count"); 1849 } 1850} 1851 1852AudioFlinger::MixerThread::~MixerThread() 1853{ 1854 delete mAudioMixer; 1855} 1856 1857bool AudioFlinger::MixerThread::threadLoop() 1858{ 1859 Vector< sp<Track> > tracksToRemove; 1860 uint32_t mixerStatus = MIXER_IDLE; 1861 nsecs_t standbyTime = systemTime(); 1862 size_t mixBufferSize = mFrameCount * mFrameSize; 1863 // FIXME: Relaxed timing because of a certain device that can't meet latency 1864 // Should be reduced to 2x after the vendor fixes the driver issue 1865 // increase threshold again due to low power audio mode. The way this warning threshold is 1866 // calculated and its usefulness should be reconsidered anyway. 1867 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1868 nsecs_t lastWarning = 0; 1869 bool longStandbyExit = false; 1870 uint32_t activeSleepTime = activeSleepTimeUs(); 1871 uint32_t idleSleepTime = idleSleepTimeUs(); 1872 uint32_t sleepTime = idleSleepTime; 1873 uint32_t sleepTimeShift = 0; 1874 Vector< sp<EffectChain> > effectChains; 1875#ifdef DEBUG_CPU_USAGE 1876 ThreadCpuUsage cpu; 1877 const CentralTendencyStatistics& stats = cpu.statistics(); 1878#endif 1879 1880 acquireWakeLock(); 1881 1882 while (!exitPending()) 1883 { 1884#ifdef DEBUG_CPU_USAGE 1885 cpu.sampleAndEnable(); 1886 unsigned n = stats.n(); 1887 // cpu.elapsed() is expensive, so don't call it every loop 1888 if ((n & 127) == 1) { 1889 long long elapsed = cpu.elapsed(); 1890 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1891 double perLoop = elapsed / (double) n; 1892 double perLoop100 = perLoop * 0.01; 1893 double mean = stats.mean(); 1894 double stddev = stats.stddev(); 1895 double minimum = stats.minimum(); 1896 double maximum = stats.maximum(); 1897 cpu.resetStatistics(); 1898 LOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1899 elapsed * .000000001, n, perLoop * .000001, 1900 mean * .001, 1901 stddev * .001, 1902 minimum * .001, 1903 maximum * .001, 1904 mean / perLoop100, 1905 stddev / perLoop100, 1906 minimum / perLoop100, 1907 maximum / perLoop100); 1908 } 1909 } 1910#endif 1911 processConfigEvents(); 1912 1913 mixerStatus = MIXER_IDLE; 1914 { // scope for mLock 1915 1916 Mutex::Autolock _l(mLock); 1917 1918 if (checkForNewParameters_l()) { 1919 mixBufferSize = mFrameCount * mFrameSize; 1920 // FIXME: Relaxed timing because of a certain device that can't meet latency 1921 // Should be reduced to 2x after the vendor fixes the driver issue 1922 // increase threshold again due to low power audio mode. The way this warning 1923 // threshold is calculated and its usefulness should be reconsidered anyway. 1924 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1925 activeSleepTime = activeSleepTimeUs(); 1926 idleSleepTime = idleSleepTimeUs(); 1927 } 1928 1929 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1930 1931 // put audio hardware into standby after short delay 1932 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1933 mSuspended) { 1934 if (!mStandby) { 1935 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1936 mOutput->stream->common.standby(&mOutput->stream->common); 1937 mStandby = true; 1938 mBytesWritten = 0; 1939 } 1940 1941 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1942 // we're about to wait, flush the binder command buffer 1943 IPCThreadState::self()->flushCommands(); 1944 1945 if (exitPending()) break; 1946 1947 releaseWakeLock_l(); 1948 // wait until we have something to do... 1949 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1950 mWaitWorkCV.wait(mLock); 1951 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1952 acquireWakeLock_l(); 1953 1954 if (mMasterMute == false) { 1955 char value[PROPERTY_VALUE_MAX]; 1956 property_get("ro.audio.silent", value, "0"); 1957 if (atoi(value)) { 1958 LOGD("Silence is golden"); 1959 setMasterMute(true); 1960 } 1961 } 1962 1963 standbyTime = systemTime() + kStandbyTimeInNsecs; 1964 sleepTime = idleSleepTime; 1965 sleepTimeShift = 0; 1966 continue; 1967 } 1968 } 1969 1970 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1971 1972 // prevent any changes in effect chain list and in each effect chain 1973 // during mixing and effect process as the audio buffers could be deleted 1974 // or modified if an effect is created or deleted 1975 lockEffectChains_l(effectChains); 1976 } 1977 1978 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1979 // mix buffers... 1980 mAudioMixer->process(); 1981 sleepTime = 0; 1982 // increase sleep time progressively when application underrun condition clears 1983 if (sleepTimeShift > 0) { 1984 sleepTimeShift--; 1985 } 1986 standbyTime = systemTime() + kStandbyTimeInNsecs; 1987 //TODO: delay standby when effects have a tail 1988 } else { 1989 // If no tracks are ready, sleep once for the duration of an output 1990 // buffer size, then write 0s to the output 1991 if (sleepTime == 0) { 1992 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1993 sleepTime = activeSleepTime >> sleepTimeShift; 1994 if (sleepTime < kMinThreadSleepTimeUs) { 1995 sleepTime = kMinThreadSleepTimeUs; 1996 } 1997 // reduce sleep time in case of consecutive application underruns to avoid 1998 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 1999 // duration we would end up writing less data than needed by the audio HAL if 2000 // the condition persists. 2001 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2002 sleepTimeShift++; 2003 } 2004 } else { 2005 sleepTime = idleSleepTime; 2006 } 2007 } else if (mBytesWritten != 0 || 2008 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2009 memset (mMixBuffer, 0, mixBufferSize); 2010 sleepTime = 0; 2011 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2012 } 2013 // TODO add standby time extension fct of effect tail 2014 } 2015 2016 if (mSuspended) { 2017 sleepTime = suspendSleepTimeUs(); 2018 } 2019 // sleepTime == 0 means we must write to audio hardware 2020 if (sleepTime == 0) { 2021 for (size_t i = 0; i < effectChains.size(); i ++) { 2022 effectChains[i]->process_l(); 2023 } 2024 // enable changes in effect chain 2025 unlockEffectChains(effectChains); 2026 mLastWriteTime = systemTime(); 2027 mInWrite = true; 2028 mBytesWritten += mixBufferSize; 2029 2030 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2031 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2032 mNumWrites++; 2033 mInWrite = false; 2034 nsecs_t now = systemTime(); 2035 nsecs_t delta = now - mLastWriteTime; 2036 if (!mStandby && delta > maxPeriod) { 2037 mNumDelayedWrites++; 2038 if ((now - lastWarning) > kWarningThrottleNs) { 2039 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2040 ns2ms(delta), mNumDelayedWrites, this); 2041 lastWarning = now; 2042 } 2043 if (mStandby) { 2044 longStandbyExit = true; 2045 } 2046 } 2047 mStandby = false; 2048 } else { 2049 // enable changes in effect chain 2050 unlockEffectChains(effectChains); 2051 usleep(sleepTime); 2052 } 2053 2054 // finally let go of all our tracks, without the lock held 2055 // since we can't guarantee the destructors won't acquire that 2056 // same lock. 2057 tracksToRemove.clear(); 2058 2059 // Effect chains will be actually deleted here if they were removed from 2060 // mEffectChains list during mixing or effects processing 2061 effectChains.clear(); 2062 } 2063 2064 if (!mStandby) { 2065 mOutput->stream->common.standby(&mOutput->stream->common); 2066 } 2067 2068 releaseWakeLock(); 2069 2070 ALOGV("MixerThread %p exiting", this); 2071 return false; 2072} 2073 2074// prepareTracks_l() must be called with ThreadBase::mLock held 2075uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2076{ 2077 2078 uint32_t mixerStatus = MIXER_IDLE; 2079 // find out which tracks need to be processed 2080 size_t count = activeTracks.size(); 2081 size_t mixedTracks = 0; 2082 size_t tracksWithEffect = 0; 2083 2084 float masterVolume = mMasterVolume; 2085 bool masterMute = mMasterMute; 2086 2087 if (masterMute) { 2088 masterVolume = 0; 2089 } 2090 // Delegate master volume control to effect in output mix effect chain if needed 2091 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2092 if (chain != 0) { 2093 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2094 chain->setVolume_l(&v, &v); 2095 masterVolume = (float)((v + (1 << 23)) >> 24); 2096 chain.clear(); 2097 } 2098 2099 for (size_t i=0 ; i<count ; i++) { 2100 sp<Track> t = activeTracks[i].promote(); 2101 if (t == 0) continue; 2102 2103 Track* const track = t.get(); 2104 audio_track_cblk_t* cblk = track->cblk(); 2105 2106 // The first time a track is added we wait 2107 // for all its buffers to be filled before processing it 2108 mAudioMixer->setActiveTrack(track->name()); 2109 // make sure that we have enough frames to mix one full buffer. 2110 // enforce this condition only once to enable draining the buffer in case the client 2111 // app does not call stop() and relies on underrun to stop: 2112 // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed 2113 // during last round 2114 uint32_t minFrames = 1; 2115 if (!track->isStopped() && !track->isPausing() && 2116 (track->mRetryCount >= kMaxTrackRetries)) { 2117 if (t->sampleRate() == (int)mSampleRate) { 2118 minFrames = mFrameCount; 2119 } else { 2120 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1; 2121 } 2122 } 2123 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2124 !track->isPaused() && !track->isTerminated()) 2125 { 2126 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 2127 2128 mixedTracks++; 2129 2130 // track->mainBuffer() != mMixBuffer means there is an effect chain 2131 // connected to the track 2132 chain.clear(); 2133 if (track->mainBuffer() != mMixBuffer) { 2134 chain = getEffectChain_l(track->sessionId()); 2135 // Delegate volume control to effect in track effect chain if needed 2136 if (chain != 0) { 2137 tracksWithEffect++; 2138 } else { 2139 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 2140 track->name(), track->sessionId()); 2141 } 2142 } 2143 2144 2145 int param = AudioMixer::VOLUME; 2146 if (track->mFillingUpStatus == Track::FS_FILLED) { 2147 // no ramp for the first volume setting 2148 track->mFillingUpStatus = Track::FS_ACTIVE; 2149 if (track->mState == TrackBase::RESUMING) { 2150 track->mState = TrackBase::ACTIVE; 2151 param = AudioMixer::RAMP_VOLUME; 2152 } 2153 mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2154 } else if (cblk->server != 0) { 2155 // If the track is stopped before the first frame was mixed, 2156 // do not apply ramp 2157 param = AudioMixer::RAMP_VOLUME; 2158 } 2159 2160 // compute volume for this track 2161 uint32_t vl, vr, va; 2162 if (track->isMuted() || track->isPausing() || 2163 mStreamTypes[track->type()].mute) { 2164 vl = vr = va = 0; 2165 if (track->isPausing()) { 2166 track->setPaused(); 2167 } 2168 } else { 2169 2170 // read original volumes with volume control 2171 float typeVolume = mStreamTypes[track->type()].volume; 2172 float v = masterVolume * typeVolume; 2173 vl = (uint32_t)(v * cblk->volume[0]) << 12; 2174 vr = (uint32_t)(v * cblk->volume[1]) << 12; 2175 2176 va = (uint32_t)(v * cblk->sendLevel); 2177 } 2178 // Delegate volume control to effect in track effect chain if needed 2179 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2180 // Do not ramp volume if volume is controlled by effect 2181 param = AudioMixer::VOLUME; 2182 track->mHasVolumeController = true; 2183 } else { 2184 // force no volume ramp when volume controller was just disabled or removed 2185 // from effect chain to avoid volume spike 2186 if (track->mHasVolumeController) { 2187 param = AudioMixer::VOLUME; 2188 } 2189 track->mHasVolumeController = false; 2190 } 2191 2192 // Convert volumes from 8.24 to 4.12 format 2193 int16_t left, right, aux; 2194 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2195 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2196 left = int16_t(v_clamped); 2197 v_clamped = (vr + (1 << 11)) >> 12; 2198 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2199 right = int16_t(v_clamped); 2200 2201 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2202 aux = int16_t(va); 2203 2204 // XXX: these things DON'T need to be done each time 2205 mAudioMixer->setBufferProvider(track); 2206 mAudioMixer->enable(AudioMixer::MIXING); 2207 2208 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 2209 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 2210 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 2211 mAudioMixer->setParameter( 2212 AudioMixer::TRACK, 2213 AudioMixer::FORMAT, (void *)track->format()); 2214 mAudioMixer->setParameter( 2215 AudioMixer::TRACK, 2216 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2217 mAudioMixer->setParameter( 2218 AudioMixer::RESAMPLE, 2219 AudioMixer::SAMPLE_RATE, 2220 (void *)(cblk->sampleRate)); 2221 mAudioMixer->setParameter( 2222 AudioMixer::TRACK, 2223 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2224 mAudioMixer->setParameter( 2225 AudioMixer::TRACK, 2226 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2227 2228 // reset retry count 2229 track->mRetryCount = kMaxTrackRetries; 2230 mixerStatus = MIXER_TRACKS_READY; 2231 } else { 2232 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 2233 if (track->isStopped()) { 2234 track->reset(); 2235 } 2236 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2237 // We have consumed all the buffers of this track. 2238 // Remove it from the list of active tracks. 2239 tracksToRemove->add(track); 2240 } else { 2241 // No buffers for this track. Give it a few chances to 2242 // fill a buffer, then remove it from active list. 2243 if (--(track->mRetryCount) <= 0) { 2244 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 2245 tracksToRemove->add(track); 2246 // indicate to client process that the track was disabled because of underrun 2247 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2248 } else if (mixerStatus != MIXER_TRACKS_READY) { 2249 mixerStatus = MIXER_TRACKS_ENABLED; 2250 } 2251 } 2252 mAudioMixer->disable(AudioMixer::MIXING); 2253 } 2254 } 2255 2256 // remove all the tracks that need to be... 2257 count = tracksToRemove->size(); 2258 if (UNLIKELY(count)) { 2259 for (size_t i=0 ; i<count ; i++) { 2260 const sp<Track>& track = tracksToRemove->itemAt(i); 2261 mActiveTracks.remove(track); 2262 if (track->mainBuffer() != mMixBuffer) { 2263 chain = getEffectChain_l(track->sessionId()); 2264 if (chain != 0) { 2265 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2266 chain->decActiveTrackCnt(); 2267 } 2268 } 2269 if (track->isTerminated()) { 2270 removeTrack_l(track); 2271 } 2272 } 2273 } 2274 2275 // mix buffer must be cleared if all tracks are connected to an 2276 // effect chain as in this case the mixer will not write to 2277 // mix buffer and track effects will accumulate into it 2278 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2279 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2280 } 2281 2282 return mixerStatus; 2283} 2284 2285void AudioFlinger::MixerThread::invalidateTracks(int streamType) 2286{ 2287 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2288 this, streamType, mTracks.size()); 2289 Mutex::Autolock _l(mLock); 2290 2291 size_t size = mTracks.size(); 2292 for (size_t i = 0; i < size; i++) { 2293 sp<Track> t = mTracks[i]; 2294 if (t->type() == streamType) { 2295 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2296 t->mCblk->cv.signal(); 2297 } 2298 } 2299} 2300 2301void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid) 2302{ 2303 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2304 this, streamType, valid); 2305 Mutex::Autolock _l(mLock); 2306 2307 mStreamTypes[streamType].valid = valid; 2308} 2309 2310// getTrackName_l() must be called with ThreadBase::mLock held 2311int AudioFlinger::MixerThread::getTrackName_l() 2312{ 2313 return mAudioMixer->getTrackName(); 2314} 2315 2316// deleteTrackName_l() must be called with ThreadBase::mLock held 2317void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2318{ 2319 ALOGV("remove track (%d) and delete from mixer", name); 2320 mAudioMixer->deleteTrackName(name); 2321} 2322 2323// checkForNewParameters_l() must be called with ThreadBase::mLock held 2324bool AudioFlinger::MixerThread::checkForNewParameters_l() 2325{ 2326 bool reconfig = false; 2327 2328 while (!mNewParameters.isEmpty()) { 2329 status_t status = NO_ERROR; 2330 String8 keyValuePair = mNewParameters[0]; 2331 AudioParameter param = AudioParameter(keyValuePair); 2332 int value; 2333 2334 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2335 reconfig = true; 2336 } 2337 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2338 if (value != AUDIO_FORMAT_PCM_16_BIT) { 2339 status = BAD_VALUE; 2340 } else { 2341 reconfig = true; 2342 } 2343 } 2344 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2345 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2346 status = BAD_VALUE; 2347 } else { 2348 reconfig = true; 2349 } 2350 } 2351 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2352 // do not accept frame count changes if tracks are open as the track buffer 2353 // size depends on frame count and correct behavior would not be guaranteed 2354 // if frame count is changed after track creation 2355 if (!mTracks.isEmpty()) { 2356 status = INVALID_OPERATION; 2357 } else { 2358 reconfig = true; 2359 } 2360 } 2361 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2362 // when changing the audio output device, call addBatteryData to notify 2363 // the change 2364 if ((int)mDevice != value) { 2365 uint32_t params = 0; 2366 // check whether speaker is on 2367 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2368 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2369 } 2370 2371 int deviceWithoutSpeaker 2372 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2373 // check if any other device (except speaker) is on 2374 if (value & deviceWithoutSpeaker ) { 2375 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2376 } 2377 2378 if (params != 0) { 2379 addBatteryData(params); 2380 } 2381 } 2382 2383 // forward device change to effects that have requested to be 2384 // aware of attached audio device. 2385 mDevice = (uint32_t)value; 2386 for (size_t i = 0; i < mEffectChains.size(); i++) { 2387 mEffectChains[i]->setDevice_l(mDevice); 2388 } 2389 } 2390 2391 if (status == NO_ERROR) { 2392 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2393 keyValuePair.string()); 2394 if (!mStandby && status == INVALID_OPERATION) { 2395 mOutput->stream->common.standby(&mOutput->stream->common); 2396 mStandby = true; 2397 mBytesWritten = 0; 2398 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2399 keyValuePair.string()); 2400 } 2401 if (status == NO_ERROR && reconfig) { 2402 delete mAudioMixer; 2403 readOutputParameters(); 2404 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2405 for (size_t i = 0; i < mTracks.size() ; i++) { 2406 int name = getTrackName_l(); 2407 if (name < 0) break; 2408 mTracks[i]->mName = name; 2409 // limit track sample rate to 2 x new output sample rate 2410 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2411 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2412 } 2413 } 2414 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2415 } 2416 } 2417 2418 mNewParameters.removeAt(0); 2419 2420 mParamStatus = status; 2421 mParamCond.signal(); 2422 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2423 // already timed out waiting for the status and will never signal the condition. 2424 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2425 } 2426 return reconfig; 2427} 2428 2429status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2430{ 2431 const size_t SIZE = 256; 2432 char buffer[SIZE]; 2433 String8 result; 2434 2435 PlaybackThread::dumpInternals(fd, args); 2436 2437 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2438 result.append(buffer); 2439 write(fd, result.string(), result.size()); 2440 return NO_ERROR; 2441} 2442 2443uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2444{ 2445 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2446} 2447 2448uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2449{ 2450 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2451} 2452 2453// ---------------------------------------------------------------------------- 2454AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2455 : PlaybackThread(audioFlinger, output, id, device) 2456{ 2457 mType = ThreadBase::DIRECT; 2458} 2459 2460AudioFlinger::DirectOutputThread::~DirectOutputThread() 2461{ 2462} 2463 2464 2465static inline int16_t clamp16(int32_t sample) 2466{ 2467 if ((sample>>15) ^ (sample>>31)) 2468 sample = 0x7FFF ^ (sample>>31); 2469 return sample; 2470} 2471 2472static inline 2473int32_t mul(int16_t in, int16_t v) 2474{ 2475#if defined(__arm__) && !defined(__thumb__) 2476 int32_t out; 2477 asm( "smulbb %[out], %[in], %[v] \n" 2478 : [out]"=r"(out) 2479 : [in]"%r"(in), [v]"r"(v) 2480 : ); 2481 return out; 2482#else 2483 return in * int32_t(v); 2484#endif 2485} 2486 2487void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2488{ 2489 // Do not apply volume on compressed audio 2490 if (!audio_is_linear_pcm(mFormat)) { 2491 return; 2492 } 2493 2494 // convert to signed 16 bit before volume calculation 2495 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2496 size_t count = mFrameCount * mChannelCount; 2497 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2498 int16_t *dst = mMixBuffer + count-1; 2499 while(count--) { 2500 *dst-- = (int16_t)(*src--^0x80) << 8; 2501 } 2502 } 2503 2504 size_t frameCount = mFrameCount; 2505 int16_t *out = mMixBuffer; 2506 if (ramp) { 2507 if (mChannelCount == 1) { 2508 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2509 int32_t vlInc = d / (int32_t)frameCount; 2510 int32_t vl = ((int32_t)mLeftVolShort << 16); 2511 do { 2512 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2513 out++; 2514 vl += vlInc; 2515 } while (--frameCount); 2516 2517 } else { 2518 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2519 int32_t vlInc = d / (int32_t)frameCount; 2520 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2521 int32_t vrInc = d / (int32_t)frameCount; 2522 int32_t vl = ((int32_t)mLeftVolShort << 16); 2523 int32_t vr = ((int32_t)mRightVolShort << 16); 2524 do { 2525 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2526 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2527 out += 2; 2528 vl += vlInc; 2529 vr += vrInc; 2530 } while (--frameCount); 2531 } 2532 } else { 2533 if (mChannelCount == 1) { 2534 do { 2535 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2536 out++; 2537 } while (--frameCount); 2538 } else { 2539 do { 2540 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2541 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2542 out += 2; 2543 } while (--frameCount); 2544 } 2545 } 2546 2547 // convert back to unsigned 8 bit after volume calculation 2548 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2549 size_t count = mFrameCount * mChannelCount; 2550 int16_t *src = mMixBuffer; 2551 uint8_t *dst = (uint8_t *)mMixBuffer; 2552 while(count--) { 2553 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2554 } 2555 } 2556 2557 mLeftVolShort = leftVol; 2558 mRightVolShort = rightVol; 2559} 2560 2561bool AudioFlinger::DirectOutputThread::threadLoop() 2562{ 2563 uint32_t mixerStatus = MIXER_IDLE; 2564 sp<Track> trackToRemove; 2565 sp<Track> activeTrack; 2566 nsecs_t standbyTime = systemTime(); 2567 int8_t *curBuf; 2568 size_t mixBufferSize = mFrameCount*mFrameSize; 2569 uint32_t activeSleepTime = activeSleepTimeUs(); 2570 uint32_t idleSleepTime = idleSleepTimeUs(); 2571 uint32_t sleepTime = idleSleepTime; 2572 // use shorter standby delay as on normal output to release 2573 // hardware resources as soon as possible 2574 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2575 2576 acquireWakeLock(); 2577 2578 while (!exitPending()) 2579 { 2580 bool rampVolume; 2581 uint16_t leftVol; 2582 uint16_t rightVol; 2583 Vector< sp<EffectChain> > effectChains; 2584 2585 processConfigEvents(); 2586 2587 mixerStatus = MIXER_IDLE; 2588 2589 { // scope for the mLock 2590 2591 Mutex::Autolock _l(mLock); 2592 2593 if (checkForNewParameters_l()) { 2594 mixBufferSize = mFrameCount*mFrameSize; 2595 activeSleepTime = activeSleepTimeUs(); 2596 idleSleepTime = idleSleepTimeUs(); 2597 standbyDelay = microseconds(activeSleepTime*2); 2598 } 2599 2600 // put audio hardware into standby after short delay 2601 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2602 mSuspended) { 2603 // wait until we have something to do... 2604 if (!mStandby) { 2605 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2606 mOutput->stream->common.standby(&mOutput->stream->common); 2607 mStandby = true; 2608 mBytesWritten = 0; 2609 } 2610 2611 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2612 // we're about to wait, flush the binder command buffer 2613 IPCThreadState::self()->flushCommands(); 2614 2615 if (exitPending()) break; 2616 2617 releaseWakeLock_l(); 2618 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2619 mWaitWorkCV.wait(mLock); 2620 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2621 acquireWakeLock_l(); 2622 2623 if (mMasterMute == false) { 2624 char value[PROPERTY_VALUE_MAX]; 2625 property_get("ro.audio.silent", value, "0"); 2626 if (atoi(value)) { 2627 LOGD("Silence is golden"); 2628 setMasterMute(true); 2629 } 2630 } 2631 2632 standbyTime = systemTime() + standbyDelay; 2633 sleepTime = idleSleepTime; 2634 continue; 2635 } 2636 } 2637 2638 effectChains = mEffectChains; 2639 2640 // find out which tracks need to be processed 2641 if (mActiveTracks.size() != 0) { 2642 sp<Track> t = mActiveTracks[0].promote(); 2643 if (t == 0) continue; 2644 2645 Track* const track = t.get(); 2646 audio_track_cblk_t* cblk = track->cblk(); 2647 2648 // The first time a track is added we wait 2649 // for all its buffers to be filled before processing it 2650 if (cblk->framesReady() && track->isReady() && 2651 !track->isPaused() && !track->isTerminated()) 2652 { 2653 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2654 2655 if (track->mFillingUpStatus == Track::FS_FILLED) { 2656 track->mFillingUpStatus = Track::FS_ACTIVE; 2657 mLeftVolFloat = mRightVolFloat = 0; 2658 mLeftVolShort = mRightVolShort = 0; 2659 if (track->mState == TrackBase::RESUMING) { 2660 track->mState = TrackBase::ACTIVE; 2661 rampVolume = true; 2662 } 2663 } else if (cblk->server != 0) { 2664 // If the track is stopped before the first frame was mixed, 2665 // do not apply ramp 2666 rampVolume = true; 2667 } 2668 // compute volume for this track 2669 float left, right; 2670 if (track->isMuted() || mMasterMute || track->isPausing() || 2671 mStreamTypes[track->type()].mute) { 2672 left = right = 0; 2673 if (track->isPausing()) { 2674 track->setPaused(); 2675 } 2676 } else { 2677 float typeVolume = mStreamTypes[track->type()].volume; 2678 float v = mMasterVolume * typeVolume; 2679 float v_clamped = v * cblk->volume[0]; 2680 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2681 left = v_clamped/MAX_GAIN; 2682 v_clamped = v * cblk->volume[1]; 2683 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2684 right = v_clamped/MAX_GAIN; 2685 } 2686 2687 if (left != mLeftVolFloat || right != mRightVolFloat) { 2688 mLeftVolFloat = left; 2689 mRightVolFloat = right; 2690 2691 // If audio HAL implements volume control, 2692 // force software volume to nominal value 2693 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2694 left = 1.0f; 2695 right = 1.0f; 2696 } 2697 2698 // Convert volumes from float to 8.24 2699 uint32_t vl = (uint32_t)(left * (1 << 24)); 2700 uint32_t vr = (uint32_t)(right * (1 << 24)); 2701 2702 // Delegate volume control to effect in track effect chain if needed 2703 // only one effect chain can be present on DirectOutputThread, so if 2704 // there is one, the track is connected to it 2705 if (!effectChains.isEmpty()) { 2706 // Do not ramp volume if volume is controlled by effect 2707 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2708 rampVolume = false; 2709 } 2710 } 2711 2712 // Convert volumes from 8.24 to 4.12 format 2713 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2714 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2715 leftVol = (uint16_t)v_clamped; 2716 v_clamped = (vr + (1 << 11)) >> 12; 2717 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2718 rightVol = (uint16_t)v_clamped; 2719 } else { 2720 leftVol = mLeftVolShort; 2721 rightVol = mRightVolShort; 2722 rampVolume = false; 2723 } 2724 2725 // reset retry count 2726 track->mRetryCount = kMaxTrackRetriesDirect; 2727 activeTrack = t; 2728 mixerStatus = MIXER_TRACKS_READY; 2729 } else { 2730 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2731 if (track->isStopped()) { 2732 track->reset(); 2733 } 2734 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2735 // We have consumed all the buffers of this track. 2736 // Remove it from the list of active tracks. 2737 trackToRemove = track; 2738 } else { 2739 // No buffers for this track. Give it a few chances to 2740 // fill a buffer, then remove it from active list. 2741 if (--(track->mRetryCount) <= 0) { 2742 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2743 trackToRemove = track; 2744 } else { 2745 mixerStatus = MIXER_TRACKS_ENABLED; 2746 } 2747 } 2748 } 2749 } 2750 2751 // remove all the tracks that need to be... 2752 if (UNLIKELY(trackToRemove != 0)) { 2753 mActiveTracks.remove(trackToRemove); 2754 if (!effectChains.isEmpty()) { 2755 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2756 trackToRemove->sessionId()); 2757 effectChains[0]->decActiveTrackCnt(); 2758 } 2759 if (trackToRemove->isTerminated()) { 2760 removeTrack_l(trackToRemove); 2761 } 2762 } 2763 2764 lockEffectChains_l(effectChains); 2765 } 2766 2767 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2768 AudioBufferProvider::Buffer buffer; 2769 size_t frameCount = mFrameCount; 2770 curBuf = (int8_t *)mMixBuffer; 2771 // output audio to hardware 2772 while (frameCount) { 2773 buffer.frameCount = frameCount; 2774 activeTrack->getNextBuffer(&buffer); 2775 if (UNLIKELY(buffer.raw == 0)) { 2776 memset(curBuf, 0, frameCount * mFrameSize); 2777 break; 2778 } 2779 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2780 frameCount -= buffer.frameCount; 2781 curBuf += buffer.frameCount * mFrameSize; 2782 activeTrack->releaseBuffer(&buffer); 2783 } 2784 sleepTime = 0; 2785 standbyTime = systemTime() + standbyDelay; 2786 } else { 2787 if (sleepTime == 0) { 2788 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2789 sleepTime = activeSleepTime; 2790 } else { 2791 sleepTime = idleSleepTime; 2792 } 2793 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2794 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2795 sleepTime = 0; 2796 } 2797 } 2798 2799 if (mSuspended) { 2800 sleepTime = suspendSleepTimeUs(); 2801 } 2802 // sleepTime == 0 means we must write to audio hardware 2803 if (sleepTime == 0) { 2804 if (mixerStatus == MIXER_TRACKS_READY) { 2805 applyVolume(leftVol, rightVol, rampVolume); 2806 } 2807 for (size_t i = 0; i < effectChains.size(); i ++) { 2808 effectChains[i]->process_l(); 2809 } 2810 unlockEffectChains(effectChains); 2811 2812 mLastWriteTime = systemTime(); 2813 mInWrite = true; 2814 mBytesWritten += mixBufferSize; 2815 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2816 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2817 mNumWrites++; 2818 mInWrite = false; 2819 mStandby = false; 2820 } else { 2821 unlockEffectChains(effectChains); 2822 usleep(sleepTime); 2823 } 2824 2825 // finally let go of removed track, without the lock held 2826 // since we can't guarantee the destructors won't acquire that 2827 // same lock. 2828 trackToRemove.clear(); 2829 activeTrack.clear(); 2830 2831 // Effect chains will be actually deleted here if they were removed from 2832 // mEffectChains list during mixing or effects processing 2833 effectChains.clear(); 2834 } 2835 2836 if (!mStandby) { 2837 mOutput->stream->common.standby(&mOutput->stream->common); 2838 } 2839 2840 releaseWakeLock(); 2841 2842 ALOGV("DirectOutputThread %p exiting", this); 2843 return false; 2844} 2845 2846// getTrackName_l() must be called with ThreadBase::mLock held 2847int AudioFlinger::DirectOutputThread::getTrackName_l() 2848{ 2849 return 0; 2850} 2851 2852// deleteTrackName_l() must be called with ThreadBase::mLock held 2853void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2854{ 2855} 2856 2857// checkForNewParameters_l() must be called with ThreadBase::mLock held 2858bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2859{ 2860 bool reconfig = false; 2861 2862 while (!mNewParameters.isEmpty()) { 2863 status_t status = NO_ERROR; 2864 String8 keyValuePair = mNewParameters[0]; 2865 AudioParameter param = AudioParameter(keyValuePair); 2866 int value; 2867 2868 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2869 // do not accept frame count changes if tracks are open as the track buffer 2870 // size depends on frame count and correct behavior would not be garantied 2871 // if frame count is changed after track creation 2872 if (!mTracks.isEmpty()) { 2873 status = INVALID_OPERATION; 2874 } else { 2875 reconfig = true; 2876 } 2877 } 2878 if (status == NO_ERROR) { 2879 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2880 keyValuePair.string()); 2881 if (!mStandby && status == INVALID_OPERATION) { 2882 mOutput->stream->common.standby(&mOutput->stream->common); 2883 mStandby = true; 2884 mBytesWritten = 0; 2885 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2886 keyValuePair.string()); 2887 } 2888 if (status == NO_ERROR && reconfig) { 2889 readOutputParameters(); 2890 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2891 } 2892 } 2893 2894 mNewParameters.removeAt(0); 2895 2896 mParamStatus = status; 2897 mParamCond.signal(); 2898 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2899 // already timed out waiting for the status and will never signal the condition. 2900 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2901 } 2902 return reconfig; 2903} 2904 2905uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2906{ 2907 uint32_t time; 2908 if (audio_is_linear_pcm(mFormat)) { 2909 time = PlaybackThread::activeSleepTimeUs(); 2910 } else { 2911 time = 10000; 2912 } 2913 return time; 2914} 2915 2916uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2917{ 2918 uint32_t time; 2919 if (audio_is_linear_pcm(mFormat)) { 2920 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2921 } else { 2922 time = 10000; 2923 } 2924 return time; 2925} 2926 2927uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2928{ 2929 uint32_t time; 2930 if (audio_is_linear_pcm(mFormat)) { 2931 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2932 } else { 2933 time = 10000; 2934 } 2935 return time; 2936} 2937 2938 2939// ---------------------------------------------------------------------------- 2940 2941AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2942 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2943{ 2944 mType = ThreadBase::DUPLICATING; 2945 addOutputTrack(mainThread); 2946} 2947 2948AudioFlinger::DuplicatingThread::~DuplicatingThread() 2949{ 2950 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2951 mOutputTracks[i]->destroy(); 2952 } 2953 mOutputTracks.clear(); 2954} 2955 2956bool AudioFlinger::DuplicatingThread::threadLoop() 2957{ 2958 Vector< sp<Track> > tracksToRemove; 2959 uint32_t mixerStatus = MIXER_IDLE; 2960 nsecs_t standbyTime = systemTime(); 2961 size_t mixBufferSize = mFrameCount*mFrameSize; 2962 SortedVector< sp<OutputTrack> > outputTracks; 2963 uint32_t writeFrames = 0; 2964 uint32_t activeSleepTime = activeSleepTimeUs(); 2965 uint32_t idleSleepTime = idleSleepTimeUs(); 2966 uint32_t sleepTime = idleSleepTime; 2967 Vector< sp<EffectChain> > effectChains; 2968 2969 acquireWakeLock(); 2970 2971 while (!exitPending()) 2972 { 2973 processConfigEvents(); 2974 2975 mixerStatus = MIXER_IDLE; 2976 { // scope for the mLock 2977 2978 Mutex::Autolock _l(mLock); 2979 2980 if (checkForNewParameters_l()) { 2981 mixBufferSize = mFrameCount*mFrameSize; 2982 updateWaitTime(); 2983 activeSleepTime = activeSleepTimeUs(); 2984 idleSleepTime = idleSleepTimeUs(); 2985 } 2986 2987 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2988 2989 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2990 outputTracks.add(mOutputTracks[i]); 2991 } 2992 2993 // put audio hardware into standby after short delay 2994 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2995 mSuspended) { 2996 if (!mStandby) { 2997 for (size_t i = 0; i < outputTracks.size(); i++) { 2998 outputTracks[i]->stop(); 2999 } 3000 mStandby = true; 3001 mBytesWritten = 0; 3002 } 3003 3004 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3005 // we're about to wait, flush the binder command buffer 3006 IPCThreadState::self()->flushCommands(); 3007 outputTracks.clear(); 3008 3009 if (exitPending()) break; 3010 3011 releaseWakeLock_l(); 3012 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3013 mWaitWorkCV.wait(mLock); 3014 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3015 acquireWakeLock_l(); 3016 3017 if (mMasterMute == false) { 3018 char value[PROPERTY_VALUE_MAX]; 3019 property_get("ro.audio.silent", value, "0"); 3020 if (atoi(value)) { 3021 LOGD("Silence is golden"); 3022 setMasterMute(true); 3023 } 3024 } 3025 3026 standbyTime = systemTime() + kStandbyTimeInNsecs; 3027 sleepTime = idleSleepTime; 3028 continue; 3029 } 3030 } 3031 3032 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3033 3034 // prevent any changes in effect chain list and in each effect chain 3035 // during mixing and effect process as the audio buffers could be deleted 3036 // or modified if an effect is created or deleted 3037 lockEffectChains_l(effectChains); 3038 } 3039 3040 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3041 // mix buffers... 3042 if (outputsReady(outputTracks)) { 3043 mAudioMixer->process(); 3044 } else { 3045 memset(mMixBuffer, 0, mixBufferSize); 3046 } 3047 sleepTime = 0; 3048 writeFrames = mFrameCount; 3049 } else { 3050 if (sleepTime == 0) { 3051 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3052 sleepTime = activeSleepTime; 3053 } else { 3054 sleepTime = idleSleepTime; 3055 } 3056 } else if (mBytesWritten != 0) { 3057 // flush remaining overflow buffers in output tracks 3058 for (size_t i = 0; i < outputTracks.size(); i++) { 3059 if (outputTracks[i]->isActive()) { 3060 sleepTime = 0; 3061 writeFrames = 0; 3062 memset(mMixBuffer, 0, mixBufferSize); 3063 break; 3064 } 3065 } 3066 } 3067 } 3068 3069 if (mSuspended) { 3070 sleepTime = suspendSleepTimeUs(); 3071 } 3072 // sleepTime == 0 means we must write to audio hardware 3073 if (sleepTime == 0) { 3074 for (size_t i = 0; i < effectChains.size(); i ++) { 3075 effectChains[i]->process_l(); 3076 } 3077 // enable changes in effect chain 3078 unlockEffectChains(effectChains); 3079 3080 standbyTime = systemTime() + kStandbyTimeInNsecs; 3081 for (size_t i = 0; i < outputTracks.size(); i++) { 3082 outputTracks[i]->write(mMixBuffer, writeFrames); 3083 } 3084 mStandby = false; 3085 mBytesWritten += mixBufferSize; 3086 } else { 3087 // enable changes in effect chain 3088 unlockEffectChains(effectChains); 3089 usleep(sleepTime); 3090 } 3091 3092 // finally let go of all our tracks, without the lock held 3093 // since we can't guarantee the destructors won't acquire that 3094 // same lock. 3095 tracksToRemove.clear(); 3096 outputTracks.clear(); 3097 3098 // Effect chains will be actually deleted here if they were removed from 3099 // mEffectChains list during mixing or effects processing 3100 effectChains.clear(); 3101 } 3102 3103 releaseWakeLock(); 3104 3105 return false; 3106} 3107 3108void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3109{ 3110 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3111 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3112 this, 3113 mSampleRate, 3114 mFormat, 3115 mChannelMask, 3116 frameCount); 3117 if (outputTrack->cblk() != NULL) { 3118 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3119 mOutputTracks.add(outputTrack); 3120 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3121 updateWaitTime(); 3122 } 3123} 3124 3125void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3126{ 3127 Mutex::Autolock _l(mLock); 3128 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3129 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3130 mOutputTracks[i]->destroy(); 3131 mOutputTracks.removeAt(i); 3132 updateWaitTime(); 3133 return; 3134 } 3135 } 3136 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3137} 3138 3139void AudioFlinger::DuplicatingThread::updateWaitTime() 3140{ 3141 mWaitTimeMs = UINT_MAX; 3142 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3143 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3144 if (strong != NULL) { 3145 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3146 if (waitTimeMs < mWaitTimeMs) { 3147 mWaitTimeMs = waitTimeMs; 3148 } 3149 } 3150 } 3151} 3152 3153 3154bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3155{ 3156 for (size_t i = 0; i < outputTracks.size(); i++) { 3157 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3158 if (thread == 0) { 3159 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3160 return false; 3161 } 3162 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3163 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3164 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3165 return false; 3166 } 3167 } 3168 return true; 3169} 3170 3171uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3172{ 3173 return (mWaitTimeMs * 1000) / 2; 3174} 3175 3176// ---------------------------------------------------------------------------- 3177 3178// TrackBase constructor must be called with AudioFlinger::mLock held 3179AudioFlinger::ThreadBase::TrackBase::TrackBase( 3180 const wp<ThreadBase>& thread, 3181 const sp<Client>& client, 3182 uint32_t sampleRate, 3183 uint32_t format, 3184 uint32_t channelMask, 3185 int frameCount, 3186 uint32_t flags, 3187 const sp<IMemory>& sharedBuffer, 3188 int sessionId) 3189 : RefBase(), 3190 mThread(thread), 3191 mClient(client), 3192 mCblk(0), 3193 mFrameCount(0), 3194 mState(IDLE), 3195 mClientTid(-1), 3196 mFormat(format), 3197 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3198 mSessionId(sessionId) 3199{ 3200 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3201 3202 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3203 size_t size = sizeof(audio_track_cblk_t); 3204 uint8_t channelCount = popcount(channelMask); 3205 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3206 if (sharedBuffer == 0) { 3207 size += bufferSize; 3208 } 3209 3210 if (client != NULL) { 3211 mCblkMemory = client->heap()->allocate(size); 3212 if (mCblkMemory != 0) { 3213 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3214 if (mCblk) { // construct the shared structure in-place. 3215 new(mCblk) audio_track_cblk_t(); 3216 // clear all buffers 3217 mCblk->frameCount = frameCount; 3218 mCblk->sampleRate = sampleRate; 3219 mChannelCount = channelCount; 3220 mChannelMask = channelMask; 3221 if (sharedBuffer == 0) { 3222 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3223 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3224 // Force underrun condition to avoid false underrun callback until first data is 3225 // written to buffer (other flags are cleared) 3226 mCblk->flags = CBLK_UNDERRUN_ON; 3227 } else { 3228 mBuffer = sharedBuffer->pointer(); 3229 } 3230 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3231 } 3232 } else { 3233 LOGE("not enough memory for AudioTrack size=%u", size); 3234 client->heap()->dump("AudioTrack"); 3235 return; 3236 } 3237 } else { 3238 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3239 if (mCblk) { // construct the shared structure in-place. 3240 new(mCblk) audio_track_cblk_t(); 3241 // clear all buffers 3242 mCblk->frameCount = frameCount; 3243 mCblk->sampleRate = sampleRate; 3244 mChannelCount = channelCount; 3245 mChannelMask = channelMask; 3246 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3247 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3248 // Force underrun condition to avoid false underrun callback until first data is 3249 // written to buffer (other flags are cleared) 3250 mCblk->flags = CBLK_UNDERRUN_ON; 3251 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3252 } 3253 } 3254} 3255 3256AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3257{ 3258 if (mCblk) { 3259 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3260 if (mClient == NULL) { 3261 delete mCblk; 3262 } 3263 } 3264 mCblkMemory.clear(); // and free the shared memory 3265 if (mClient != NULL) { 3266 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3267 mClient.clear(); 3268 } 3269} 3270 3271void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3272{ 3273 buffer->raw = 0; 3274 mFrameCount = buffer->frameCount; 3275 step(); 3276 buffer->frameCount = 0; 3277} 3278 3279bool AudioFlinger::ThreadBase::TrackBase::step() { 3280 bool result; 3281 audio_track_cblk_t* cblk = this->cblk(); 3282 3283 result = cblk->stepServer(mFrameCount); 3284 if (!result) { 3285 ALOGV("stepServer failed acquiring cblk mutex"); 3286 mFlags |= STEPSERVER_FAILED; 3287 } 3288 return result; 3289} 3290 3291void AudioFlinger::ThreadBase::TrackBase::reset() { 3292 audio_track_cblk_t* cblk = this->cblk(); 3293 3294 cblk->user = 0; 3295 cblk->server = 0; 3296 cblk->userBase = 0; 3297 cblk->serverBase = 0; 3298 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3299 ALOGV("TrackBase::reset"); 3300} 3301 3302sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3303{ 3304 return mCblkMemory; 3305} 3306 3307int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3308 return (int)mCblk->sampleRate; 3309} 3310 3311int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3312 return (const int)mChannelCount; 3313} 3314 3315uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3316 return mChannelMask; 3317} 3318 3319void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3320 audio_track_cblk_t* cblk = this->cblk(); 3321 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 3322 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 3323 3324 // Check validity of returned pointer in case the track control block would have been corrupted. 3325 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3326 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 3327 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3328 server %d, serverBase %d, user %d, userBase %d", 3329 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3330 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3331 return 0; 3332 } 3333 3334 return bufferStart; 3335} 3336 3337// ---------------------------------------------------------------------------- 3338 3339// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3340AudioFlinger::PlaybackThread::Track::Track( 3341 const wp<ThreadBase>& thread, 3342 const sp<Client>& client, 3343 int streamType, 3344 uint32_t sampleRate, 3345 uint32_t format, 3346 uint32_t channelMask, 3347 int frameCount, 3348 const sp<IMemory>& sharedBuffer, 3349 int sessionId) 3350 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3351 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3352 mAuxEffectId(0), mHasVolumeController(false) 3353{ 3354 if (mCblk != NULL) { 3355 sp<ThreadBase> baseThread = thread.promote(); 3356 if (baseThread != 0) { 3357 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3358 mName = playbackThread->getTrackName_l(); 3359 mMainBuffer = playbackThread->mixBuffer(); 3360 } 3361 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3362 if (mName < 0) { 3363 LOGE("no more track names available"); 3364 } 3365 mVolume[0] = 1.0f; 3366 mVolume[1] = 1.0f; 3367 mStreamType = streamType; 3368 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3369 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3370 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3371 } 3372} 3373 3374AudioFlinger::PlaybackThread::Track::~Track() 3375{ 3376 ALOGV("PlaybackThread::Track destructor"); 3377 sp<ThreadBase> thread = mThread.promote(); 3378 if (thread != 0) { 3379 Mutex::Autolock _l(thread->mLock); 3380 mState = TERMINATED; 3381 } 3382} 3383 3384void AudioFlinger::PlaybackThread::Track::destroy() 3385{ 3386 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3387 // by removing it from mTracks vector, so there is a risk that this Tracks's 3388 // desctructor is called. As the destructor needs to lock mLock, 3389 // we must acquire a strong reference on this Track before locking mLock 3390 // here so that the destructor is called only when exiting this function. 3391 // On the other hand, as long as Track::destroy() is only called by 3392 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3393 // this Track with its member mTrack. 3394 sp<Track> keep(this); 3395 { // scope for mLock 3396 sp<ThreadBase> thread = mThread.promote(); 3397 if (thread != 0) { 3398 if (!isOutputTrack()) { 3399 if (mState == ACTIVE || mState == RESUMING) { 3400 AudioSystem::stopOutput(thread->id(), 3401 (audio_stream_type_t)mStreamType, 3402 mSessionId); 3403 3404 // to track the speaker usage 3405 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3406 } 3407 AudioSystem::releaseOutput(thread->id()); 3408 } 3409 Mutex::Autolock _l(thread->mLock); 3410 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3411 playbackThread->destroyTrack_l(this); 3412 } 3413 } 3414} 3415 3416void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3417{ 3418 snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 3419 mName - AudioMixer::TRACK0, 3420 (mClient == NULL) ? getpid() : mClient->pid(), 3421 mStreamType, 3422 mFormat, 3423 mChannelMask, 3424 mSessionId, 3425 mFrameCount, 3426 mState, 3427 mMute, 3428 mFillingUpStatus, 3429 mCblk->sampleRate, 3430 mCblk->volume[0], 3431 mCblk->volume[1], 3432 mCblk->server, 3433 mCblk->user, 3434 (int)mMainBuffer, 3435 (int)mAuxBuffer); 3436} 3437 3438status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3439{ 3440 audio_track_cblk_t* cblk = this->cblk(); 3441 uint32_t framesReady; 3442 uint32_t framesReq = buffer->frameCount; 3443 3444 // Check if last stepServer failed, try to step now 3445 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3446 if (!step()) goto getNextBuffer_exit; 3447 ALOGV("stepServer recovered"); 3448 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3449 } 3450 3451 framesReady = cblk->framesReady(); 3452 3453 if (LIKELY(framesReady)) { 3454 uint32_t s = cblk->server; 3455 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3456 3457 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3458 if (framesReq > framesReady) { 3459 framesReq = framesReady; 3460 } 3461 if (s + framesReq > bufferEnd) { 3462 framesReq = bufferEnd - s; 3463 } 3464 3465 buffer->raw = getBuffer(s, framesReq); 3466 if (buffer->raw == 0) goto getNextBuffer_exit; 3467 3468 buffer->frameCount = framesReq; 3469 return NO_ERROR; 3470 } 3471 3472getNextBuffer_exit: 3473 buffer->raw = 0; 3474 buffer->frameCount = 0; 3475 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3476 return NOT_ENOUGH_DATA; 3477} 3478 3479bool AudioFlinger::PlaybackThread::Track::isReady() const { 3480 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3481 3482 if (mCblk->framesReady() >= mCblk->frameCount || 3483 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3484 mFillingUpStatus = FS_FILLED; 3485 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3486 return true; 3487 } 3488 return false; 3489} 3490 3491status_t AudioFlinger::PlaybackThread::Track::start() 3492{ 3493 status_t status = NO_ERROR; 3494 ALOGV("start(%d), calling thread %d session %d", 3495 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3496 sp<ThreadBase> thread = mThread.promote(); 3497 if (thread != 0) { 3498 Mutex::Autolock _l(thread->mLock); 3499 int state = mState; 3500 // here the track could be either new, or restarted 3501 // in both cases "unstop" the track 3502 if (mState == PAUSED) { 3503 mState = TrackBase::RESUMING; 3504 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3505 } else { 3506 mState = TrackBase::ACTIVE; 3507 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3508 } 3509 3510 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3511 thread->mLock.unlock(); 3512 status = AudioSystem::startOutput(thread->id(), 3513 (audio_stream_type_t)mStreamType, 3514 mSessionId); 3515 thread->mLock.lock(); 3516 3517 // to track the speaker usage 3518 if (status == NO_ERROR) { 3519 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3520 } 3521 } 3522 if (status == NO_ERROR) { 3523 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3524 playbackThread->addTrack_l(this); 3525 } else { 3526 mState = state; 3527 } 3528 } else { 3529 status = BAD_VALUE; 3530 } 3531 return status; 3532} 3533 3534void AudioFlinger::PlaybackThread::Track::stop() 3535{ 3536 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3537 sp<ThreadBase> thread = mThread.promote(); 3538 if (thread != 0) { 3539 Mutex::Autolock _l(thread->mLock); 3540 int state = mState; 3541 if (mState > STOPPED) { 3542 mState = STOPPED; 3543 // If the track is not active (PAUSED and buffers full), flush buffers 3544 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3545 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3546 reset(); 3547 } 3548 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3549 } 3550 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3551 thread->mLock.unlock(); 3552 AudioSystem::stopOutput(thread->id(), 3553 (audio_stream_type_t)mStreamType, 3554 mSessionId); 3555 thread->mLock.lock(); 3556 3557 // to track the speaker usage 3558 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3559 } 3560 } 3561} 3562 3563void AudioFlinger::PlaybackThread::Track::pause() 3564{ 3565 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3566 sp<ThreadBase> thread = mThread.promote(); 3567 if (thread != 0) { 3568 Mutex::Autolock _l(thread->mLock); 3569 if (mState == ACTIVE || mState == RESUMING) { 3570 mState = PAUSING; 3571 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3572 if (!isOutputTrack()) { 3573 thread->mLock.unlock(); 3574 AudioSystem::stopOutput(thread->id(), 3575 (audio_stream_type_t)mStreamType, 3576 mSessionId); 3577 thread->mLock.lock(); 3578 3579 // to track the speaker usage 3580 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3581 } 3582 } 3583 } 3584} 3585 3586void AudioFlinger::PlaybackThread::Track::flush() 3587{ 3588 ALOGV("flush(%d)", mName); 3589 sp<ThreadBase> thread = mThread.promote(); 3590 if (thread != 0) { 3591 Mutex::Autolock _l(thread->mLock); 3592 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3593 return; 3594 } 3595 // No point remaining in PAUSED state after a flush => go to 3596 // STOPPED state 3597 mState = STOPPED; 3598 3599 // do not reset the track if it is still in the process of being stopped or paused. 3600 // this will be done by prepareTracks_l() when the track is stopped. 3601 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3602 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3603 reset(); 3604 } 3605 } 3606} 3607 3608void AudioFlinger::PlaybackThread::Track::reset() 3609{ 3610 // Do not reset twice to avoid discarding data written just after a flush and before 3611 // the audioflinger thread detects the track is stopped. 3612 if (!mResetDone) { 3613 TrackBase::reset(); 3614 // Force underrun condition to avoid false underrun callback until first data is 3615 // written to buffer 3616 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3617 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3618 mFillingUpStatus = FS_FILLING; 3619 mResetDone = true; 3620 } 3621} 3622 3623void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3624{ 3625 mMute = muted; 3626} 3627 3628void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3629{ 3630 mVolume[0] = left; 3631 mVolume[1] = right; 3632} 3633 3634status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3635{ 3636 status_t status = DEAD_OBJECT; 3637 sp<ThreadBase> thread = mThread.promote(); 3638 if (thread != 0) { 3639 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3640 status = playbackThread->attachAuxEffect(this, EffectId); 3641 } 3642 return status; 3643} 3644 3645void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3646{ 3647 mAuxEffectId = EffectId; 3648 mAuxBuffer = buffer; 3649} 3650 3651// ---------------------------------------------------------------------------- 3652 3653// RecordTrack constructor must be called with AudioFlinger::mLock held 3654AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3655 const wp<ThreadBase>& thread, 3656 const sp<Client>& client, 3657 uint32_t sampleRate, 3658 uint32_t format, 3659 uint32_t channelMask, 3660 int frameCount, 3661 uint32_t flags, 3662 int sessionId) 3663 : TrackBase(thread, client, sampleRate, format, 3664 channelMask, frameCount, flags, 0, sessionId), 3665 mOverflow(false) 3666{ 3667 if (mCblk != NULL) { 3668 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3669 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3670 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3671 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3672 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3673 } else { 3674 mCblk->frameSize = sizeof(int8_t); 3675 } 3676 } 3677} 3678 3679AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3680{ 3681 sp<ThreadBase> thread = mThread.promote(); 3682 if (thread != 0) { 3683 AudioSystem::releaseInput(thread->id()); 3684 } 3685} 3686 3687status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3688{ 3689 audio_track_cblk_t* cblk = this->cblk(); 3690 uint32_t framesAvail; 3691 uint32_t framesReq = buffer->frameCount; 3692 3693 // Check if last stepServer failed, try to step now 3694 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3695 if (!step()) goto getNextBuffer_exit; 3696 ALOGV("stepServer recovered"); 3697 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3698 } 3699 3700 framesAvail = cblk->framesAvailable_l(); 3701 3702 if (LIKELY(framesAvail)) { 3703 uint32_t s = cblk->server; 3704 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3705 3706 if (framesReq > framesAvail) { 3707 framesReq = framesAvail; 3708 } 3709 if (s + framesReq > bufferEnd) { 3710 framesReq = bufferEnd - s; 3711 } 3712 3713 buffer->raw = getBuffer(s, framesReq); 3714 if (buffer->raw == 0) goto getNextBuffer_exit; 3715 3716 buffer->frameCount = framesReq; 3717 return NO_ERROR; 3718 } 3719 3720getNextBuffer_exit: 3721 buffer->raw = 0; 3722 buffer->frameCount = 0; 3723 return NOT_ENOUGH_DATA; 3724} 3725 3726status_t AudioFlinger::RecordThread::RecordTrack::start() 3727{ 3728 sp<ThreadBase> thread = mThread.promote(); 3729 if (thread != 0) { 3730 RecordThread *recordThread = (RecordThread *)thread.get(); 3731 return recordThread->start(this); 3732 } else { 3733 return BAD_VALUE; 3734 } 3735} 3736 3737void AudioFlinger::RecordThread::RecordTrack::stop() 3738{ 3739 sp<ThreadBase> thread = mThread.promote(); 3740 if (thread != 0) { 3741 RecordThread *recordThread = (RecordThread *)thread.get(); 3742 recordThread->stop(this); 3743 TrackBase::reset(); 3744 // Force overerrun condition to avoid false overrun callback until first data is 3745 // read from buffer 3746 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3747 } 3748} 3749 3750void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3751{ 3752 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3753 (mClient == NULL) ? getpid() : mClient->pid(), 3754 mFormat, 3755 mChannelMask, 3756 mSessionId, 3757 mFrameCount, 3758 mState, 3759 mCblk->sampleRate, 3760 mCblk->server, 3761 mCblk->user); 3762} 3763 3764 3765// ---------------------------------------------------------------------------- 3766 3767AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3768 const wp<ThreadBase>& thread, 3769 DuplicatingThread *sourceThread, 3770 uint32_t sampleRate, 3771 uint32_t format, 3772 uint32_t channelMask, 3773 int frameCount) 3774 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3775 mActive(false), mSourceThread(sourceThread) 3776{ 3777 3778 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3779 if (mCblk != NULL) { 3780 mCblk->flags |= CBLK_DIRECTION_OUT; 3781 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3782 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3783 mOutBuffer.frameCount = 0; 3784 playbackThread->mTracks.add(this); 3785 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3786 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3787 mCblk, mBuffer, mCblk->buffers, 3788 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3789 } else { 3790 LOGW("Error creating output track on thread %p", playbackThread); 3791 } 3792} 3793 3794AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3795{ 3796 clearBufferQueue(); 3797} 3798 3799status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3800{ 3801 status_t status = Track::start(); 3802 if (status != NO_ERROR) { 3803 return status; 3804 } 3805 3806 mActive = true; 3807 mRetryCount = 127; 3808 return status; 3809} 3810 3811void AudioFlinger::PlaybackThread::OutputTrack::stop() 3812{ 3813 Track::stop(); 3814 clearBufferQueue(); 3815 mOutBuffer.frameCount = 0; 3816 mActive = false; 3817} 3818 3819bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3820{ 3821 Buffer *pInBuffer; 3822 Buffer inBuffer; 3823 uint32_t channelCount = mChannelCount; 3824 bool outputBufferFull = false; 3825 inBuffer.frameCount = frames; 3826 inBuffer.i16 = data; 3827 3828 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3829 3830 if (!mActive && frames != 0) { 3831 start(); 3832 sp<ThreadBase> thread = mThread.promote(); 3833 if (thread != 0) { 3834 MixerThread *mixerThread = (MixerThread *)thread.get(); 3835 if (mCblk->frameCount > frames){ 3836 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3837 uint32_t startFrames = (mCblk->frameCount - frames); 3838 pInBuffer = new Buffer; 3839 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3840 pInBuffer->frameCount = startFrames; 3841 pInBuffer->i16 = pInBuffer->mBuffer; 3842 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3843 mBufferQueue.add(pInBuffer); 3844 } else { 3845 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3846 } 3847 } 3848 } 3849 } 3850 3851 while (waitTimeLeftMs) { 3852 // First write pending buffers, then new data 3853 if (mBufferQueue.size()) { 3854 pInBuffer = mBufferQueue.itemAt(0); 3855 } else { 3856 pInBuffer = &inBuffer; 3857 } 3858 3859 if (pInBuffer->frameCount == 0) { 3860 break; 3861 } 3862 3863 if (mOutBuffer.frameCount == 0) { 3864 mOutBuffer.frameCount = pInBuffer->frameCount; 3865 nsecs_t startTime = systemTime(); 3866 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3867 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3868 outputBufferFull = true; 3869 break; 3870 } 3871 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3872 if (waitTimeLeftMs >= waitTimeMs) { 3873 waitTimeLeftMs -= waitTimeMs; 3874 } else { 3875 waitTimeLeftMs = 0; 3876 } 3877 } 3878 3879 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3880 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3881 mCblk->stepUser(outFrames); 3882 pInBuffer->frameCount -= outFrames; 3883 pInBuffer->i16 += outFrames * channelCount; 3884 mOutBuffer.frameCount -= outFrames; 3885 mOutBuffer.i16 += outFrames * channelCount; 3886 3887 if (pInBuffer->frameCount == 0) { 3888 if (mBufferQueue.size()) { 3889 mBufferQueue.removeAt(0); 3890 delete [] pInBuffer->mBuffer; 3891 delete pInBuffer; 3892 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3893 } else { 3894 break; 3895 } 3896 } 3897 } 3898 3899 // If we could not write all frames, allocate a buffer and queue it for next time. 3900 if (inBuffer.frameCount) { 3901 sp<ThreadBase> thread = mThread.promote(); 3902 if (thread != 0 && !thread->standby()) { 3903 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3904 pInBuffer = new Buffer; 3905 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3906 pInBuffer->frameCount = inBuffer.frameCount; 3907 pInBuffer->i16 = pInBuffer->mBuffer; 3908 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3909 mBufferQueue.add(pInBuffer); 3910 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3911 } else { 3912 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3913 } 3914 } 3915 } 3916 3917 // Calling write() with a 0 length buffer, means that no more data will be written: 3918 // If no more buffers are pending, fill output track buffer to make sure it is started 3919 // by output mixer. 3920 if (frames == 0 && mBufferQueue.size() == 0) { 3921 if (mCblk->user < mCblk->frameCount) { 3922 frames = mCblk->frameCount - mCblk->user; 3923 pInBuffer = new Buffer; 3924 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3925 pInBuffer->frameCount = frames; 3926 pInBuffer->i16 = pInBuffer->mBuffer; 3927 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3928 mBufferQueue.add(pInBuffer); 3929 } else if (mActive) { 3930 stop(); 3931 } 3932 } 3933 3934 return outputBufferFull; 3935} 3936 3937status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3938{ 3939 int active; 3940 status_t result; 3941 audio_track_cblk_t* cblk = mCblk; 3942 uint32_t framesReq = buffer->frameCount; 3943 3944// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3945 buffer->frameCount = 0; 3946 3947 uint32_t framesAvail = cblk->framesAvailable(); 3948 3949 3950 if (framesAvail == 0) { 3951 Mutex::Autolock _l(cblk->lock); 3952 goto start_loop_here; 3953 while (framesAvail == 0) { 3954 active = mActive; 3955 if (UNLIKELY(!active)) { 3956 ALOGV("Not active and NO_MORE_BUFFERS"); 3957 return AudioTrack::NO_MORE_BUFFERS; 3958 } 3959 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3960 if (result != NO_ERROR) { 3961 return AudioTrack::NO_MORE_BUFFERS; 3962 } 3963 // read the server count again 3964 start_loop_here: 3965 framesAvail = cblk->framesAvailable_l(); 3966 } 3967 } 3968 3969// if (framesAvail < framesReq) { 3970// return AudioTrack::NO_MORE_BUFFERS; 3971// } 3972 3973 if (framesReq > framesAvail) { 3974 framesReq = framesAvail; 3975 } 3976 3977 uint32_t u = cblk->user; 3978 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3979 3980 if (u + framesReq > bufferEnd) { 3981 framesReq = bufferEnd - u; 3982 } 3983 3984 buffer->frameCount = framesReq; 3985 buffer->raw = (void *)cblk->buffer(u); 3986 return NO_ERROR; 3987} 3988 3989 3990void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3991{ 3992 size_t size = mBufferQueue.size(); 3993 Buffer *pBuffer; 3994 3995 for (size_t i = 0; i < size; i++) { 3996 pBuffer = mBufferQueue.itemAt(i); 3997 delete [] pBuffer->mBuffer; 3998 delete pBuffer; 3999 } 4000 mBufferQueue.clear(); 4001} 4002 4003// ---------------------------------------------------------------------------- 4004 4005AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4006 : RefBase(), 4007 mAudioFlinger(audioFlinger), 4008 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4009 mPid(pid) 4010{ 4011 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4012} 4013 4014// Client destructor must be called with AudioFlinger::mLock held 4015AudioFlinger::Client::~Client() 4016{ 4017 mAudioFlinger->removeClient_l(mPid); 4018} 4019 4020const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4021{ 4022 return mMemoryDealer; 4023} 4024 4025// ---------------------------------------------------------------------------- 4026 4027AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4028 const sp<IAudioFlingerClient>& client, 4029 pid_t pid) 4030 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4031{ 4032} 4033 4034AudioFlinger::NotificationClient::~NotificationClient() 4035{ 4036 mClient.clear(); 4037} 4038 4039void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4040{ 4041 sp<NotificationClient> keep(this); 4042 { 4043 mAudioFlinger->removeNotificationClient(mPid); 4044 } 4045} 4046 4047// ---------------------------------------------------------------------------- 4048 4049AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4050 : BnAudioTrack(), 4051 mTrack(track) 4052{ 4053} 4054 4055AudioFlinger::TrackHandle::~TrackHandle() { 4056 // just stop the track on deletion, associated resources 4057 // will be freed from the main thread once all pending buffers have 4058 // been played. Unless it's not in the active track list, in which 4059 // case we free everything now... 4060 mTrack->destroy(); 4061} 4062 4063status_t AudioFlinger::TrackHandle::start() { 4064 return mTrack->start(); 4065} 4066 4067void AudioFlinger::TrackHandle::stop() { 4068 mTrack->stop(); 4069} 4070 4071void AudioFlinger::TrackHandle::flush() { 4072 mTrack->flush(); 4073} 4074 4075void AudioFlinger::TrackHandle::mute(bool e) { 4076 mTrack->mute(e); 4077} 4078 4079void AudioFlinger::TrackHandle::pause() { 4080 mTrack->pause(); 4081} 4082 4083void AudioFlinger::TrackHandle::setVolume(float left, float right) { 4084 mTrack->setVolume(left, right); 4085} 4086 4087sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4088 return mTrack->getCblk(); 4089} 4090 4091status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4092{ 4093 return mTrack->attachAuxEffect(EffectId); 4094} 4095 4096status_t AudioFlinger::TrackHandle::onTransact( 4097 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4098{ 4099 return BnAudioTrack::onTransact(code, data, reply, flags); 4100} 4101 4102// ---------------------------------------------------------------------------- 4103 4104sp<IAudioRecord> AudioFlinger::openRecord( 4105 pid_t pid, 4106 int input, 4107 uint32_t sampleRate, 4108 uint32_t format, 4109 uint32_t channelMask, 4110 int frameCount, 4111 uint32_t flags, 4112 int *sessionId, 4113 status_t *status) 4114{ 4115 sp<RecordThread::RecordTrack> recordTrack; 4116 sp<RecordHandle> recordHandle; 4117 sp<Client> client; 4118 wp<Client> wclient; 4119 status_t lStatus; 4120 RecordThread *thread; 4121 size_t inFrameCount; 4122 int lSessionId; 4123 4124 // check calling permissions 4125 if (!recordingAllowed()) { 4126 lStatus = PERMISSION_DENIED; 4127 goto Exit; 4128 } 4129 4130 // add client to list 4131 { // scope for mLock 4132 Mutex::Autolock _l(mLock); 4133 thread = checkRecordThread_l(input); 4134 if (thread == NULL) { 4135 lStatus = BAD_VALUE; 4136 goto Exit; 4137 } 4138 4139 wclient = mClients.valueFor(pid); 4140 if (wclient != NULL) { 4141 client = wclient.promote(); 4142 } else { 4143 client = new Client(this, pid); 4144 mClients.add(pid, client); 4145 } 4146 4147 // If no audio session id is provided, create one here 4148 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4149 lSessionId = *sessionId; 4150 } else { 4151 lSessionId = nextUniqueId(); 4152 if (sessionId != NULL) { 4153 *sessionId = lSessionId; 4154 } 4155 } 4156 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4157 recordTrack = thread->createRecordTrack_l(client, 4158 sampleRate, 4159 format, 4160 channelMask, 4161 frameCount, 4162 flags, 4163 lSessionId, 4164 &lStatus); 4165 } 4166 if (lStatus != NO_ERROR) { 4167 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4168 // destructor is called by the TrackBase destructor with mLock held 4169 client.clear(); 4170 recordTrack.clear(); 4171 goto Exit; 4172 } 4173 4174 // return to handle to client 4175 recordHandle = new RecordHandle(recordTrack); 4176 lStatus = NO_ERROR; 4177 4178Exit: 4179 if (status) { 4180 *status = lStatus; 4181 } 4182 return recordHandle; 4183} 4184 4185// ---------------------------------------------------------------------------- 4186 4187AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4188 : BnAudioRecord(), 4189 mRecordTrack(recordTrack) 4190{ 4191} 4192 4193AudioFlinger::RecordHandle::~RecordHandle() { 4194 stop(); 4195} 4196 4197status_t AudioFlinger::RecordHandle::start() { 4198 ALOGV("RecordHandle::start()"); 4199 return mRecordTrack->start(); 4200} 4201 4202void AudioFlinger::RecordHandle::stop() { 4203 ALOGV("RecordHandle::stop()"); 4204 mRecordTrack->stop(); 4205} 4206 4207sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4208 return mRecordTrack->getCblk(); 4209} 4210 4211status_t AudioFlinger::RecordHandle::onTransact( 4212 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4213{ 4214 return BnAudioRecord::onTransact(code, data, reply, flags); 4215} 4216 4217// ---------------------------------------------------------------------------- 4218 4219AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4220 AudioStreamIn *input, 4221 uint32_t sampleRate, 4222 uint32_t channels, 4223 int id, 4224 uint32_t device) : 4225 ThreadBase(audioFlinger, id, device), 4226 mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 4227{ 4228 mType = ThreadBase::RECORD; 4229 4230 snprintf(mName, kNameLength, "AudioIn_%d", id); 4231 4232 mReqChannelCount = popcount(channels); 4233 mReqSampleRate = sampleRate; 4234 readInputParameters(); 4235} 4236 4237 4238AudioFlinger::RecordThread::~RecordThread() 4239{ 4240 delete[] mRsmpInBuffer; 4241 if (mResampler != 0) { 4242 delete mResampler; 4243 delete[] mRsmpOutBuffer; 4244 } 4245} 4246 4247void AudioFlinger::RecordThread::onFirstRef() 4248{ 4249 run(mName, PRIORITY_URGENT_AUDIO); 4250} 4251 4252status_t AudioFlinger::RecordThread::readyToRun() 4253{ 4254 status_t status = initCheck(); 4255 LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4256 return status; 4257} 4258 4259bool AudioFlinger::RecordThread::threadLoop() 4260{ 4261 AudioBufferProvider::Buffer buffer; 4262 sp<RecordTrack> activeTrack; 4263 Vector< sp<EffectChain> > effectChains; 4264 4265 nsecs_t lastWarning = 0; 4266 4267 acquireWakeLock(); 4268 4269 // start recording 4270 while (!exitPending()) { 4271 4272 processConfigEvents(); 4273 4274 { // scope for mLock 4275 Mutex::Autolock _l(mLock); 4276 checkForNewParameters_l(); 4277 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4278 if (!mStandby) { 4279 mInput->stream->common.standby(&mInput->stream->common); 4280 mStandby = true; 4281 } 4282 4283 if (exitPending()) break; 4284 4285 releaseWakeLock_l(); 4286 ALOGV("RecordThread: loop stopping"); 4287 // go to sleep 4288 mWaitWorkCV.wait(mLock); 4289 ALOGV("RecordThread: loop starting"); 4290 acquireWakeLock_l(); 4291 continue; 4292 } 4293 if (mActiveTrack != 0) { 4294 if (mActiveTrack->mState == TrackBase::PAUSING) { 4295 if (!mStandby) { 4296 mInput->stream->common.standby(&mInput->stream->common); 4297 mStandby = true; 4298 } 4299 mActiveTrack.clear(); 4300 mStartStopCond.broadcast(); 4301 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4302 if (mReqChannelCount != mActiveTrack->channelCount()) { 4303 mActiveTrack.clear(); 4304 mStartStopCond.broadcast(); 4305 } else if (mBytesRead != 0) { 4306 // record start succeeds only if first read from audio input 4307 // succeeds 4308 if (mBytesRead > 0) { 4309 mActiveTrack->mState = TrackBase::ACTIVE; 4310 } else { 4311 mActiveTrack.clear(); 4312 } 4313 mStartStopCond.broadcast(); 4314 } 4315 mStandby = false; 4316 } 4317 } 4318 lockEffectChains_l(effectChains); 4319 } 4320 4321 if (mActiveTrack != 0) { 4322 if (mActiveTrack->mState != TrackBase::ACTIVE && 4323 mActiveTrack->mState != TrackBase::RESUMING) { 4324 unlockEffectChains(effectChains); 4325 usleep(kRecordThreadSleepUs); 4326 continue; 4327 } 4328 for (size_t i = 0; i < effectChains.size(); i ++) { 4329 effectChains[i]->process_l(); 4330 } 4331 4332 buffer.frameCount = mFrameCount; 4333 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4334 size_t framesOut = buffer.frameCount; 4335 if (mResampler == 0) { 4336 // no resampling 4337 while (framesOut) { 4338 size_t framesIn = mFrameCount - mRsmpInIndex; 4339 if (framesIn) { 4340 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4341 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4342 if (framesIn > framesOut) 4343 framesIn = framesOut; 4344 mRsmpInIndex += framesIn; 4345 framesOut -= framesIn; 4346 if ((int)mChannelCount == mReqChannelCount || 4347 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4348 memcpy(dst, src, framesIn * mFrameSize); 4349 } else { 4350 int16_t *src16 = (int16_t *)src; 4351 int16_t *dst16 = (int16_t *)dst; 4352 if (mChannelCount == 1) { 4353 while (framesIn--) { 4354 *dst16++ = *src16; 4355 *dst16++ = *src16++; 4356 } 4357 } else { 4358 while (framesIn--) { 4359 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4360 src16 += 2; 4361 } 4362 } 4363 } 4364 } 4365 if (framesOut && mFrameCount == mRsmpInIndex) { 4366 if (framesOut == mFrameCount && 4367 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4368 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4369 framesOut = 0; 4370 } else { 4371 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4372 mRsmpInIndex = 0; 4373 } 4374 if (mBytesRead < 0) { 4375 LOGE("Error reading audio input"); 4376 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4377 // Force input into standby so that it tries to 4378 // recover at next read attempt 4379 mInput->stream->common.standby(&mInput->stream->common); 4380 usleep(kRecordThreadSleepUs); 4381 } 4382 mRsmpInIndex = mFrameCount; 4383 framesOut = 0; 4384 buffer.frameCount = 0; 4385 } 4386 } 4387 } 4388 } else { 4389 // resampling 4390 4391 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4392 // alter output frame count as if we were expecting stereo samples 4393 if (mChannelCount == 1 && mReqChannelCount == 1) { 4394 framesOut >>= 1; 4395 } 4396 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4397 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4398 // are 32 bit aligned which should be always true. 4399 if (mChannelCount == 2 && mReqChannelCount == 1) { 4400 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4401 // the resampler always outputs stereo samples: do post stereo to mono conversion 4402 int16_t *src = (int16_t *)mRsmpOutBuffer; 4403 int16_t *dst = buffer.i16; 4404 while (framesOut--) { 4405 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4406 src += 2; 4407 } 4408 } else { 4409 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4410 } 4411 4412 } 4413 mActiveTrack->releaseBuffer(&buffer); 4414 mActiveTrack->overflow(); 4415 } 4416 // client isn't retrieving buffers fast enough 4417 else { 4418 if (!mActiveTrack->setOverflow()) { 4419 nsecs_t now = systemTime(); 4420 if ((now - lastWarning) > kWarningThrottleNs) { 4421 LOGW("RecordThread: buffer overflow"); 4422 lastWarning = now; 4423 } 4424 } 4425 // Release the processor for a while before asking for a new buffer. 4426 // This will give the application more chance to read from the buffer and 4427 // clear the overflow. 4428 usleep(kRecordThreadSleepUs); 4429 } 4430 } 4431 // enable changes in effect chain 4432 unlockEffectChains(effectChains); 4433 effectChains.clear(); 4434 } 4435 4436 if (!mStandby) { 4437 mInput->stream->common.standby(&mInput->stream->common); 4438 } 4439 mActiveTrack.clear(); 4440 4441 mStartStopCond.broadcast(); 4442 4443 releaseWakeLock(); 4444 4445 ALOGV("RecordThread %p exiting", this); 4446 return false; 4447} 4448 4449 4450sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4451 const sp<AudioFlinger::Client>& client, 4452 uint32_t sampleRate, 4453 int format, 4454 int channelMask, 4455 int frameCount, 4456 uint32_t flags, 4457 int sessionId, 4458 status_t *status) 4459{ 4460 sp<RecordTrack> track; 4461 status_t lStatus; 4462 4463 lStatus = initCheck(); 4464 if (lStatus != NO_ERROR) { 4465 LOGE("Audio driver not initialized."); 4466 goto Exit; 4467 } 4468 4469 { // scope for mLock 4470 Mutex::Autolock _l(mLock); 4471 4472 track = new RecordTrack(this, client, sampleRate, 4473 format, channelMask, frameCount, flags, sessionId); 4474 4475 if (track->getCblk() == NULL) { 4476 lStatus = NO_MEMORY; 4477 goto Exit; 4478 } 4479 4480 mTrack = track.get(); 4481 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4482 bool suspend = audio_is_bluetooth_sco_device( 4483 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4484 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4485 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4486 } 4487 lStatus = NO_ERROR; 4488 4489Exit: 4490 if (status) { 4491 *status = lStatus; 4492 } 4493 return track; 4494} 4495 4496status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4497{ 4498 ALOGV("RecordThread::start"); 4499 sp <ThreadBase> strongMe = this; 4500 status_t status = NO_ERROR; 4501 { 4502 AutoMutex lock(&mLock); 4503 if (mActiveTrack != 0) { 4504 if (recordTrack != mActiveTrack.get()) { 4505 status = -EBUSY; 4506 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4507 mActiveTrack->mState = TrackBase::ACTIVE; 4508 } 4509 return status; 4510 } 4511 4512 recordTrack->mState = TrackBase::IDLE; 4513 mActiveTrack = recordTrack; 4514 mLock.unlock(); 4515 status_t status = AudioSystem::startInput(mId); 4516 mLock.lock(); 4517 if (status != NO_ERROR) { 4518 mActiveTrack.clear(); 4519 return status; 4520 } 4521 mRsmpInIndex = mFrameCount; 4522 mBytesRead = 0; 4523 if (mResampler != NULL) { 4524 mResampler->reset(); 4525 } 4526 mActiveTrack->mState = TrackBase::RESUMING; 4527 // signal thread to start 4528 ALOGV("Signal record thread"); 4529 mWaitWorkCV.signal(); 4530 // do not wait for mStartStopCond if exiting 4531 if (mExiting) { 4532 mActiveTrack.clear(); 4533 status = INVALID_OPERATION; 4534 goto startError; 4535 } 4536 mStartStopCond.wait(mLock); 4537 if (mActiveTrack == 0) { 4538 ALOGV("Record failed to start"); 4539 status = BAD_VALUE; 4540 goto startError; 4541 } 4542 ALOGV("Record started OK"); 4543 return status; 4544 } 4545startError: 4546 AudioSystem::stopInput(mId); 4547 return status; 4548} 4549 4550void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4551 ALOGV("RecordThread::stop"); 4552 sp <ThreadBase> strongMe = this; 4553 { 4554 AutoMutex lock(&mLock); 4555 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4556 mActiveTrack->mState = TrackBase::PAUSING; 4557 // do not wait for mStartStopCond if exiting 4558 if (mExiting) { 4559 return; 4560 } 4561 mStartStopCond.wait(mLock); 4562 // if we have been restarted, recordTrack == mActiveTrack.get() here 4563 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4564 mLock.unlock(); 4565 AudioSystem::stopInput(mId); 4566 mLock.lock(); 4567 ALOGV("Record stopped OK"); 4568 } 4569 } 4570 } 4571} 4572 4573status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4574{ 4575 const size_t SIZE = 256; 4576 char buffer[SIZE]; 4577 String8 result; 4578 pid_t pid = 0; 4579 4580 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4581 result.append(buffer); 4582 4583 if (mActiveTrack != 0) { 4584 result.append("Active Track:\n"); 4585 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4586 mActiveTrack->dump(buffer, SIZE); 4587 result.append(buffer); 4588 4589 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4590 result.append(buffer); 4591 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4592 result.append(buffer); 4593 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4594 result.append(buffer); 4595 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4596 result.append(buffer); 4597 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4598 result.append(buffer); 4599 4600 4601 } else { 4602 result.append("No record client\n"); 4603 } 4604 write(fd, result.string(), result.size()); 4605 4606 dumpBase(fd, args); 4607 dumpEffectChains(fd, args); 4608 4609 return NO_ERROR; 4610} 4611 4612status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4613{ 4614 size_t framesReq = buffer->frameCount; 4615 size_t framesReady = mFrameCount - mRsmpInIndex; 4616 int channelCount; 4617 4618 if (framesReady == 0) { 4619 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4620 if (mBytesRead < 0) { 4621 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4622 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4623 // Force input into standby so that it tries to 4624 // recover at next read attempt 4625 mInput->stream->common.standby(&mInput->stream->common); 4626 usleep(kRecordThreadSleepUs); 4627 } 4628 buffer->raw = 0; 4629 buffer->frameCount = 0; 4630 return NOT_ENOUGH_DATA; 4631 } 4632 mRsmpInIndex = 0; 4633 framesReady = mFrameCount; 4634 } 4635 4636 if (framesReq > framesReady) { 4637 framesReq = framesReady; 4638 } 4639 4640 if (mChannelCount == 1 && mReqChannelCount == 2) { 4641 channelCount = 1; 4642 } else { 4643 channelCount = 2; 4644 } 4645 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4646 buffer->frameCount = framesReq; 4647 return NO_ERROR; 4648} 4649 4650void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4651{ 4652 mRsmpInIndex += buffer->frameCount; 4653 buffer->frameCount = 0; 4654} 4655 4656bool AudioFlinger::RecordThread::checkForNewParameters_l() 4657{ 4658 bool reconfig = false; 4659 4660 while (!mNewParameters.isEmpty()) { 4661 status_t status = NO_ERROR; 4662 String8 keyValuePair = mNewParameters[0]; 4663 AudioParameter param = AudioParameter(keyValuePair); 4664 int value; 4665 int reqFormat = mFormat; 4666 int reqSamplingRate = mReqSampleRate; 4667 int reqChannelCount = mReqChannelCount; 4668 4669 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4670 reqSamplingRate = value; 4671 reconfig = true; 4672 } 4673 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4674 reqFormat = value; 4675 reconfig = true; 4676 } 4677 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4678 reqChannelCount = popcount(value); 4679 reconfig = true; 4680 } 4681 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4682 // do not accept frame count changes if tracks are open as the track buffer 4683 // size depends on frame count and correct behavior would not be garantied 4684 // if frame count is changed after track creation 4685 if (mActiveTrack != 0) { 4686 status = INVALID_OPERATION; 4687 } else { 4688 reconfig = true; 4689 } 4690 } 4691 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4692 // forward device change to effects that have requested to be 4693 // aware of attached audio device. 4694 for (size_t i = 0; i < mEffectChains.size(); i++) { 4695 mEffectChains[i]->setDevice_l(value); 4696 } 4697 // store input device and output device but do not forward output device to audio HAL. 4698 // Note that status is ignored by the caller for output device 4699 // (see AudioFlinger::setParameters() 4700 if (value & AUDIO_DEVICE_OUT_ALL) { 4701 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4702 status = BAD_VALUE; 4703 } else { 4704 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4705 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4706 if (mTrack != NULL) { 4707 bool suspend = audio_is_bluetooth_sco_device( 4708 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4709 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4710 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4711 } 4712 } 4713 mDevice |= (uint32_t)value; 4714 } 4715 if (status == NO_ERROR) { 4716 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4717 if (status == INVALID_OPERATION) { 4718 mInput->stream->common.standby(&mInput->stream->common); 4719 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4720 } 4721 if (reconfig) { 4722 if (status == BAD_VALUE && 4723 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4724 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4725 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4726 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4727 (reqChannelCount < 3)) { 4728 status = NO_ERROR; 4729 } 4730 if (status == NO_ERROR) { 4731 readInputParameters(); 4732 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4733 } 4734 } 4735 } 4736 4737 mNewParameters.removeAt(0); 4738 4739 mParamStatus = status; 4740 mParamCond.signal(); 4741 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4742 // already timed out waiting for the status and will never signal the condition. 4743 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4744 } 4745 return reconfig; 4746} 4747 4748String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4749{ 4750 char *s; 4751 String8 out_s8 = String8(); 4752 4753 Mutex::Autolock _l(mLock); 4754 if (initCheck() != NO_ERROR) { 4755 return out_s8; 4756 } 4757 4758 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4759 out_s8 = String8(s); 4760 free(s); 4761 return out_s8; 4762} 4763 4764void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4765 AudioSystem::OutputDescriptor desc; 4766 void *param2 = 0; 4767 4768 switch (event) { 4769 case AudioSystem::INPUT_OPENED: 4770 case AudioSystem::INPUT_CONFIG_CHANGED: 4771 desc.channels = mChannelMask; 4772 desc.samplingRate = mSampleRate; 4773 desc.format = mFormat; 4774 desc.frameCount = mFrameCount; 4775 desc.latency = 0; 4776 param2 = &desc; 4777 break; 4778 4779 case AudioSystem::INPUT_CLOSED: 4780 default: 4781 break; 4782 } 4783 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4784} 4785 4786void AudioFlinger::RecordThread::readInputParameters() 4787{ 4788 if (mRsmpInBuffer) delete mRsmpInBuffer; 4789 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4790 if (mResampler) delete mResampler; 4791 mResampler = 0; 4792 4793 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4794 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4795 mChannelCount = (uint16_t)popcount(mChannelMask); 4796 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4797 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); 4798 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4799 mFrameCount = mInputBytes / mFrameSize; 4800 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4801 4802 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4803 { 4804 int channelCount; 4805 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4806 // stereo to mono post process as the resampler always outputs stereo. 4807 if (mChannelCount == 1 && mReqChannelCount == 2) { 4808 channelCount = 1; 4809 } else { 4810 channelCount = 2; 4811 } 4812 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4813 mResampler->setSampleRate(mSampleRate); 4814 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4815 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4816 4817 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4818 if (mChannelCount == 1 && mReqChannelCount == 1) { 4819 mFrameCount >>= 1; 4820 } 4821 4822 } 4823 mRsmpInIndex = mFrameCount; 4824} 4825 4826unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4827{ 4828 Mutex::Autolock _l(mLock); 4829 if (initCheck() != NO_ERROR) { 4830 return 0; 4831 } 4832 4833 return mInput->stream->get_input_frames_lost(mInput->stream); 4834} 4835 4836uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4837{ 4838 Mutex::Autolock _l(mLock); 4839 uint32_t result = 0; 4840 if (getEffectChain_l(sessionId) != 0) { 4841 result = EFFECT_SESSION; 4842 } 4843 4844 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4845 result |= TRACK_SESSION; 4846 } 4847 4848 return result; 4849} 4850 4851AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4852{ 4853 Mutex::Autolock _l(mLock); 4854 return mTrack; 4855} 4856 4857AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() 4858{ 4859 Mutex::Autolock _l(mLock); 4860 return mInput; 4861} 4862 4863AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4864{ 4865 Mutex::Autolock _l(mLock); 4866 AudioStreamIn *input = mInput; 4867 mInput = NULL; 4868 return input; 4869} 4870 4871// this method must always be called either with ThreadBase mLock held or inside the thread loop 4872audio_stream_t* AudioFlinger::RecordThread::stream() 4873{ 4874 if (mInput == NULL) { 4875 return NULL; 4876 } 4877 return &mInput->stream->common; 4878} 4879 4880 4881// ---------------------------------------------------------------------------- 4882 4883int AudioFlinger::openOutput(uint32_t *pDevices, 4884 uint32_t *pSamplingRate, 4885 uint32_t *pFormat, 4886 uint32_t *pChannels, 4887 uint32_t *pLatencyMs, 4888 uint32_t flags) 4889{ 4890 status_t status; 4891 PlaybackThread *thread = NULL; 4892 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4893 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4894 uint32_t format = pFormat ? *pFormat : 0; 4895 uint32_t channels = pChannels ? *pChannels : 0; 4896 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4897 audio_stream_out_t *outStream; 4898 audio_hw_device_t *outHwDev; 4899 4900 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4901 pDevices ? *pDevices : 0, 4902 samplingRate, 4903 format, 4904 channels, 4905 flags); 4906 4907 if (pDevices == NULL || *pDevices == 0) { 4908 return 0; 4909 } 4910 4911 Mutex::Autolock _l(mLock); 4912 4913 outHwDev = findSuitableHwDev_l(*pDevices); 4914 if (outHwDev == NULL) 4915 return 0; 4916 4917 status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, 4918 &channels, &samplingRate, &outStream); 4919 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4920 outStream, 4921 samplingRate, 4922 format, 4923 channels, 4924 status); 4925 4926 mHardwareStatus = AUDIO_HW_IDLE; 4927 if (outStream != NULL) { 4928 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4929 int id = nextUniqueId(); 4930 4931 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4932 (format != AUDIO_FORMAT_PCM_16_BIT) || 4933 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4934 thread = new DirectOutputThread(this, output, id, *pDevices); 4935 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4936 } else { 4937 thread = new MixerThread(this, output, id, *pDevices); 4938 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4939 } 4940 mPlaybackThreads.add(id, thread); 4941 4942 if (pSamplingRate) *pSamplingRate = samplingRate; 4943 if (pFormat) *pFormat = format; 4944 if (pChannels) *pChannels = channels; 4945 if (pLatencyMs) *pLatencyMs = thread->latency(); 4946 4947 // notify client processes of the new output creation 4948 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4949 return id; 4950 } 4951 4952 return 0; 4953} 4954 4955int AudioFlinger::openDuplicateOutput(int output1, int output2) 4956{ 4957 Mutex::Autolock _l(mLock); 4958 MixerThread *thread1 = checkMixerThread_l(output1); 4959 MixerThread *thread2 = checkMixerThread_l(output2); 4960 4961 if (thread1 == NULL || thread2 == NULL) { 4962 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4963 return 0; 4964 } 4965 4966 int id = nextUniqueId(); 4967 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4968 thread->addOutputTrack(thread2); 4969 mPlaybackThreads.add(id, thread); 4970 // notify client processes of the new output creation 4971 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4972 return id; 4973} 4974 4975status_t AudioFlinger::closeOutput(int output) 4976{ 4977 // keep strong reference on the playback thread so that 4978 // it is not destroyed while exit() is executed 4979 sp <PlaybackThread> thread; 4980 { 4981 Mutex::Autolock _l(mLock); 4982 thread = checkPlaybackThread_l(output); 4983 if (thread == NULL) { 4984 return BAD_VALUE; 4985 } 4986 4987 ALOGV("closeOutput() %d", output); 4988 4989 if (thread->type() == ThreadBase::MIXER) { 4990 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4991 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4992 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4993 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4994 } 4995 } 4996 } 4997 void *param2 = 0; 4998 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4999 mPlaybackThreads.removeItem(output); 5000 } 5001 thread->exit(); 5002 5003 if (thread->type() != ThreadBase::DUPLICATING) { 5004 AudioStreamOut *out = thread->clearOutput(); 5005 // from now on thread->mOutput is NULL 5006 out->hwDev->close_output_stream(out->hwDev, out->stream); 5007 delete out; 5008 } 5009 return NO_ERROR; 5010} 5011 5012status_t AudioFlinger::suspendOutput(int output) 5013{ 5014 Mutex::Autolock _l(mLock); 5015 PlaybackThread *thread = checkPlaybackThread_l(output); 5016 5017 if (thread == NULL) { 5018 return BAD_VALUE; 5019 } 5020 5021 ALOGV("suspendOutput() %d", output); 5022 thread->suspend(); 5023 5024 return NO_ERROR; 5025} 5026 5027status_t AudioFlinger::restoreOutput(int output) 5028{ 5029 Mutex::Autolock _l(mLock); 5030 PlaybackThread *thread = checkPlaybackThread_l(output); 5031 5032 if (thread == NULL) { 5033 return BAD_VALUE; 5034 } 5035 5036 ALOGV("restoreOutput() %d", output); 5037 5038 thread->restore(); 5039 5040 return NO_ERROR; 5041} 5042 5043int AudioFlinger::openInput(uint32_t *pDevices, 5044 uint32_t *pSamplingRate, 5045 uint32_t *pFormat, 5046 uint32_t *pChannels, 5047 uint32_t acoustics) 5048{ 5049 status_t status; 5050 RecordThread *thread = NULL; 5051 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5052 uint32_t format = pFormat ? *pFormat : 0; 5053 uint32_t channels = pChannels ? *pChannels : 0; 5054 uint32_t reqSamplingRate = samplingRate; 5055 uint32_t reqFormat = format; 5056 uint32_t reqChannels = channels; 5057 audio_stream_in_t *inStream; 5058 audio_hw_device_t *inHwDev; 5059 5060 if (pDevices == NULL || *pDevices == 0) { 5061 return 0; 5062 } 5063 5064 Mutex::Autolock _l(mLock); 5065 5066 inHwDev = findSuitableHwDev_l(*pDevices); 5067 if (inHwDev == NULL) 5068 return 0; 5069 5070 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5071 &channels, &samplingRate, 5072 (audio_in_acoustics_t)acoustics, 5073 &inStream); 5074 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5075 inStream, 5076 samplingRate, 5077 format, 5078 channels, 5079 acoustics, 5080 status); 5081 5082 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5083 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5084 // or stereo to mono conversions on 16 bit PCM inputs. 5085 if (inStream == NULL && status == BAD_VALUE && 5086 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5087 (samplingRate <= 2 * reqSamplingRate) && 5088 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5089 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5090 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5091 &channels, &samplingRate, 5092 (audio_in_acoustics_t)acoustics, 5093 &inStream); 5094 } 5095 5096 if (inStream != NULL) { 5097 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5098 5099 int id = nextUniqueId(); 5100 // Start record thread 5101 // RecorThread require both input and output device indication to forward to audio 5102 // pre processing modules 5103 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5104 thread = new RecordThread(this, 5105 input, 5106 reqSamplingRate, 5107 reqChannels, 5108 id, 5109 device); 5110 mRecordThreads.add(id, thread); 5111 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5112 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5113 if (pFormat) *pFormat = format; 5114 if (pChannels) *pChannels = reqChannels; 5115 5116 input->stream->common.standby(&input->stream->common); 5117 5118 // notify client processes of the new input creation 5119 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5120 return id; 5121 } 5122 5123 return 0; 5124} 5125 5126status_t AudioFlinger::closeInput(int input) 5127{ 5128 // keep strong reference on the record thread so that 5129 // it is not destroyed while exit() is executed 5130 sp <RecordThread> thread; 5131 { 5132 Mutex::Autolock _l(mLock); 5133 thread = checkRecordThread_l(input); 5134 if (thread == NULL) { 5135 return BAD_VALUE; 5136 } 5137 5138 ALOGV("closeInput() %d", input); 5139 void *param2 = 0; 5140 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5141 mRecordThreads.removeItem(input); 5142 } 5143 thread->exit(); 5144 5145 AudioStreamIn *in = thread->clearInput(); 5146 // from now on thread->mInput is NULL 5147 in->hwDev->close_input_stream(in->hwDev, in->stream); 5148 delete in; 5149 5150 return NO_ERROR; 5151} 5152 5153status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 5154{ 5155 Mutex::Autolock _l(mLock); 5156 MixerThread *dstThread = checkMixerThread_l(output); 5157 if (dstThread == NULL) { 5158 LOGW("setStreamOutput() bad output id %d", output); 5159 return BAD_VALUE; 5160 } 5161 5162 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5163 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5164 5165 dstThread->setStreamValid(stream, true); 5166 5167 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5168 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5169 if (thread != dstThread && 5170 thread->type() != ThreadBase::DIRECT) { 5171 MixerThread *srcThread = (MixerThread *)thread; 5172 srcThread->setStreamValid(stream, false); 5173 srcThread->invalidateTracks(stream); 5174 } 5175 } 5176 5177 return NO_ERROR; 5178} 5179 5180 5181int AudioFlinger::newAudioSessionId() 5182{ 5183 return nextUniqueId(); 5184} 5185 5186void AudioFlinger::acquireAudioSessionId(int audioSession) 5187{ 5188 Mutex::Autolock _l(mLock); 5189 int caller = IPCThreadState::self()->getCallingPid(); 5190 ALOGV("acquiring %d from %d", audioSession, caller); 5191 int num = mAudioSessionRefs.size(); 5192 for (int i = 0; i< num; i++) { 5193 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5194 if (ref->sessionid == audioSession && ref->pid == caller) { 5195 ref->cnt++; 5196 ALOGV(" incremented refcount to %d", ref->cnt); 5197 return; 5198 } 5199 } 5200 AudioSessionRef *ref = new AudioSessionRef(); 5201 ref->sessionid = audioSession; 5202 ref->pid = caller; 5203 ref->cnt = 1; 5204 mAudioSessionRefs.push(ref); 5205 ALOGV(" added new entry for %d", ref->sessionid); 5206} 5207 5208void AudioFlinger::releaseAudioSessionId(int audioSession) 5209{ 5210 Mutex::Autolock _l(mLock); 5211 int caller = IPCThreadState::self()->getCallingPid(); 5212 ALOGV("releasing %d from %d", audioSession, caller); 5213 int num = mAudioSessionRefs.size(); 5214 for (int i = 0; i< num; i++) { 5215 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5216 if (ref->sessionid == audioSession && ref->pid == caller) { 5217 ref->cnt--; 5218 ALOGV(" decremented refcount to %d", ref->cnt); 5219 if (ref->cnt == 0) { 5220 mAudioSessionRefs.removeAt(i); 5221 delete ref; 5222 purgeStaleEffects_l(); 5223 } 5224 return; 5225 } 5226 } 5227 LOGW("session id %d not found for pid %d", audioSession, caller); 5228} 5229 5230void AudioFlinger::purgeStaleEffects_l() { 5231 5232 ALOGV("purging stale effects"); 5233 5234 Vector< sp<EffectChain> > chains; 5235 5236 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5237 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5238 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5239 sp<EffectChain> ec = t->mEffectChains[j]; 5240 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5241 chains.push(ec); 5242 } 5243 } 5244 } 5245 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5246 sp<RecordThread> t = mRecordThreads.valueAt(i); 5247 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5248 sp<EffectChain> ec = t->mEffectChains[j]; 5249 chains.push(ec); 5250 } 5251 } 5252 5253 for (size_t i = 0; i < chains.size(); i++) { 5254 sp<EffectChain> ec = chains[i]; 5255 int sessionid = ec->sessionId(); 5256 sp<ThreadBase> t = ec->mThread.promote(); 5257 if (t == 0) { 5258 continue; 5259 } 5260 size_t numsessionrefs = mAudioSessionRefs.size(); 5261 bool found = false; 5262 for (size_t k = 0; k < numsessionrefs; k++) { 5263 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5264 if (ref->sessionid == sessionid) { 5265 ALOGV(" session %d still exists for %d with %d refs", 5266 sessionid, ref->pid, ref->cnt); 5267 found = true; 5268 break; 5269 } 5270 } 5271 if (!found) { 5272 // remove all effects from the chain 5273 while (ec->mEffects.size()) { 5274 sp<EffectModule> effect = ec->mEffects[0]; 5275 effect->unPin(); 5276 Mutex::Autolock _l (t->mLock); 5277 t->removeEffect_l(effect); 5278 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5279 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5280 if (handle != 0) { 5281 handle->mEffect.clear(); 5282 if (handle->mHasControl && handle->mEnabled) { 5283 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5284 } 5285 } 5286 } 5287 AudioSystem::unregisterEffect(effect->id()); 5288 } 5289 } 5290 } 5291 return; 5292} 5293 5294// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5295AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5296{ 5297 PlaybackThread *thread = NULL; 5298 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5299 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5300 } 5301 return thread; 5302} 5303 5304// checkMixerThread_l() must be called with AudioFlinger::mLock held 5305AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5306{ 5307 PlaybackThread *thread = checkPlaybackThread_l(output); 5308 if (thread != NULL) { 5309 if (thread->type() == ThreadBase::DIRECT) { 5310 thread = NULL; 5311 } 5312 } 5313 return (MixerThread *)thread; 5314} 5315 5316// checkRecordThread_l() must be called with AudioFlinger::mLock held 5317AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5318{ 5319 RecordThread *thread = NULL; 5320 if (mRecordThreads.indexOfKey(input) >= 0) { 5321 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5322 } 5323 return thread; 5324} 5325 5326uint32_t AudioFlinger::nextUniqueId() 5327{ 5328 return android_atomic_inc(&mNextUniqueId); 5329} 5330 5331AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5332{ 5333 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5334 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5335 AudioStreamOut *output = thread->getOutput(); 5336 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5337 return thread; 5338 } 5339 } 5340 return NULL; 5341} 5342 5343uint32_t AudioFlinger::primaryOutputDevice_l() 5344{ 5345 PlaybackThread *thread = primaryPlaybackThread_l(); 5346 5347 if (thread == NULL) { 5348 return 0; 5349 } 5350 5351 return thread->device(); 5352} 5353 5354 5355// ---------------------------------------------------------------------------- 5356// Effect management 5357// ---------------------------------------------------------------------------- 5358 5359 5360status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5361{ 5362 Mutex::Autolock _l(mLock); 5363 return EffectQueryNumberEffects(numEffects); 5364} 5365 5366status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5367{ 5368 Mutex::Autolock _l(mLock); 5369 return EffectQueryEffect(index, descriptor); 5370} 5371 5372status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5373{ 5374 Mutex::Autolock _l(mLock); 5375 return EffectGetDescriptor(pUuid, descriptor); 5376} 5377 5378 5379sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5380 effect_descriptor_t *pDesc, 5381 const sp<IEffectClient>& effectClient, 5382 int32_t priority, 5383 int io, 5384 int sessionId, 5385 status_t *status, 5386 int *id, 5387 int *enabled) 5388{ 5389 status_t lStatus = NO_ERROR; 5390 sp<EffectHandle> handle; 5391 effect_descriptor_t desc; 5392 sp<Client> client; 5393 wp<Client> wclient; 5394 5395 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5396 pid, effectClient.get(), priority, sessionId, io); 5397 5398 if (pDesc == NULL) { 5399 lStatus = BAD_VALUE; 5400 goto Exit; 5401 } 5402 5403 // check audio settings permission for global effects 5404 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5405 lStatus = PERMISSION_DENIED; 5406 goto Exit; 5407 } 5408 5409 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5410 // that can only be created by audio policy manager (running in same process) 5411 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5412 lStatus = PERMISSION_DENIED; 5413 goto Exit; 5414 } 5415 5416 if (io == 0) { 5417 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5418 // output must be specified by AudioPolicyManager when using session 5419 // AUDIO_SESSION_OUTPUT_STAGE 5420 lStatus = BAD_VALUE; 5421 goto Exit; 5422 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5423 // if the output returned by getOutputForEffect() is removed before we lock the 5424 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5425 // and we will exit safely 5426 io = AudioSystem::getOutputForEffect(&desc); 5427 } 5428 } 5429 5430 { 5431 Mutex::Autolock _l(mLock); 5432 5433 5434 if (!EffectIsNullUuid(&pDesc->uuid)) { 5435 // if uuid is specified, request effect descriptor 5436 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5437 if (lStatus < 0) { 5438 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5439 goto Exit; 5440 } 5441 } else { 5442 // if uuid is not specified, look for an available implementation 5443 // of the required type in effect factory 5444 if (EffectIsNullUuid(&pDesc->type)) { 5445 LOGW("createEffect() no effect type"); 5446 lStatus = BAD_VALUE; 5447 goto Exit; 5448 } 5449 uint32_t numEffects = 0; 5450 effect_descriptor_t d; 5451 d.flags = 0; // prevent compiler warning 5452 bool found = false; 5453 5454 lStatus = EffectQueryNumberEffects(&numEffects); 5455 if (lStatus < 0) { 5456 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5457 goto Exit; 5458 } 5459 for (uint32_t i = 0; i < numEffects; i++) { 5460 lStatus = EffectQueryEffect(i, &desc); 5461 if (lStatus < 0) { 5462 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5463 continue; 5464 } 5465 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5466 // If matching type found save effect descriptor. If the session is 5467 // 0 and the effect is not auxiliary, continue enumeration in case 5468 // an auxiliary version of this effect type is available 5469 found = true; 5470 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5471 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5472 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5473 break; 5474 } 5475 } 5476 } 5477 if (!found) { 5478 lStatus = BAD_VALUE; 5479 LOGW("createEffect() effect not found"); 5480 goto Exit; 5481 } 5482 // For same effect type, chose auxiliary version over insert version if 5483 // connect to output mix (Compliance to OpenSL ES) 5484 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5485 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5486 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5487 } 5488 } 5489 5490 // Do not allow auxiliary effects on a session different from 0 (output mix) 5491 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5492 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5493 lStatus = INVALID_OPERATION; 5494 goto Exit; 5495 } 5496 5497 // check recording permission for visualizer 5498 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5499 !recordingAllowed()) { 5500 lStatus = PERMISSION_DENIED; 5501 goto Exit; 5502 } 5503 5504 // return effect descriptor 5505 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5506 5507 // If output is not specified try to find a matching audio session ID in one of the 5508 // output threads. 5509 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5510 // because of code checking output when entering the function. 5511 // Note: io is never 0 when creating an effect on an input 5512 if (io == 0) { 5513 // look for the thread where the specified audio session is present 5514 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5515 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5516 io = mPlaybackThreads.keyAt(i); 5517 break; 5518 } 5519 } 5520 if (io == 0) { 5521 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5522 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5523 io = mRecordThreads.keyAt(i); 5524 break; 5525 } 5526 } 5527 } 5528 // If no output thread contains the requested session ID, default to 5529 // first output. The effect chain will be moved to the correct output 5530 // thread when a track with the same session ID is created 5531 if (io == 0 && mPlaybackThreads.size()) { 5532 io = mPlaybackThreads.keyAt(0); 5533 } 5534 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5535 } 5536 ThreadBase *thread = checkRecordThread_l(io); 5537 if (thread == NULL) { 5538 thread = checkPlaybackThread_l(io); 5539 if (thread == NULL) { 5540 LOGE("createEffect() unknown output thread"); 5541 lStatus = BAD_VALUE; 5542 goto Exit; 5543 } 5544 } 5545 5546 wclient = mClients.valueFor(pid); 5547 5548 if (wclient != NULL) { 5549 client = wclient.promote(); 5550 } else { 5551 client = new Client(this, pid); 5552 mClients.add(pid, client); 5553 } 5554 5555 // create effect on selected output thread 5556 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5557 &desc, enabled, &lStatus); 5558 if (handle != 0 && id != NULL) { 5559 *id = handle->id(); 5560 } 5561 } 5562 5563Exit: 5564 if(status) { 5565 *status = lStatus; 5566 } 5567 return handle; 5568} 5569 5570status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5571{ 5572 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5573 sessionId, srcOutput, dstOutput); 5574 Mutex::Autolock _l(mLock); 5575 if (srcOutput == dstOutput) { 5576 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 5577 return NO_ERROR; 5578 } 5579 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5580 if (srcThread == NULL) { 5581 LOGW("moveEffects() bad srcOutput %d", srcOutput); 5582 return BAD_VALUE; 5583 } 5584 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5585 if (dstThread == NULL) { 5586 LOGW("moveEffects() bad dstOutput %d", dstOutput); 5587 return BAD_VALUE; 5588 } 5589 5590 Mutex::Autolock _dl(dstThread->mLock); 5591 Mutex::Autolock _sl(srcThread->mLock); 5592 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5593 5594 return NO_ERROR; 5595} 5596 5597// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5598status_t AudioFlinger::moveEffectChain_l(int sessionId, 5599 AudioFlinger::PlaybackThread *srcThread, 5600 AudioFlinger::PlaybackThread *dstThread, 5601 bool reRegister) 5602{ 5603 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5604 sessionId, srcThread, dstThread); 5605 5606 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5607 if (chain == 0) { 5608 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5609 sessionId, srcThread); 5610 return INVALID_OPERATION; 5611 } 5612 5613 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5614 // so that a new chain is created with correct parameters when first effect is added. This is 5615 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5616 // removed. 5617 srcThread->removeEffectChain_l(chain); 5618 5619 // transfer all effects one by one so that new effect chain is created on new thread with 5620 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5621 int dstOutput = dstThread->id(); 5622 sp<EffectChain> dstChain; 5623 uint32_t strategy = 0; // prevent compiler warning 5624 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5625 while (effect != 0) { 5626 srcThread->removeEffect_l(effect); 5627 dstThread->addEffect_l(effect); 5628 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5629 if (effect->state() == EffectModule::ACTIVE || 5630 effect->state() == EffectModule::STOPPING) { 5631 effect->start(); 5632 } 5633 // if the move request is not received from audio policy manager, the effect must be 5634 // re-registered with the new strategy and output 5635 if (dstChain == 0) { 5636 dstChain = effect->chain().promote(); 5637 if (dstChain == 0) { 5638 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5639 srcThread->addEffect_l(effect); 5640 return NO_INIT; 5641 } 5642 strategy = dstChain->strategy(); 5643 } 5644 if (reRegister) { 5645 AudioSystem::unregisterEffect(effect->id()); 5646 AudioSystem::registerEffect(&effect->desc(), 5647 dstOutput, 5648 strategy, 5649 sessionId, 5650 effect->id()); 5651 } 5652 effect = chain->getEffectFromId_l(0); 5653 } 5654 5655 return NO_ERROR; 5656} 5657 5658 5659// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5660sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5661 const sp<AudioFlinger::Client>& client, 5662 const sp<IEffectClient>& effectClient, 5663 int32_t priority, 5664 int sessionId, 5665 effect_descriptor_t *desc, 5666 int *enabled, 5667 status_t *status 5668 ) 5669{ 5670 sp<EffectModule> effect; 5671 sp<EffectHandle> handle; 5672 status_t lStatus; 5673 sp<EffectChain> chain; 5674 bool chainCreated = false; 5675 bool effectCreated = false; 5676 bool effectRegistered = false; 5677 5678 lStatus = initCheck(); 5679 if (lStatus != NO_ERROR) { 5680 LOGW("createEffect_l() Audio driver not initialized."); 5681 goto Exit; 5682 } 5683 5684 // Do not allow effects with session ID 0 on direct output or duplicating threads 5685 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5686 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5687 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5688 desc->name, sessionId); 5689 lStatus = BAD_VALUE; 5690 goto Exit; 5691 } 5692 // Only Pre processor effects are allowed on input threads and only on input threads 5693 if ((mType == RECORD && 5694 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5695 (mType != RECORD && 5696 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5697 LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5698 desc->name, desc->flags, mType); 5699 lStatus = BAD_VALUE; 5700 goto Exit; 5701 } 5702 5703 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5704 5705 { // scope for mLock 5706 Mutex::Autolock _l(mLock); 5707 5708 // check for existing effect chain with the requested audio session 5709 chain = getEffectChain_l(sessionId); 5710 if (chain == 0) { 5711 // create a new chain for this session 5712 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5713 chain = new EffectChain(this, sessionId); 5714 addEffectChain_l(chain); 5715 chain->setStrategy(getStrategyForSession_l(sessionId)); 5716 chainCreated = true; 5717 } else { 5718 effect = chain->getEffectFromDesc_l(desc); 5719 } 5720 5721 ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 5722 5723 if (effect == 0) { 5724 int id = mAudioFlinger->nextUniqueId(); 5725 // Check CPU and memory usage 5726 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5727 if (lStatus != NO_ERROR) { 5728 goto Exit; 5729 } 5730 effectRegistered = true; 5731 // create a new effect module if none present in the chain 5732 effect = new EffectModule(this, chain, desc, id, sessionId); 5733 lStatus = effect->status(); 5734 if (lStatus != NO_ERROR) { 5735 goto Exit; 5736 } 5737 lStatus = chain->addEffect_l(effect); 5738 if (lStatus != NO_ERROR) { 5739 goto Exit; 5740 } 5741 effectCreated = true; 5742 5743 effect->setDevice(mDevice); 5744 effect->setMode(mAudioFlinger->getMode()); 5745 } 5746 // create effect handle and connect it to effect module 5747 handle = new EffectHandle(effect, client, effectClient, priority); 5748 lStatus = effect->addHandle(handle); 5749 if (enabled) { 5750 *enabled = (int)effect->isEnabled(); 5751 } 5752 } 5753 5754Exit: 5755 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5756 Mutex::Autolock _l(mLock); 5757 if (effectCreated) { 5758 chain->removeEffect_l(effect); 5759 } 5760 if (effectRegistered) { 5761 AudioSystem::unregisterEffect(effect->id()); 5762 } 5763 if (chainCreated) { 5764 removeEffectChain_l(chain); 5765 } 5766 handle.clear(); 5767 } 5768 5769 if(status) { 5770 *status = lStatus; 5771 } 5772 return handle; 5773} 5774 5775sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5776{ 5777 sp<EffectModule> effect; 5778 5779 sp<EffectChain> chain = getEffectChain_l(sessionId); 5780 if (chain != 0) { 5781 effect = chain->getEffectFromId_l(effectId); 5782 } 5783 return effect; 5784} 5785 5786// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5787// PlaybackThread::mLock held 5788status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5789{ 5790 // check for existing effect chain with the requested audio session 5791 int sessionId = effect->sessionId(); 5792 sp<EffectChain> chain = getEffectChain_l(sessionId); 5793 bool chainCreated = false; 5794 5795 if (chain == 0) { 5796 // create a new chain for this session 5797 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5798 chain = new EffectChain(this, sessionId); 5799 addEffectChain_l(chain); 5800 chain->setStrategy(getStrategyForSession_l(sessionId)); 5801 chainCreated = true; 5802 } 5803 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5804 5805 if (chain->getEffectFromId_l(effect->id()) != 0) { 5806 LOGW("addEffect_l() %p effect %s already present in chain %p", 5807 this, effect->desc().name, chain.get()); 5808 return BAD_VALUE; 5809 } 5810 5811 status_t status = chain->addEffect_l(effect); 5812 if (status != NO_ERROR) { 5813 if (chainCreated) { 5814 removeEffectChain_l(chain); 5815 } 5816 return status; 5817 } 5818 5819 effect->setDevice(mDevice); 5820 effect->setMode(mAudioFlinger->getMode()); 5821 return NO_ERROR; 5822} 5823 5824void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5825 5826 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5827 effect_descriptor_t desc = effect->desc(); 5828 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5829 detachAuxEffect_l(effect->id()); 5830 } 5831 5832 sp<EffectChain> chain = effect->chain().promote(); 5833 if (chain != 0) { 5834 // remove effect chain if removing last effect 5835 if (chain->removeEffect_l(effect) == 0) { 5836 removeEffectChain_l(chain); 5837 } 5838 } else { 5839 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5840 } 5841} 5842 5843void AudioFlinger::ThreadBase::lockEffectChains_l( 5844 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5845{ 5846 effectChains = mEffectChains; 5847 for (size_t i = 0; i < mEffectChains.size(); i++) { 5848 mEffectChains[i]->lock(); 5849 } 5850} 5851 5852void AudioFlinger::ThreadBase::unlockEffectChains( 5853 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5854{ 5855 for (size_t i = 0; i < effectChains.size(); i++) { 5856 effectChains[i]->unlock(); 5857 } 5858} 5859 5860sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5861{ 5862 Mutex::Autolock _l(mLock); 5863 return getEffectChain_l(sessionId); 5864} 5865 5866sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5867{ 5868 sp<EffectChain> chain; 5869 5870 size_t size = mEffectChains.size(); 5871 for (size_t i = 0; i < size; i++) { 5872 if (mEffectChains[i]->sessionId() == sessionId) { 5873 chain = mEffectChains[i]; 5874 break; 5875 } 5876 } 5877 return chain; 5878} 5879 5880void AudioFlinger::ThreadBase::setMode(uint32_t mode) 5881{ 5882 Mutex::Autolock _l(mLock); 5883 size_t size = mEffectChains.size(); 5884 for (size_t i = 0; i < size; i++) { 5885 mEffectChains[i]->setMode_l(mode); 5886 } 5887} 5888 5889void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5890 const wp<EffectHandle>& handle, 5891 bool unpiniflast) { 5892 5893 Mutex::Autolock _l(mLock); 5894 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5895 // delete the effect module if removing last handle on it 5896 if (effect->removeHandle(handle) == 0) { 5897 if (!effect->isPinned() || unpiniflast) { 5898 removeEffect_l(effect); 5899 AudioSystem::unregisterEffect(effect->id()); 5900 } 5901 } 5902} 5903 5904status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5905{ 5906 int session = chain->sessionId(); 5907 int16_t *buffer = mMixBuffer; 5908 bool ownsBuffer = false; 5909 5910 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5911 if (session > 0) { 5912 // Only one effect chain can be present in direct output thread and it uses 5913 // the mix buffer as input 5914 if (mType != DIRECT) { 5915 size_t numSamples = mFrameCount * mChannelCount; 5916 buffer = new int16_t[numSamples]; 5917 memset(buffer, 0, numSamples * sizeof(int16_t)); 5918 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5919 ownsBuffer = true; 5920 } 5921 5922 // Attach all tracks with same session ID to this chain. 5923 for (size_t i = 0; i < mTracks.size(); ++i) { 5924 sp<Track> track = mTracks[i]; 5925 if (session == track->sessionId()) { 5926 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5927 track->setMainBuffer(buffer); 5928 chain->incTrackCnt(); 5929 } 5930 } 5931 5932 // indicate all active tracks in the chain 5933 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5934 sp<Track> track = mActiveTracks[i].promote(); 5935 if (track == 0) continue; 5936 if (session == track->sessionId()) { 5937 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5938 chain->incActiveTrackCnt(); 5939 } 5940 } 5941 } 5942 5943 chain->setInBuffer(buffer, ownsBuffer); 5944 chain->setOutBuffer(mMixBuffer); 5945 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5946 // chains list in order to be processed last as it contains output stage effects 5947 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5948 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5949 // after track specific effects and before output stage 5950 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5951 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5952 // Effect chain for other sessions are inserted at beginning of effect 5953 // chains list to be processed before output mix effects. Relative order between other 5954 // sessions is not important 5955 size_t size = mEffectChains.size(); 5956 size_t i = 0; 5957 for (i = 0; i < size; i++) { 5958 if (mEffectChains[i]->sessionId() < session) break; 5959 } 5960 mEffectChains.insertAt(chain, i); 5961 checkSuspendOnAddEffectChain_l(chain); 5962 5963 return NO_ERROR; 5964} 5965 5966size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5967{ 5968 int session = chain->sessionId(); 5969 5970 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5971 5972 for (size_t i = 0; i < mEffectChains.size(); i++) { 5973 if (chain == mEffectChains[i]) { 5974 mEffectChains.removeAt(i); 5975 // detach all active tracks from the chain 5976 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5977 sp<Track> track = mActiveTracks[i].promote(); 5978 if (track == 0) continue; 5979 if (session == track->sessionId()) { 5980 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5981 chain.get(), session); 5982 chain->decActiveTrackCnt(); 5983 } 5984 } 5985 5986 // detach all tracks with same session ID from this chain 5987 for (size_t i = 0; i < mTracks.size(); ++i) { 5988 sp<Track> track = mTracks[i]; 5989 if (session == track->sessionId()) { 5990 track->setMainBuffer(mMixBuffer); 5991 chain->decTrackCnt(); 5992 } 5993 } 5994 break; 5995 } 5996 } 5997 return mEffectChains.size(); 5998} 5999 6000status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6001 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6002{ 6003 Mutex::Autolock _l(mLock); 6004 return attachAuxEffect_l(track, EffectId); 6005} 6006 6007status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6008 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6009{ 6010 status_t status = NO_ERROR; 6011 6012 if (EffectId == 0) { 6013 track->setAuxBuffer(0, NULL); 6014 } else { 6015 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6016 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6017 if (effect != 0) { 6018 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6019 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6020 } else { 6021 status = INVALID_OPERATION; 6022 } 6023 } else { 6024 status = BAD_VALUE; 6025 } 6026 } 6027 return status; 6028} 6029 6030void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6031{ 6032 for (size_t i = 0; i < mTracks.size(); ++i) { 6033 sp<Track> track = mTracks[i]; 6034 if (track->auxEffectId() == effectId) { 6035 attachAuxEffect_l(track, 0); 6036 } 6037 } 6038} 6039 6040status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6041{ 6042 // only one chain per input thread 6043 if (mEffectChains.size() != 0) { 6044 return INVALID_OPERATION; 6045 } 6046 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6047 6048 chain->setInBuffer(NULL); 6049 chain->setOutBuffer(NULL); 6050 6051 checkSuspendOnAddEffectChain_l(chain); 6052 6053 mEffectChains.add(chain); 6054 6055 return NO_ERROR; 6056} 6057 6058size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6059{ 6060 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6061 LOGW_IF(mEffectChains.size() != 1, 6062 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6063 chain.get(), mEffectChains.size(), this); 6064 if (mEffectChains.size() == 1) { 6065 mEffectChains.removeAt(0); 6066 } 6067 return 0; 6068} 6069 6070// ---------------------------------------------------------------------------- 6071// EffectModule implementation 6072// ---------------------------------------------------------------------------- 6073 6074#undef LOG_TAG 6075#define LOG_TAG "AudioFlinger::EffectModule" 6076 6077AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6078 const wp<AudioFlinger::EffectChain>& chain, 6079 effect_descriptor_t *desc, 6080 int id, 6081 int sessionId) 6082 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6083 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6084{ 6085 ALOGV("Constructor %p", this); 6086 int lStatus; 6087 sp<ThreadBase> thread = mThread.promote(); 6088 if (thread == 0) { 6089 return; 6090 } 6091 6092 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6093 6094 // create effect engine from effect factory 6095 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6096 6097 if (mStatus != NO_ERROR) { 6098 return; 6099 } 6100 lStatus = init(); 6101 if (lStatus < 0) { 6102 mStatus = lStatus; 6103 goto Error; 6104 } 6105 6106 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6107 mPinned = true; 6108 } 6109 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6110 return; 6111Error: 6112 EffectRelease(mEffectInterface); 6113 mEffectInterface = NULL; 6114 ALOGV("Constructor Error %d", mStatus); 6115} 6116 6117AudioFlinger::EffectModule::~EffectModule() 6118{ 6119 ALOGV("Destructor %p", this); 6120 if (mEffectInterface != NULL) { 6121 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6122 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6123 sp<ThreadBase> thread = mThread.promote(); 6124 if (thread != 0) { 6125 audio_stream_t *stream = thread->stream(); 6126 if (stream != NULL) { 6127 stream->remove_audio_effect(stream, mEffectInterface); 6128 } 6129 } 6130 } 6131 // release effect engine 6132 EffectRelease(mEffectInterface); 6133 } 6134} 6135 6136status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6137{ 6138 status_t status; 6139 6140 Mutex::Autolock _l(mLock); 6141 // First handle in mHandles has highest priority and controls the effect module 6142 int priority = handle->priority(); 6143 size_t size = mHandles.size(); 6144 sp<EffectHandle> h; 6145 size_t i; 6146 for (i = 0; i < size; i++) { 6147 h = mHandles[i].promote(); 6148 if (h == 0) continue; 6149 if (h->priority() <= priority) break; 6150 } 6151 // if inserted in first place, move effect control from previous owner to this handle 6152 if (i == 0) { 6153 bool enabled = false; 6154 if (h != 0) { 6155 enabled = h->enabled(); 6156 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6157 } 6158 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6159 status = NO_ERROR; 6160 } else { 6161 status = ALREADY_EXISTS; 6162 } 6163 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6164 mHandles.insertAt(handle, i); 6165 return status; 6166} 6167 6168size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6169{ 6170 Mutex::Autolock _l(mLock); 6171 size_t size = mHandles.size(); 6172 size_t i; 6173 for (i = 0; i < size; i++) { 6174 if (mHandles[i] == handle) break; 6175 } 6176 if (i == size) { 6177 return size; 6178 } 6179 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6180 6181 bool enabled = false; 6182 EffectHandle *hdl = handle.unsafe_get(); 6183 if (hdl) { 6184 ALOGV("removeHandle() unsafe_get OK"); 6185 enabled = hdl->enabled(); 6186 } 6187 mHandles.removeAt(i); 6188 size = mHandles.size(); 6189 // if removed from first place, move effect control from this handle to next in line 6190 if (i == 0 && size != 0) { 6191 sp<EffectHandle> h = mHandles[0].promote(); 6192 if (h != 0) { 6193 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6194 } 6195 } 6196 6197 // Prevent calls to process() and other functions on effect interface from now on. 6198 // The effect engine will be released by the destructor when the last strong reference on 6199 // this object is released which can happen after next process is called. 6200 if (size == 0 && !mPinned) { 6201 mState = DESTROYED; 6202 } 6203 6204 return size; 6205} 6206 6207sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6208{ 6209 Mutex::Autolock _l(mLock); 6210 sp<EffectHandle> handle; 6211 if (mHandles.size() != 0) { 6212 handle = mHandles[0].promote(); 6213 } 6214 return handle; 6215} 6216 6217void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6218{ 6219 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6220 // keep a strong reference on this EffectModule to avoid calling the 6221 // destructor before we exit 6222 sp<EffectModule> keep(this); 6223 { 6224 sp<ThreadBase> thread = mThread.promote(); 6225 if (thread != 0) { 6226 thread->disconnectEffect(keep, handle, unpiniflast); 6227 } 6228 } 6229} 6230 6231void AudioFlinger::EffectModule::updateState() { 6232 Mutex::Autolock _l(mLock); 6233 6234 switch (mState) { 6235 case RESTART: 6236 reset_l(); 6237 // FALL THROUGH 6238 6239 case STARTING: 6240 // clear auxiliary effect input buffer for next accumulation 6241 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6242 memset(mConfig.inputCfg.buffer.raw, 6243 0, 6244 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6245 } 6246 start_l(); 6247 mState = ACTIVE; 6248 break; 6249 case STOPPING: 6250 stop_l(); 6251 mDisableWaitCnt = mMaxDisableWaitCnt; 6252 mState = STOPPED; 6253 break; 6254 case STOPPED: 6255 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6256 // turn off sequence. 6257 if (--mDisableWaitCnt == 0) { 6258 reset_l(); 6259 mState = IDLE; 6260 } 6261 break; 6262 default: //IDLE , ACTIVE, DESTROYED 6263 break; 6264 } 6265} 6266 6267void AudioFlinger::EffectModule::process() 6268{ 6269 Mutex::Autolock _l(mLock); 6270 6271 if (mState == DESTROYED || mEffectInterface == NULL || 6272 mConfig.inputCfg.buffer.raw == NULL || 6273 mConfig.outputCfg.buffer.raw == NULL) { 6274 return; 6275 } 6276 6277 if (isProcessEnabled()) { 6278 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6279 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6280 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 6281 mConfig.inputCfg.buffer.s32, 6282 mConfig.inputCfg.buffer.frameCount/2); 6283 } 6284 6285 // do the actual processing in the effect engine 6286 int ret = (*mEffectInterface)->process(mEffectInterface, 6287 &mConfig.inputCfg.buffer, 6288 &mConfig.outputCfg.buffer); 6289 6290 // force transition to IDLE state when engine is ready 6291 if (mState == STOPPED && ret == -ENODATA) { 6292 mDisableWaitCnt = 1; 6293 } 6294 6295 // clear auxiliary effect input buffer for next accumulation 6296 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6297 memset(mConfig.inputCfg.buffer.raw, 0, 6298 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6299 } 6300 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6301 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6302 // If an insert effect is idle and input buffer is different from output buffer, 6303 // accumulate input onto output 6304 sp<EffectChain> chain = mChain.promote(); 6305 if (chain != 0 && chain->activeTrackCnt() != 0) { 6306 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6307 int16_t *in = mConfig.inputCfg.buffer.s16; 6308 int16_t *out = mConfig.outputCfg.buffer.s16; 6309 for (size_t i = 0; i < frameCnt; i++) { 6310 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6311 } 6312 } 6313 } 6314} 6315 6316void AudioFlinger::EffectModule::reset_l() 6317{ 6318 if (mEffectInterface == NULL) { 6319 return; 6320 } 6321 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6322} 6323 6324status_t AudioFlinger::EffectModule::configure() 6325{ 6326 uint32_t channels; 6327 if (mEffectInterface == NULL) { 6328 return NO_INIT; 6329 } 6330 6331 sp<ThreadBase> thread = mThread.promote(); 6332 if (thread == 0) { 6333 return DEAD_OBJECT; 6334 } 6335 6336 // TODO: handle configuration of effects replacing track process 6337 if (thread->channelCount() == 1) { 6338 channels = AUDIO_CHANNEL_OUT_MONO; 6339 } else { 6340 channels = AUDIO_CHANNEL_OUT_STEREO; 6341 } 6342 6343 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6344 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6345 } else { 6346 mConfig.inputCfg.channels = channels; 6347 } 6348 mConfig.outputCfg.channels = channels; 6349 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6350 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6351 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6352 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6353 mConfig.inputCfg.bufferProvider.cookie = NULL; 6354 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6355 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6356 mConfig.outputCfg.bufferProvider.cookie = NULL; 6357 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6358 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6359 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6360 // Insert effect: 6361 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6362 // always overwrites output buffer: input buffer == output buffer 6363 // - in other sessions: 6364 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6365 // other effect: overwrites output buffer: input buffer == output buffer 6366 // Auxiliary effect: 6367 // accumulates in output buffer: input buffer != output buffer 6368 // Therefore: accumulate <=> input buffer != output buffer 6369 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6370 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6371 } else { 6372 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6373 } 6374 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6375 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6376 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6377 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6378 6379 ALOGV("configure() %p thread %p buffer %p framecount %d", 6380 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6381 6382 status_t cmdStatus; 6383 uint32_t size = sizeof(int); 6384 status_t status = (*mEffectInterface)->command(mEffectInterface, 6385 EFFECT_CMD_CONFIGURE, 6386 sizeof(effect_config_t), 6387 &mConfig, 6388 &size, 6389 &cmdStatus); 6390 if (status == 0) { 6391 status = cmdStatus; 6392 } 6393 6394 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6395 (1000 * mConfig.outputCfg.buffer.frameCount); 6396 6397 return status; 6398} 6399 6400status_t AudioFlinger::EffectModule::init() 6401{ 6402 Mutex::Autolock _l(mLock); 6403 if (mEffectInterface == NULL) { 6404 return NO_INIT; 6405 } 6406 status_t cmdStatus; 6407 uint32_t size = sizeof(status_t); 6408 status_t status = (*mEffectInterface)->command(mEffectInterface, 6409 EFFECT_CMD_INIT, 6410 0, 6411 NULL, 6412 &size, 6413 &cmdStatus); 6414 if (status == 0) { 6415 status = cmdStatus; 6416 } 6417 return status; 6418} 6419 6420status_t AudioFlinger::EffectModule::start() 6421{ 6422 Mutex::Autolock _l(mLock); 6423 return start_l(); 6424} 6425 6426status_t AudioFlinger::EffectModule::start_l() 6427{ 6428 if (mEffectInterface == NULL) { 6429 return NO_INIT; 6430 } 6431 status_t cmdStatus; 6432 uint32_t size = sizeof(status_t); 6433 status_t status = (*mEffectInterface)->command(mEffectInterface, 6434 EFFECT_CMD_ENABLE, 6435 0, 6436 NULL, 6437 &size, 6438 &cmdStatus); 6439 if (status == 0) { 6440 status = cmdStatus; 6441 } 6442 if (status == 0 && 6443 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6444 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6445 sp<ThreadBase> thread = mThread.promote(); 6446 if (thread != 0) { 6447 audio_stream_t *stream = thread->stream(); 6448 if (stream != NULL) { 6449 stream->add_audio_effect(stream, mEffectInterface); 6450 } 6451 } 6452 } 6453 return status; 6454} 6455 6456status_t AudioFlinger::EffectModule::stop() 6457{ 6458 Mutex::Autolock _l(mLock); 6459 return stop_l(); 6460} 6461 6462status_t AudioFlinger::EffectModule::stop_l() 6463{ 6464 if (mEffectInterface == NULL) { 6465 return NO_INIT; 6466 } 6467 status_t cmdStatus; 6468 uint32_t size = sizeof(status_t); 6469 status_t status = (*mEffectInterface)->command(mEffectInterface, 6470 EFFECT_CMD_DISABLE, 6471 0, 6472 NULL, 6473 &size, 6474 &cmdStatus); 6475 if (status == 0) { 6476 status = cmdStatus; 6477 } 6478 if (status == 0 && 6479 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6480 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6481 sp<ThreadBase> thread = mThread.promote(); 6482 if (thread != 0) { 6483 audio_stream_t *stream = thread->stream(); 6484 if (stream != NULL) { 6485 stream->remove_audio_effect(stream, mEffectInterface); 6486 } 6487 } 6488 } 6489 return status; 6490} 6491 6492status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6493 uint32_t cmdSize, 6494 void *pCmdData, 6495 uint32_t *replySize, 6496 void *pReplyData) 6497{ 6498 Mutex::Autolock _l(mLock); 6499// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6500 6501 if (mState == DESTROYED || mEffectInterface == NULL) { 6502 return NO_INIT; 6503 } 6504 status_t status = (*mEffectInterface)->command(mEffectInterface, 6505 cmdCode, 6506 cmdSize, 6507 pCmdData, 6508 replySize, 6509 pReplyData); 6510 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6511 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6512 for (size_t i = 1; i < mHandles.size(); i++) { 6513 sp<EffectHandle> h = mHandles[i].promote(); 6514 if (h != 0) { 6515 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6516 } 6517 } 6518 } 6519 return status; 6520} 6521 6522status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6523{ 6524 6525 Mutex::Autolock _l(mLock); 6526 ALOGV("setEnabled %p enabled %d", this, enabled); 6527 6528 if (enabled != isEnabled()) { 6529 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6530 if (enabled && status != NO_ERROR) { 6531 return status; 6532 } 6533 6534 switch (mState) { 6535 // going from disabled to enabled 6536 case IDLE: 6537 mState = STARTING; 6538 break; 6539 case STOPPED: 6540 mState = RESTART; 6541 break; 6542 case STOPPING: 6543 mState = ACTIVE; 6544 break; 6545 6546 // going from enabled to disabled 6547 case RESTART: 6548 mState = STOPPED; 6549 break; 6550 case STARTING: 6551 mState = IDLE; 6552 break; 6553 case ACTIVE: 6554 mState = STOPPING; 6555 break; 6556 case DESTROYED: 6557 return NO_ERROR; // simply ignore as we are being destroyed 6558 } 6559 for (size_t i = 1; i < mHandles.size(); i++) { 6560 sp<EffectHandle> h = mHandles[i].promote(); 6561 if (h != 0) { 6562 h->setEnabled(enabled); 6563 } 6564 } 6565 } 6566 return NO_ERROR; 6567} 6568 6569bool AudioFlinger::EffectModule::isEnabled() 6570{ 6571 switch (mState) { 6572 case RESTART: 6573 case STARTING: 6574 case ACTIVE: 6575 return true; 6576 case IDLE: 6577 case STOPPING: 6578 case STOPPED: 6579 case DESTROYED: 6580 default: 6581 return false; 6582 } 6583} 6584 6585bool AudioFlinger::EffectModule::isProcessEnabled() 6586{ 6587 switch (mState) { 6588 case RESTART: 6589 case ACTIVE: 6590 case STOPPING: 6591 case STOPPED: 6592 return true; 6593 case IDLE: 6594 case STARTING: 6595 case DESTROYED: 6596 default: 6597 return false; 6598 } 6599} 6600 6601status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6602{ 6603 Mutex::Autolock _l(mLock); 6604 status_t status = NO_ERROR; 6605 6606 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6607 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6608 if (isProcessEnabled() && 6609 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6610 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6611 status_t cmdStatus; 6612 uint32_t volume[2]; 6613 uint32_t *pVolume = NULL; 6614 uint32_t size = sizeof(volume); 6615 volume[0] = *left; 6616 volume[1] = *right; 6617 if (controller) { 6618 pVolume = volume; 6619 } 6620 status = (*mEffectInterface)->command(mEffectInterface, 6621 EFFECT_CMD_SET_VOLUME, 6622 size, 6623 volume, 6624 &size, 6625 pVolume); 6626 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6627 *left = volume[0]; 6628 *right = volume[1]; 6629 } 6630 } 6631 return status; 6632} 6633 6634status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6635{ 6636 Mutex::Autolock _l(mLock); 6637 status_t status = NO_ERROR; 6638 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6639 // audio pre processing modules on RecordThread can receive both output and 6640 // input device indication in the same call 6641 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6642 if (dev) { 6643 status_t cmdStatus; 6644 uint32_t size = sizeof(status_t); 6645 6646 status = (*mEffectInterface)->command(mEffectInterface, 6647 EFFECT_CMD_SET_DEVICE, 6648 sizeof(uint32_t), 6649 &dev, 6650 &size, 6651 &cmdStatus); 6652 if (status == NO_ERROR) { 6653 status = cmdStatus; 6654 } 6655 } 6656 dev = device & AUDIO_DEVICE_IN_ALL; 6657 if (dev) { 6658 status_t cmdStatus; 6659 uint32_t size = sizeof(status_t); 6660 6661 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6662 EFFECT_CMD_SET_INPUT_DEVICE, 6663 sizeof(uint32_t), 6664 &dev, 6665 &size, 6666 &cmdStatus); 6667 if (status2 == NO_ERROR) { 6668 status2 = cmdStatus; 6669 } 6670 if (status == NO_ERROR) { 6671 status = status2; 6672 } 6673 } 6674 } 6675 return status; 6676} 6677 6678status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 6679{ 6680 Mutex::Autolock _l(mLock); 6681 status_t status = NO_ERROR; 6682 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6683 status_t cmdStatus; 6684 uint32_t size = sizeof(status_t); 6685 status = (*mEffectInterface)->command(mEffectInterface, 6686 EFFECT_CMD_SET_AUDIO_MODE, 6687 sizeof(int), 6688 &mode, 6689 &size, 6690 &cmdStatus); 6691 if (status == NO_ERROR) { 6692 status = cmdStatus; 6693 } 6694 } 6695 return status; 6696} 6697 6698void AudioFlinger::EffectModule::setSuspended(bool suspended) 6699{ 6700 Mutex::Autolock _l(mLock); 6701 mSuspended = suspended; 6702} 6703bool AudioFlinger::EffectModule::suspended() 6704{ 6705 Mutex::Autolock _l(mLock); 6706 return mSuspended; 6707} 6708 6709status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6710{ 6711 const size_t SIZE = 256; 6712 char buffer[SIZE]; 6713 String8 result; 6714 6715 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6716 result.append(buffer); 6717 6718 bool locked = tryLock(mLock); 6719 // failed to lock - AudioFlinger is probably deadlocked 6720 if (!locked) { 6721 result.append("\t\tCould not lock Fx mutex:\n"); 6722 } 6723 6724 result.append("\t\tSession Status State Engine:\n"); 6725 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6726 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6727 result.append(buffer); 6728 6729 result.append("\t\tDescriptor:\n"); 6730 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6731 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6732 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6733 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6734 result.append(buffer); 6735 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6736 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6737 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6738 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6739 result.append(buffer); 6740 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6741 mDescriptor.apiVersion, 6742 mDescriptor.flags); 6743 result.append(buffer); 6744 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6745 mDescriptor.name); 6746 result.append(buffer); 6747 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6748 mDescriptor.implementor); 6749 result.append(buffer); 6750 6751 result.append("\t\t- Input configuration:\n"); 6752 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6753 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6754 (uint32_t)mConfig.inputCfg.buffer.raw, 6755 mConfig.inputCfg.buffer.frameCount, 6756 mConfig.inputCfg.samplingRate, 6757 mConfig.inputCfg.channels, 6758 mConfig.inputCfg.format); 6759 result.append(buffer); 6760 6761 result.append("\t\t- Output configuration:\n"); 6762 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6763 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6764 (uint32_t)mConfig.outputCfg.buffer.raw, 6765 mConfig.outputCfg.buffer.frameCount, 6766 mConfig.outputCfg.samplingRate, 6767 mConfig.outputCfg.channels, 6768 mConfig.outputCfg.format); 6769 result.append(buffer); 6770 6771 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6772 result.append(buffer); 6773 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6774 for (size_t i = 0; i < mHandles.size(); ++i) { 6775 sp<EffectHandle> handle = mHandles[i].promote(); 6776 if (handle != 0) { 6777 handle->dump(buffer, SIZE); 6778 result.append(buffer); 6779 } 6780 } 6781 6782 result.append("\n"); 6783 6784 write(fd, result.string(), result.length()); 6785 6786 if (locked) { 6787 mLock.unlock(); 6788 } 6789 6790 return NO_ERROR; 6791} 6792 6793// ---------------------------------------------------------------------------- 6794// EffectHandle implementation 6795// ---------------------------------------------------------------------------- 6796 6797#undef LOG_TAG 6798#define LOG_TAG "AudioFlinger::EffectHandle" 6799 6800AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6801 const sp<AudioFlinger::Client>& client, 6802 const sp<IEffectClient>& effectClient, 6803 int32_t priority) 6804 : BnEffect(), 6805 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6806 mPriority(priority), mHasControl(false), mEnabled(false) 6807{ 6808 ALOGV("constructor %p", this); 6809 6810 if (client == 0) { 6811 return; 6812 } 6813 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6814 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6815 if (mCblkMemory != 0) { 6816 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6817 6818 if (mCblk) { 6819 new(mCblk) effect_param_cblk_t(); 6820 mBuffer = (uint8_t *)mCblk + bufOffset; 6821 } 6822 } else { 6823 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6824 return; 6825 } 6826} 6827 6828AudioFlinger::EffectHandle::~EffectHandle() 6829{ 6830 ALOGV("Destructor %p", this); 6831 disconnect(false); 6832 ALOGV("Destructor DONE %p", this); 6833} 6834 6835status_t AudioFlinger::EffectHandle::enable() 6836{ 6837 ALOGV("enable %p", this); 6838 if (!mHasControl) return INVALID_OPERATION; 6839 if (mEffect == 0) return DEAD_OBJECT; 6840 6841 if (mEnabled) { 6842 return NO_ERROR; 6843 } 6844 6845 mEnabled = true; 6846 6847 sp<ThreadBase> thread = mEffect->thread().promote(); 6848 if (thread != 0) { 6849 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6850 } 6851 6852 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6853 if (mEffect->suspended()) { 6854 return NO_ERROR; 6855 } 6856 6857 status_t status = mEffect->setEnabled(true); 6858 if (status != NO_ERROR) { 6859 if (thread != 0) { 6860 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6861 } 6862 mEnabled = false; 6863 } 6864 return status; 6865} 6866 6867status_t AudioFlinger::EffectHandle::disable() 6868{ 6869 ALOGV("disable %p", this); 6870 if (!mHasControl) return INVALID_OPERATION; 6871 if (mEffect == 0) return DEAD_OBJECT; 6872 6873 if (!mEnabled) { 6874 return NO_ERROR; 6875 } 6876 mEnabled = false; 6877 6878 if (mEffect->suspended()) { 6879 return NO_ERROR; 6880 } 6881 6882 status_t status = mEffect->setEnabled(false); 6883 6884 sp<ThreadBase> thread = mEffect->thread().promote(); 6885 if (thread != 0) { 6886 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6887 } 6888 6889 return status; 6890} 6891 6892void AudioFlinger::EffectHandle::disconnect() 6893{ 6894 disconnect(true); 6895} 6896 6897void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6898{ 6899 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6900 if (mEffect == 0) { 6901 return; 6902 } 6903 mEffect->disconnect(this, unpiniflast); 6904 6905 if (mHasControl && mEnabled) { 6906 sp<ThreadBase> thread = mEffect->thread().promote(); 6907 if (thread != 0) { 6908 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6909 } 6910 } 6911 6912 // release sp on module => module destructor can be called now 6913 mEffect.clear(); 6914 if (mClient != 0) { 6915 if (mCblk) { 6916 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6917 } 6918 mCblkMemory.clear(); // and free the shared memory 6919 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6920 mClient.clear(); 6921 } 6922} 6923 6924status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6925 uint32_t cmdSize, 6926 void *pCmdData, 6927 uint32_t *replySize, 6928 void *pReplyData) 6929{ 6930// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6931// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6932 6933 // only get parameter command is permitted for applications not controlling the effect 6934 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6935 return INVALID_OPERATION; 6936 } 6937 if (mEffect == 0) return DEAD_OBJECT; 6938 if (mClient == 0) return INVALID_OPERATION; 6939 6940 // handle commands that are not forwarded transparently to effect engine 6941 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6942 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6943 // no risk to block the whole media server process or mixer threads is we are stuck here 6944 Mutex::Autolock _l(mCblk->lock); 6945 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6946 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6947 mCblk->serverIndex = 0; 6948 mCblk->clientIndex = 0; 6949 return BAD_VALUE; 6950 } 6951 status_t status = NO_ERROR; 6952 while (mCblk->serverIndex < mCblk->clientIndex) { 6953 int reply; 6954 uint32_t rsize = sizeof(int); 6955 int *p = (int *)(mBuffer + mCblk->serverIndex); 6956 int size = *p++; 6957 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6958 LOGW("command(): invalid parameter block size"); 6959 break; 6960 } 6961 effect_param_t *param = (effect_param_t *)p; 6962 if (param->psize == 0 || param->vsize == 0) { 6963 LOGW("command(): null parameter or value size"); 6964 mCblk->serverIndex += size; 6965 continue; 6966 } 6967 uint32_t psize = sizeof(effect_param_t) + 6968 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6969 param->vsize; 6970 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6971 psize, 6972 p, 6973 &rsize, 6974 &reply); 6975 // stop at first error encountered 6976 if (ret != NO_ERROR) { 6977 status = ret; 6978 *(int *)pReplyData = reply; 6979 break; 6980 } else if (reply != NO_ERROR) { 6981 *(int *)pReplyData = reply; 6982 break; 6983 } 6984 mCblk->serverIndex += size; 6985 } 6986 mCblk->serverIndex = 0; 6987 mCblk->clientIndex = 0; 6988 return status; 6989 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6990 *(int *)pReplyData = NO_ERROR; 6991 return enable(); 6992 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6993 *(int *)pReplyData = NO_ERROR; 6994 return disable(); 6995 } 6996 6997 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6998} 6999 7000sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7001 return mCblkMemory; 7002} 7003 7004void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7005{ 7006 ALOGV("setControl %p control %d", this, hasControl); 7007 7008 mHasControl = hasControl; 7009 mEnabled = enabled; 7010 7011 if (signal && mEffectClient != 0) { 7012 mEffectClient->controlStatusChanged(hasControl); 7013 } 7014} 7015 7016void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7017 uint32_t cmdSize, 7018 void *pCmdData, 7019 uint32_t replySize, 7020 void *pReplyData) 7021{ 7022 if (mEffectClient != 0) { 7023 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7024 } 7025} 7026 7027 7028 7029void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7030{ 7031 if (mEffectClient != 0) { 7032 mEffectClient->enableStatusChanged(enabled); 7033 } 7034} 7035 7036status_t AudioFlinger::EffectHandle::onTransact( 7037 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7038{ 7039 return BnEffect::onTransact(code, data, reply, flags); 7040} 7041 7042 7043void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7044{ 7045 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7046 7047 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7048 (mClient == NULL) ? getpid() : mClient->pid(), 7049 mPriority, 7050 mHasControl, 7051 !locked, 7052 mCblk ? mCblk->clientIndex : 0, 7053 mCblk ? mCblk->serverIndex : 0 7054 ); 7055 7056 if (locked) { 7057 mCblk->lock.unlock(); 7058 } 7059} 7060 7061#undef LOG_TAG 7062#define LOG_TAG "AudioFlinger::EffectChain" 7063 7064AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7065 int sessionId) 7066 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7067 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7068 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7069{ 7070 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7071 sp<ThreadBase> thread = mThread.promote(); 7072 if (thread == 0) { 7073 return; 7074 } 7075 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7076 thread->frameCount(); 7077} 7078 7079AudioFlinger::EffectChain::~EffectChain() 7080{ 7081 if (mOwnInBuffer) { 7082 delete mInBuffer; 7083 } 7084 7085} 7086 7087// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7088sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7089{ 7090 sp<EffectModule> effect; 7091 size_t size = mEffects.size(); 7092 7093 for (size_t i = 0; i < size; i++) { 7094 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7095 effect = mEffects[i]; 7096 break; 7097 } 7098 } 7099 return effect; 7100} 7101 7102// getEffectFromId_l() must be called with ThreadBase::mLock held 7103sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7104{ 7105 sp<EffectModule> effect; 7106 size_t size = mEffects.size(); 7107 7108 for (size_t i = 0; i < size; i++) { 7109 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7110 if (id == 0 || mEffects[i]->id() == id) { 7111 effect = mEffects[i]; 7112 break; 7113 } 7114 } 7115 return effect; 7116} 7117 7118// getEffectFromType_l() must be called with ThreadBase::mLock held 7119sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7120 const effect_uuid_t *type) 7121{ 7122 sp<EffectModule> effect; 7123 size_t size = mEffects.size(); 7124 7125 for (size_t i = 0; i < size; i++) { 7126 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7127 effect = mEffects[i]; 7128 break; 7129 } 7130 } 7131 return effect; 7132} 7133 7134// Must be called with EffectChain::mLock locked 7135void AudioFlinger::EffectChain::process_l() 7136{ 7137 sp<ThreadBase> thread = mThread.promote(); 7138 if (thread == 0) { 7139 LOGW("process_l(): cannot promote mixer thread"); 7140 return; 7141 } 7142 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7143 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7144 // always process effects unless no more tracks are on the session and the effect tail 7145 // has been rendered 7146 bool doProcess = true; 7147 if (!isGlobalSession) { 7148 bool tracksOnSession = (trackCnt() != 0); 7149 7150 if (!tracksOnSession && mTailBufferCount == 0) { 7151 doProcess = false; 7152 } 7153 7154 if (activeTrackCnt() == 0) { 7155 // if no track is active and the effect tail has not been rendered, 7156 // the input buffer must be cleared here as the mixer process will not do it 7157 if (tracksOnSession || mTailBufferCount > 0) { 7158 size_t numSamples = thread->frameCount() * thread->channelCount(); 7159 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7160 if (mTailBufferCount > 0) { 7161 mTailBufferCount--; 7162 } 7163 } 7164 } 7165 } 7166 7167 size_t size = mEffects.size(); 7168 if (doProcess) { 7169 for (size_t i = 0; i < size; i++) { 7170 mEffects[i]->process(); 7171 } 7172 } 7173 for (size_t i = 0; i < size; i++) { 7174 mEffects[i]->updateState(); 7175 } 7176} 7177 7178// addEffect_l() must be called with PlaybackThread::mLock held 7179status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7180{ 7181 effect_descriptor_t desc = effect->desc(); 7182 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7183 7184 Mutex::Autolock _l(mLock); 7185 effect->setChain(this); 7186 sp<ThreadBase> thread = mThread.promote(); 7187 if (thread == 0) { 7188 return NO_INIT; 7189 } 7190 effect->setThread(thread); 7191 7192 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7193 // Auxiliary effects are inserted at the beginning of mEffects vector as 7194 // they are processed first and accumulated in chain input buffer 7195 mEffects.insertAt(effect, 0); 7196 7197 // the input buffer for auxiliary effect contains mono samples in 7198 // 32 bit format. This is to avoid saturation in AudoMixer 7199 // accumulation stage. Saturation is done in EffectModule::process() before 7200 // calling the process in effect engine 7201 size_t numSamples = thread->frameCount(); 7202 int32_t *buffer = new int32_t[numSamples]; 7203 memset(buffer, 0, numSamples * sizeof(int32_t)); 7204 effect->setInBuffer((int16_t *)buffer); 7205 // auxiliary effects output samples to chain input buffer for further processing 7206 // by insert effects 7207 effect->setOutBuffer(mInBuffer); 7208 } else { 7209 // Insert effects are inserted at the end of mEffects vector as they are processed 7210 // after track and auxiliary effects. 7211 // Insert effect order as a function of indicated preference: 7212 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7213 // another effect is present 7214 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7215 // last effect claiming first position 7216 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7217 // first effect claiming last position 7218 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7219 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7220 // already present 7221 7222 int size = (int)mEffects.size(); 7223 int idx_insert = size; 7224 int idx_insert_first = -1; 7225 int idx_insert_last = -1; 7226 7227 for (int i = 0; i < size; i++) { 7228 effect_descriptor_t d = mEffects[i]->desc(); 7229 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7230 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7231 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7232 // check invalid effect chaining combinations 7233 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7234 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7235 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7236 return INVALID_OPERATION; 7237 } 7238 // remember position of first insert effect and by default 7239 // select this as insert position for new effect 7240 if (idx_insert == size) { 7241 idx_insert = i; 7242 } 7243 // remember position of last insert effect claiming 7244 // first position 7245 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7246 idx_insert_first = i; 7247 } 7248 // remember position of first insert effect claiming 7249 // last position 7250 if (iPref == EFFECT_FLAG_INSERT_LAST && 7251 idx_insert_last == -1) { 7252 idx_insert_last = i; 7253 } 7254 } 7255 } 7256 7257 // modify idx_insert from first position if needed 7258 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7259 if (idx_insert_last != -1) { 7260 idx_insert = idx_insert_last; 7261 } else { 7262 idx_insert = size; 7263 } 7264 } else { 7265 if (idx_insert_first != -1) { 7266 idx_insert = idx_insert_first + 1; 7267 } 7268 } 7269 7270 // always read samples from chain input buffer 7271 effect->setInBuffer(mInBuffer); 7272 7273 // if last effect in the chain, output samples to chain 7274 // output buffer, otherwise to chain input buffer 7275 if (idx_insert == size) { 7276 if (idx_insert != 0) { 7277 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7278 mEffects[idx_insert-1]->configure(); 7279 } 7280 effect->setOutBuffer(mOutBuffer); 7281 } else { 7282 effect->setOutBuffer(mInBuffer); 7283 } 7284 mEffects.insertAt(effect, idx_insert); 7285 7286 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7287 } 7288 effect->configure(); 7289 return NO_ERROR; 7290} 7291 7292// removeEffect_l() must be called with PlaybackThread::mLock held 7293size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7294{ 7295 Mutex::Autolock _l(mLock); 7296 int size = (int)mEffects.size(); 7297 int i; 7298 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7299 7300 for (i = 0; i < size; i++) { 7301 if (effect == mEffects[i]) { 7302 // calling stop here will remove pre-processing effect from the audio HAL. 7303 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7304 // the middle of a read from audio HAL 7305 if (mEffects[i]->state() == EffectModule::ACTIVE || 7306 mEffects[i]->state() == EffectModule::STOPPING) { 7307 mEffects[i]->stop(); 7308 } 7309 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7310 delete[] effect->inBuffer(); 7311 } else { 7312 if (i == size - 1 && i != 0) { 7313 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7314 mEffects[i - 1]->configure(); 7315 } 7316 } 7317 mEffects.removeAt(i); 7318 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7319 break; 7320 } 7321 } 7322 7323 return mEffects.size(); 7324} 7325 7326// setDevice_l() must be called with PlaybackThread::mLock held 7327void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7328{ 7329 size_t size = mEffects.size(); 7330 for (size_t i = 0; i < size; i++) { 7331 mEffects[i]->setDevice(device); 7332 } 7333} 7334 7335// setMode_l() must be called with PlaybackThread::mLock held 7336void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 7337{ 7338 size_t size = mEffects.size(); 7339 for (size_t i = 0; i < size; i++) { 7340 mEffects[i]->setMode(mode); 7341 } 7342} 7343 7344// setVolume_l() must be called with PlaybackThread::mLock held 7345bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7346{ 7347 uint32_t newLeft = *left; 7348 uint32_t newRight = *right; 7349 bool hasControl = false; 7350 int ctrlIdx = -1; 7351 size_t size = mEffects.size(); 7352 7353 // first update volume controller 7354 for (size_t i = size; i > 0; i--) { 7355 if (mEffects[i - 1]->isProcessEnabled() && 7356 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7357 ctrlIdx = i - 1; 7358 hasControl = true; 7359 break; 7360 } 7361 } 7362 7363 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7364 if (hasControl) { 7365 *left = mNewLeftVolume; 7366 *right = mNewRightVolume; 7367 } 7368 return hasControl; 7369 } 7370 7371 mVolumeCtrlIdx = ctrlIdx; 7372 mLeftVolume = newLeft; 7373 mRightVolume = newRight; 7374 7375 // second get volume update from volume controller 7376 if (ctrlIdx >= 0) { 7377 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7378 mNewLeftVolume = newLeft; 7379 mNewRightVolume = newRight; 7380 } 7381 // then indicate volume to all other effects in chain. 7382 // Pass altered volume to effects before volume controller 7383 // and requested volume to effects after controller 7384 uint32_t lVol = newLeft; 7385 uint32_t rVol = newRight; 7386 7387 for (size_t i = 0; i < size; i++) { 7388 if ((int)i == ctrlIdx) continue; 7389 // this also works for ctrlIdx == -1 when there is no volume controller 7390 if ((int)i > ctrlIdx) { 7391 lVol = *left; 7392 rVol = *right; 7393 } 7394 mEffects[i]->setVolume(&lVol, &rVol, false); 7395 } 7396 *left = newLeft; 7397 *right = newRight; 7398 7399 return hasControl; 7400} 7401 7402status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7403{ 7404 const size_t SIZE = 256; 7405 char buffer[SIZE]; 7406 String8 result; 7407 7408 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7409 result.append(buffer); 7410 7411 bool locked = tryLock(mLock); 7412 // failed to lock - AudioFlinger is probably deadlocked 7413 if (!locked) { 7414 result.append("\tCould not lock mutex:\n"); 7415 } 7416 7417 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7418 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7419 mEffects.size(), 7420 (uint32_t)mInBuffer, 7421 (uint32_t)mOutBuffer, 7422 mActiveTrackCnt); 7423 result.append(buffer); 7424 write(fd, result.string(), result.size()); 7425 7426 for (size_t i = 0; i < mEffects.size(); ++i) { 7427 sp<EffectModule> effect = mEffects[i]; 7428 if (effect != 0) { 7429 effect->dump(fd, args); 7430 } 7431 } 7432 7433 if (locked) { 7434 mLock.unlock(); 7435 } 7436 7437 return NO_ERROR; 7438} 7439 7440// must be called with ThreadBase::mLock held 7441void AudioFlinger::EffectChain::setEffectSuspended_l( 7442 const effect_uuid_t *type, bool suspend) 7443{ 7444 sp<SuspendedEffectDesc> desc; 7445 // use effect type UUID timelow as key as there is no real risk of identical 7446 // timeLow fields among effect type UUIDs. 7447 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7448 if (suspend) { 7449 if (index >= 0) { 7450 desc = mSuspendedEffects.valueAt(index); 7451 } else { 7452 desc = new SuspendedEffectDesc(); 7453 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7454 mSuspendedEffects.add(type->timeLow, desc); 7455 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7456 } 7457 if (desc->mRefCount++ == 0) { 7458 sp<EffectModule> effect = getEffectIfEnabled(type); 7459 if (effect != 0) { 7460 desc->mEffect = effect; 7461 effect->setSuspended(true); 7462 effect->setEnabled(false); 7463 } 7464 } 7465 } else { 7466 if (index < 0) { 7467 return; 7468 } 7469 desc = mSuspendedEffects.valueAt(index); 7470 if (desc->mRefCount <= 0) { 7471 LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7472 desc->mRefCount = 1; 7473 } 7474 if (--desc->mRefCount == 0) { 7475 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7476 if (desc->mEffect != 0) { 7477 sp<EffectModule> effect = desc->mEffect.promote(); 7478 if (effect != 0) { 7479 effect->setSuspended(false); 7480 sp<EffectHandle> handle = effect->controlHandle(); 7481 if (handle != 0) { 7482 effect->setEnabled(handle->enabled()); 7483 } 7484 } 7485 desc->mEffect.clear(); 7486 } 7487 mSuspendedEffects.removeItemsAt(index); 7488 } 7489 } 7490} 7491 7492// must be called with ThreadBase::mLock held 7493void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7494{ 7495 sp<SuspendedEffectDesc> desc; 7496 7497 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7498 if (suspend) { 7499 if (index >= 0) { 7500 desc = mSuspendedEffects.valueAt(index); 7501 } else { 7502 desc = new SuspendedEffectDesc(); 7503 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7504 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7505 } 7506 if (desc->mRefCount++ == 0) { 7507 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7508 for (size_t i = 0; i < effects.size(); i++) { 7509 setEffectSuspended_l(&effects[i]->desc().type, true); 7510 } 7511 } 7512 } else { 7513 if (index < 0) { 7514 return; 7515 } 7516 desc = mSuspendedEffects.valueAt(index); 7517 if (desc->mRefCount <= 0) { 7518 LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7519 desc->mRefCount = 1; 7520 } 7521 if (--desc->mRefCount == 0) { 7522 Vector<const effect_uuid_t *> types; 7523 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7524 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7525 continue; 7526 } 7527 types.add(&mSuspendedEffects.valueAt(i)->mType); 7528 } 7529 for (size_t i = 0; i < types.size(); i++) { 7530 setEffectSuspended_l(types[i], false); 7531 } 7532 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7533 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7534 } 7535 } 7536} 7537 7538 7539// The volume effect is used for automated tests only 7540#ifndef OPENSL_ES_H_ 7541static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7542 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7543const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7544#endif //OPENSL_ES_H_ 7545 7546bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7547{ 7548 // auxiliary effects and visualizer are never suspended on output mix 7549 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7550 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7551 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7552 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7553 return false; 7554 } 7555 return true; 7556} 7557 7558Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7559{ 7560 Vector< sp<EffectModule> > effects; 7561 for (size_t i = 0; i < mEffects.size(); i++) { 7562 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7563 continue; 7564 } 7565 effects.add(mEffects[i]); 7566 } 7567 return effects; 7568} 7569 7570sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7571 const effect_uuid_t *type) 7572{ 7573 sp<EffectModule> effect; 7574 effect = getEffectFromType_l(type); 7575 if (effect != 0 && !effect->isEnabled()) { 7576 effect.clear(); 7577 } 7578 return effect; 7579} 7580 7581void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7582 bool enabled) 7583{ 7584 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7585 if (enabled) { 7586 if (index < 0) { 7587 // if the effect is not suspend check if all effects are suspended 7588 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7589 if (index < 0) { 7590 return; 7591 } 7592 if (!isEffectEligibleForSuspend(effect->desc())) { 7593 return; 7594 } 7595 setEffectSuspended_l(&effect->desc().type, enabled); 7596 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7597 if (index < 0) { 7598 LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7599 return; 7600 } 7601 } 7602 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7603 effect->desc().type.timeLow); 7604 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7605 // if effect is requested to suspended but was not yet enabled, supend it now. 7606 if (desc->mEffect == 0) { 7607 desc->mEffect = effect; 7608 effect->setEnabled(false); 7609 effect->setSuspended(true); 7610 } 7611 } else { 7612 if (index < 0) { 7613 return; 7614 } 7615 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7616 effect->desc().type.timeLow); 7617 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7618 desc->mEffect.clear(); 7619 effect->setSuspended(false); 7620 } 7621} 7622 7623#undef LOG_TAG 7624#define LOG_TAG "AudioFlinger" 7625 7626// ---------------------------------------------------------------------------- 7627 7628status_t AudioFlinger::onTransact( 7629 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7630{ 7631 return BnAudioFlinger::onTransact(code, data, reply, flags); 7632} 7633 7634}; // namespace android 7635