AudioFlinger.cpp revision 7ab41c9f773ba599646f1b0d00955c1be80f92fd
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 <audio_utils/primitives.h> 58 59#include <cpustats/ThreadCpuUsage.h> 60#include <powermanager/PowerManager.h> 61// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 62 63// ---------------------------------------------------------------------------- 64 65 66namespace android { 67 68static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 69static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 70 71//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 72static const float MAX_GAIN = 4096.0f; 73static const float MAX_GAIN_INT = 0x1000; 74 75// retry counts for buffer fill timeout 76// 50 * ~20msecs = 1 second 77static const int8_t kMaxTrackRetries = 50; 78static const int8_t kMaxTrackStartupRetries = 50; 79// allow less retry attempts on direct output thread. 80// direct outputs can be a scarce resource in audio hardware and should 81// be released as quickly as possible. 82static const int8_t kMaxTrackRetriesDirect = 2; 83 84static const int kDumpLockRetries = 50; 85static const int kDumpLockSleepUs = 20000; 86 87// don't warn about blocked writes or record buffer overflows more often than this 88static const nsecs_t kWarningThrottleNs = seconds(5); 89 90// RecordThread loop sleep time upon application overrun or audio HAL read error 91static const int kRecordThreadSleepUs = 5000; 92 93// maximum time to wait for setParameters to complete 94static const nsecs_t kSetParametersTimeoutNs = seconds(2); 95 96// minimum sleep time for the mixer thread loop when tracks are active but in underrun 97static const uint32_t kMinThreadSleepTimeUs = 5000; 98// maximum divider applied to the active sleep time in the mixer thread loop 99static const uint32_t kMaxThreadSleepTimeShift = 2; 100 101 102// ---------------------------------------------------------------------------- 103 104static bool recordingAllowed() { 105 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 106 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 107 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 108 return ok; 109} 110 111static bool settingsAllowed() { 112 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 113 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 114 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 115 return ok; 116} 117 118// To collect the amplifier usage 119static void addBatteryData(uint32_t params) { 120 sp<IBinder> binder = 121 defaultServiceManager()->getService(String16("media.player")); 122 sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder); 123 if (service.get() == NULL) { 124 LOGW("Cannot connect to the MediaPlayerService for battery tracking"); 125 return; 126 } 127 128 service->addBatteryData(params); 129} 130 131static int load_audio_interface(const char *if_name, const hw_module_t **mod, 132 audio_hw_device_t **dev) 133{ 134 int rc; 135 136 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 137 if (rc) 138 goto out; 139 140 rc = audio_hw_device_open(*mod, dev); 141 LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 142 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 143 if (rc) 144 goto out; 145 146 return 0; 147 148out: 149 *mod = NULL; 150 *dev = NULL; 151 return rc; 152} 153 154static const char * const audio_interfaces[] = { 155 "primary", 156 "a2dp", 157 "usb", 158}; 159#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 160 161// ---------------------------------------------------------------------------- 162 163AudioFlinger::AudioFlinger() 164 : BnAudioFlinger(), 165 mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 166 mBtNrecIsOff(false) 167{ 168} 169 170void AudioFlinger::onFirstRef() 171{ 172 int rc = 0; 173 174 Mutex::Autolock _l(mLock); 175 176 /* TODO: move all this work into an Init() function */ 177 mHardwareStatus = AUDIO_HW_IDLE; 178 179 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 180 const hw_module_t *mod; 181 audio_hw_device_t *dev; 182 183 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 184 if (rc) 185 continue; 186 187 LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 188 mod->name, mod->id); 189 mAudioHwDevs.push(dev); 190 191 if (!mPrimaryHardwareDev) { 192 mPrimaryHardwareDev = dev; 193 LOGI("Using '%s' (%s.%s) as the primary audio interface", 194 mod->name, mod->id, audio_interfaces[i]); 195 } 196 } 197 198 mHardwareStatus = AUDIO_HW_INIT; 199 200 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 201 LOGE("Primary audio interface not found"); 202 return; 203 } 204 205 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 206 audio_hw_device_t *dev = mAudioHwDevs[i]; 207 208 mHardwareStatus = AUDIO_HW_INIT; 209 rc = dev->init_check(dev); 210 if (rc == 0) { 211 AutoMutex lock(mHardwareLock); 212 213 mMode = AUDIO_MODE_NORMAL; 214 mHardwareStatus = AUDIO_HW_SET_MODE; 215 dev->set_mode(dev, mMode); 216 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 217 dev->set_master_volume(dev, 1.0f); 218 mHardwareStatus = AUDIO_HW_IDLE; 219 } 220 } 221} 222 223status_t AudioFlinger::initCheck() const 224{ 225 Mutex::Autolock _l(mLock); 226 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 227 return NO_INIT; 228 return NO_ERROR; 229} 230 231AudioFlinger::~AudioFlinger() 232{ 233 int num_devs = mAudioHwDevs.size(); 234 235 while (!mRecordThreads.isEmpty()) { 236 // closeInput() will remove first entry from mRecordThreads 237 closeInput(mRecordThreads.keyAt(0)); 238 } 239 while (!mPlaybackThreads.isEmpty()) { 240 // closeOutput() will remove first entry from mPlaybackThreads 241 closeOutput(mPlaybackThreads.keyAt(0)); 242 } 243 244 for (int i = 0; i < num_devs; i++) { 245 audio_hw_device_t *dev = mAudioHwDevs[i]; 246 audio_hw_device_close(dev); 247 } 248 mAudioHwDevs.clear(); 249} 250 251audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 252{ 253 /* first matching HW device is returned */ 254 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 255 audio_hw_device_t *dev = mAudioHwDevs[i]; 256 if ((dev->get_supported_devices(dev) & devices) == devices) 257 return dev; 258 } 259 return NULL; 260} 261 262status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 263{ 264 const size_t SIZE = 256; 265 char buffer[SIZE]; 266 String8 result; 267 268 result.append("Clients:\n"); 269 for (size_t i = 0; i < mClients.size(); ++i) { 270 wp<Client> wClient = mClients.valueAt(i); 271 if (wClient != 0) { 272 sp<Client> client = wClient.promote(); 273 if (client != 0) { 274 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 275 result.append(buffer); 276 } 277 } 278 } 279 280 result.append("Global session refs:\n"); 281 result.append(" session pid cnt\n"); 282 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 283 AudioSessionRef *r = mAudioSessionRefs[i]; 284 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 285 result.append(buffer); 286 } 287 write(fd, result.string(), result.size()); 288 return NO_ERROR; 289} 290 291 292status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 293{ 294 const size_t SIZE = 256; 295 char buffer[SIZE]; 296 String8 result; 297 int hardwareStatus = mHardwareStatus; 298 299 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 300 result.append(buffer); 301 write(fd, result.string(), result.size()); 302 return NO_ERROR; 303} 304 305status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 306{ 307 const size_t SIZE = 256; 308 char buffer[SIZE]; 309 String8 result; 310 snprintf(buffer, SIZE, "Permission Denial: " 311 "can't dump AudioFlinger from pid=%d, uid=%d\n", 312 IPCThreadState::self()->getCallingPid(), 313 IPCThreadState::self()->getCallingUid()); 314 result.append(buffer); 315 write(fd, result.string(), result.size()); 316 return NO_ERROR; 317} 318 319static bool tryLock(Mutex& mutex) 320{ 321 bool locked = false; 322 for (int i = 0; i < kDumpLockRetries; ++i) { 323 if (mutex.tryLock() == NO_ERROR) { 324 locked = true; 325 break; 326 } 327 usleep(kDumpLockSleepUs); 328 } 329 return locked; 330} 331 332status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 333{ 334 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 335 dumpPermissionDenial(fd, args); 336 } else { 337 // get state of hardware lock 338 bool hardwareLocked = tryLock(mHardwareLock); 339 if (!hardwareLocked) { 340 String8 result(kHardwareLockedString); 341 write(fd, result.string(), result.size()); 342 } else { 343 mHardwareLock.unlock(); 344 } 345 346 bool locked = tryLock(mLock); 347 348 // failed to lock - AudioFlinger is probably deadlocked 349 if (!locked) { 350 String8 result(kDeadlockedString); 351 write(fd, result.string(), result.size()); 352 } 353 354 dumpClients(fd, args); 355 dumpInternals(fd, args); 356 357 // dump playback threads 358 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 359 mPlaybackThreads.valueAt(i)->dump(fd, args); 360 } 361 362 // dump record threads 363 for (size_t i = 0; i < mRecordThreads.size(); i++) { 364 mRecordThreads.valueAt(i)->dump(fd, args); 365 } 366 367 // dump all hardware devs 368 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 369 audio_hw_device_t *dev = mAudioHwDevs[i]; 370 dev->dump(dev, fd); 371 } 372 if (locked) mLock.unlock(); 373 } 374 return NO_ERROR; 375} 376 377 378// IAudioFlinger interface 379 380 381sp<IAudioTrack> AudioFlinger::createTrack( 382 pid_t pid, 383 int streamType, 384 uint32_t sampleRate, 385 uint32_t format, 386 uint32_t channelMask, 387 int frameCount, 388 uint32_t flags, 389 const sp<IMemory>& sharedBuffer, 390 int output, 391 int *sessionId, 392 status_t *status) 393{ 394 sp<PlaybackThread::Track> track; 395 sp<TrackHandle> trackHandle; 396 sp<Client> client; 397 wp<Client> wclient; 398 status_t lStatus; 399 int lSessionId; 400 401 if (streamType >= AUDIO_STREAM_CNT) { 402 LOGE("createTrack() invalid stream type %d", streamType); 403 lStatus = BAD_VALUE; 404 goto Exit; 405 } 406 407 { 408 Mutex::Autolock _l(mLock); 409 PlaybackThread *thread = checkPlaybackThread_l(output); 410 PlaybackThread *effectThread = NULL; 411 if (thread == NULL) { 412 LOGE("unknown output thread"); 413 lStatus = BAD_VALUE; 414 goto Exit; 415 } 416 417 wclient = mClients.valueFor(pid); 418 419 if (wclient != NULL) { 420 client = wclient.promote(); 421 } else { 422 client = new Client(this, pid); 423 mClients.add(pid, client); 424 } 425 426 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 427 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 428 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 429 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 430 if (mPlaybackThreads.keyAt(i) != output) { 431 // prevent same audio session on different output threads 432 uint32_t sessions = t->hasAudioSession(*sessionId); 433 if (sessions & PlaybackThread::TRACK_SESSION) { 434 LOGE("createTrack() session ID %d already in use", *sessionId); 435 lStatus = BAD_VALUE; 436 goto Exit; 437 } 438 // check if an effect with same session ID is waiting for a track to be created 439 if (sessions & PlaybackThread::EFFECT_SESSION) { 440 effectThread = t.get(); 441 } 442 } 443 } 444 lSessionId = *sessionId; 445 } else { 446 // if no audio session id is provided, create one here 447 lSessionId = nextUniqueId(); 448 if (sessionId != NULL) { 449 *sessionId = lSessionId; 450 } 451 } 452 ALOGV("createTrack() lSessionId: %d", lSessionId); 453 454 track = thread->createTrack_l(client, streamType, sampleRate, format, 455 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 456 457 // move effect chain to this output thread if an effect on same session was waiting 458 // for a track to be created 459 if (lStatus == NO_ERROR && effectThread != NULL) { 460 Mutex::Autolock _dl(thread->mLock); 461 Mutex::Autolock _sl(effectThread->mLock); 462 moveEffectChain_l(lSessionId, effectThread, thread, true); 463 } 464 } 465 if (lStatus == NO_ERROR) { 466 trackHandle = new TrackHandle(track); 467 } else { 468 // remove local strong reference to Client before deleting the Track so that the Client 469 // destructor is called by the TrackBase destructor with mLock held 470 client.clear(); 471 track.clear(); 472 } 473 474Exit: 475 if(status) { 476 *status = lStatus; 477 } 478 return trackHandle; 479} 480 481uint32_t AudioFlinger::sampleRate(int output) const 482{ 483 Mutex::Autolock _l(mLock); 484 PlaybackThread *thread = checkPlaybackThread_l(output); 485 if (thread == NULL) { 486 LOGW("sampleRate() unknown thread %d", output); 487 return 0; 488 } 489 return thread->sampleRate(); 490} 491 492int AudioFlinger::channelCount(int output) const 493{ 494 Mutex::Autolock _l(mLock); 495 PlaybackThread *thread = checkPlaybackThread_l(output); 496 if (thread == NULL) { 497 LOGW("channelCount() unknown thread %d", output); 498 return 0; 499 } 500 return thread->channelCount(); 501} 502 503uint32_t AudioFlinger::format(int output) const 504{ 505 Mutex::Autolock _l(mLock); 506 PlaybackThread *thread = checkPlaybackThread_l(output); 507 if (thread == NULL) { 508 LOGW("format() unknown thread %d", output); 509 return 0; 510 } 511 return thread->format(); 512} 513 514size_t AudioFlinger::frameCount(int output) const 515{ 516 Mutex::Autolock _l(mLock); 517 PlaybackThread *thread = checkPlaybackThread_l(output); 518 if (thread == NULL) { 519 LOGW("frameCount() unknown thread %d", output); 520 return 0; 521 } 522 return thread->frameCount(); 523} 524 525uint32_t AudioFlinger::latency(int output) const 526{ 527 Mutex::Autolock _l(mLock); 528 PlaybackThread *thread = checkPlaybackThread_l(output); 529 if (thread == NULL) { 530 LOGW("latency() unknown thread %d", output); 531 return 0; 532 } 533 return thread->latency(); 534} 535 536status_t AudioFlinger::setMasterVolume(float value) 537{ 538 status_t ret = initCheck(); 539 if (ret != NO_ERROR) { 540 return ret; 541 } 542 543 // check calling permissions 544 if (!settingsAllowed()) { 545 return PERMISSION_DENIED; 546 } 547 548 // when hw supports master volume, don't scale in sw mixer 549 { // scope for the lock 550 AutoMutex lock(mHardwareLock); 551 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 552 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 553 value = 1.0f; 554 } 555 mHardwareStatus = AUDIO_HW_IDLE; 556 } 557 558 Mutex::Autolock _l(mLock); 559 mMasterVolume = value; 560 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 561 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 562 563 return NO_ERROR; 564} 565 566status_t AudioFlinger::setMode(int mode) 567{ 568 status_t ret = initCheck(); 569 if (ret != NO_ERROR) { 570 return ret; 571 } 572 573 // check calling permissions 574 if (!settingsAllowed()) { 575 return PERMISSION_DENIED; 576 } 577 if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) { 578 LOGW("Illegal value: setMode(%d)", mode); 579 return BAD_VALUE; 580 } 581 582 { // scope for the lock 583 AutoMutex lock(mHardwareLock); 584 mHardwareStatus = AUDIO_HW_SET_MODE; 585 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 586 mHardwareStatus = AUDIO_HW_IDLE; 587 } 588 589 if (NO_ERROR == ret) { 590 Mutex::Autolock _l(mLock); 591 mMode = mode; 592 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 593 mPlaybackThreads.valueAt(i)->setMode(mode); 594 } 595 596 return ret; 597} 598 599status_t AudioFlinger::setMicMute(bool state) 600{ 601 status_t ret = initCheck(); 602 if (ret != NO_ERROR) { 603 return ret; 604 } 605 606 // check calling permissions 607 if (!settingsAllowed()) { 608 return PERMISSION_DENIED; 609 } 610 611 AutoMutex lock(mHardwareLock); 612 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 613 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 614 mHardwareStatus = AUDIO_HW_IDLE; 615 return ret; 616} 617 618bool AudioFlinger::getMicMute() const 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return false; 623 } 624 625 bool state = AUDIO_MODE_INVALID; 626 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 627 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 628 mHardwareStatus = AUDIO_HW_IDLE; 629 return state; 630} 631 632status_t AudioFlinger::setMasterMute(bool muted) 633{ 634 // check calling permissions 635 if (!settingsAllowed()) { 636 return PERMISSION_DENIED; 637 } 638 639 Mutex::Autolock _l(mLock); 640 mMasterMute = muted; 641 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 642 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 643 644 return NO_ERROR; 645} 646 647float AudioFlinger::masterVolume() const 648{ 649 return mMasterVolume; 650} 651 652bool AudioFlinger::masterMute() const 653{ 654 return mMasterMute; 655} 656 657status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 658{ 659 // check calling permissions 660 if (!settingsAllowed()) { 661 return PERMISSION_DENIED; 662 } 663 664 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 665 LOGE("setStreamVolume() invalid stream %d", stream); 666 return BAD_VALUE; 667 } 668 669 AutoMutex lock(mLock); 670 PlaybackThread *thread = NULL; 671 if (output) { 672 thread = checkPlaybackThread_l(output); 673 if (thread == NULL) { 674 return BAD_VALUE; 675 } 676 } 677 678 mStreamTypes[stream].volume = value; 679 680 if (thread == NULL) { 681 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 682 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 683 } 684 } else { 685 thread->setStreamVolume(stream, value); 686 } 687 688 return NO_ERROR; 689} 690 691status_t AudioFlinger::setStreamMute(int stream, bool muted) 692{ 693 // check calling permissions 694 if (!settingsAllowed()) { 695 return PERMISSION_DENIED; 696 } 697 698 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || 699 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 700 LOGE("setStreamMute() invalid stream %d", stream); 701 return BAD_VALUE; 702 } 703 704 AutoMutex lock(mLock); 705 mStreamTypes[stream].mute = muted; 706 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 707 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 708 709 return NO_ERROR; 710} 711 712float AudioFlinger::streamVolume(int stream, int output) const 713{ 714 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 715 return 0.0f; 716 } 717 718 AutoMutex lock(mLock); 719 float volume; 720 if (output) { 721 PlaybackThread *thread = checkPlaybackThread_l(output); 722 if (thread == NULL) { 723 return 0.0f; 724 } 725 volume = thread->streamVolume(stream); 726 } else { 727 volume = mStreamTypes[stream].volume; 728 } 729 730 return volume; 731} 732 733bool AudioFlinger::streamMute(int stream) const 734{ 735 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 736 return true; 737 } 738 739 return mStreamTypes[stream].mute; 740} 741 742status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 743{ 744 status_t result; 745 746 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 747 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 748 // check calling permissions 749 if (!settingsAllowed()) { 750 return PERMISSION_DENIED; 751 } 752 753 // ioHandle == 0 means the parameters are global to the audio hardware interface 754 if (ioHandle == 0) { 755 AutoMutex lock(mHardwareLock); 756 mHardwareStatus = AUDIO_SET_PARAMETER; 757 status_t final_result = NO_ERROR; 758 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 759 audio_hw_device_t *dev = mAudioHwDevs[i]; 760 result = dev->set_parameters(dev, keyValuePairs.string()); 761 final_result = result ?: final_result; 762 } 763 mHardwareStatus = AUDIO_HW_IDLE; 764 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 765 AudioParameter param = AudioParameter(keyValuePairs); 766 String8 value; 767 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 768 Mutex::Autolock _l(mLock); 769 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 770 if (mBtNrecIsOff != btNrecIsOff) { 771 for (size_t i = 0; i < mRecordThreads.size(); i++) { 772 sp<RecordThread> thread = mRecordThreads.valueAt(i); 773 RecordThread::RecordTrack *track = thread->track(); 774 if (track != NULL) { 775 audio_devices_t device = (audio_devices_t)( 776 thread->device() & AUDIO_DEVICE_IN_ALL); 777 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 778 thread->setEffectSuspended(FX_IID_AEC, 779 suspend, 780 track->sessionId()); 781 thread->setEffectSuspended(FX_IID_NS, 782 suspend, 783 track->sessionId()); 784 } 785 } 786 mBtNrecIsOff = btNrecIsOff; 787 } 788 } 789 return final_result; 790 } 791 792 // hold a strong ref on thread in case closeOutput() or closeInput() is called 793 // and the thread is exited once the lock is released 794 sp<ThreadBase> thread; 795 { 796 Mutex::Autolock _l(mLock); 797 thread = checkPlaybackThread_l(ioHandle); 798 if (thread == NULL) { 799 thread = checkRecordThread_l(ioHandle); 800 } else if (thread.get() == primaryPlaybackThread_l()) { 801 // indicate output device change to all input threads for pre processing 802 AudioParameter param = AudioParameter(keyValuePairs); 803 int value; 804 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 805 for (size_t i = 0; i < mRecordThreads.size(); i++) { 806 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 807 } 808 } 809 } 810 } 811 if (thread != NULL) { 812 result = thread->setParameters(keyValuePairs); 813 return result; 814 } 815 return BAD_VALUE; 816} 817 818String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 819{ 820// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 821// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 822 823 if (ioHandle == 0) { 824 String8 out_s8; 825 826 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 827 audio_hw_device_t *dev = mAudioHwDevs[i]; 828 char *s = dev->get_parameters(dev, keys.string()); 829 out_s8 += String8(s); 830 free(s); 831 } 832 return out_s8; 833 } 834 835 Mutex::Autolock _l(mLock); 836 837 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 838 if (playbackThread != NULL) { 839 return playbackThread->getParameters(keys); 840 } 841 RecordThread *recordThread = checkRecordThread_l(ioHandle); 842 if (recordThread != NULL) { 843 return recordThread->getParameters(keys); 844 } 845 return String8(""); 846} 847 848size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 849{ 850 status_t ret = initCheck(); 851 if (ret != NO_ERROR) { 852 return 0; 853 } 854 855 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 856} 857 858unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 859{ 860 if (ioHandle == 0) { 861 return 0; 862 } 863 864 Mutex::Autolock _l(mLock); 865 866 RecordThread *recordThread = checkRecordThread_l(ioHandle); 867 if (recordThread != NULL) { 868 return recordThread->getInputFramesLost(); 869 } 870 return 0; 871} 872 873status_t AudioFlinger::setVoiceVolume(float value) 874{ 875 status_t ret = initCheck(); 876 if (ret != NO_ERROR) { 877 return ret; 878 } 879 880 // check calling permissions 881 if (!settingsAllowed()) { 882 return PERMISSION_DENIED; 883 } 884 885 AutoMutex lock(mHardwareLock); 886 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 887 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 888 mHardwareStatus = AUDIO_HW_IDLE; 889 890 return ret; 891} 892 893status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 894{ 895 status_t status; 896 897 Mutex::Autolock _l(mLock); 898 899 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 900 if (playbackThread != NULL) { 901 return playbackThread->getRenderPosition(halFrames, dspFrames); 902 } 903 904 return BAD_VALUE; 905} 906 907void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 908{ 909 910 Mutex::Autolock _l(mLock); 911 912 int pid = IPCThreadState::self()->getCallingPid(); 913 if (mNotificationClients.indexOfKey(pid) < 0) { 914 sp<NotificationClient> notificationClient = new NotificationClient(this, 915 client, 916 pid); 917 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 918 919 mNotificationClients.add(pid, notificationClient); 920 921 sp<IBinder> binder = client->asBinder(); 922 binder->linkToDeath(notificationClient); 923 924 // the config change is always sent from playback or record threads to avoid deadlock 925 // with AudioSystem::gLock 926 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 927 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 928 } 929 930 for (size_t i = 0; i < mRecordThreads.size(); i++) { 931 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 932 } 933 } 934} 935 936void AudioFlinger::removeNotificationClient(pid_t pid) 937{ 938 Mutex::Autolock _l(mLock); 939 940 int index = mNotificationClients.indexOfKey(pid); 941 if (index >= 0) { 942 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 943 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 944 mNotificationClients.removeItem(pid); 945 } 946 947 ALOGV("%d died, releasing its sessions", pid); 948 int num = mAudioSessionRefs.size(); 949 bool removed = false; 950 for (int i = 0; i< num; i++) { 951 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 952 ALOGV(" pid %d @ %d", ref->pid, i); 953 if (ref->pid == pid) { 954 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 955 mAudioSessionRefs.removeAt(i); 956 delete ref; 957 removed = true; 958 i--; 959 num--; 960 } 961 } 962 if (removed) { 963 purgeStaleEffects_l(); 964 } 965} 966 967// audioConfigChanged_l() must be called with AudioFlinger::mLock held 968void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 969{ 970 size_t size = mNotificationClients.size(); 971 for (size_t i = 0; i < size; i++) { 972 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 973 } 974} 975 976// removeClient_l() must be called with AudioFlinger::mLock held 977void AudioFlinger::removeClient_l(pid_t pid) 978{ 979 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 980 mClients.removeItem(pid); 981} 982 983 984// ---------------------------------------------------------------------------- 985 986AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 987 : Thread(false), 988 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 989 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), 990 mDevice(device) 991{ 992 mDeathRecipient = new PMDeathRecipient(this); 993} 994 995AudioFlinger::ThreadBase::~ThreadBase() 996{ 997 mParamCond.broadcast(); 998 // do not lock the mutex in destructor 999 releaseWakeLock_l(); 1000 if (mPowerManager != 0) { 1001 sp<IBinder> binder = mPowerManager->asBinder(); 1002 binder->unlinkToDeath(mDeathRecipient); 1003 } 1004} 1005 1006void AudioFlinger::ThreadBase::exit() 1007{ 1008 // keep a strong ref on ourself so that we won't get 1009 // destroyed in the middle of requestExitAndWait() 1010 sp <ThreadBase> strongMe = this; 1011 1012 ALOGV("ThreadBase::exit"); 1013 { 1014 AutoMutex lock(&mLock); 1015 mExiting = true; 1016 requestExit(); 1017 mWaitWorkCV.signal(); 1018 } 1019 requestExitAndWait(); 1020} 1021 1022uint32_t AudioFlinger::ThreadBase::sampleRate() const 1023{ 1024 return mSampleRate; 1025} 1026 1027int AudioFlinger::ThreadBase::channelCount() const 1028{ 1029 return (int)mChannelCount; 1030} 1031 1032uint32_t AudioFlinger::ThreadBase::format() const 1033{ 1034 return mFormat; 1035} 1036 1037size_t AudioFlinger::ThreadBase::frameCount() const 1038{ 1039 return mFrameCount; 1040} 1041 1042status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1043{ 1044 status_t status; 1045 1046 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1047 Mutex::Autolock _l(mLock); 1048 1049 mNewParameters.add(keyValuePairs); 1050 mWaitWorkCV.signal(); 1051 // wait condition with timeout in case the thread loop has exited 1052 // before the request could be processed 1053 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1054 status = mParamStatus; 1055 mWaitWorkCV.signal(); 1056 } else { 1057 status = TIMED_OUT; 1058 } 1059 return status; 1060} 1061 1062void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1063{ 1064 Mutex::Autolock _l(mLock); 1065 sendConfigEvent_l(event, param); 1066} 1067 1068// sendConfigEvent_l() must be called with ThreadBase::mLock held 1069void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1070{ 1071 ConfigEvent configEvent; 1072 configEvent.mEvent = event; 1073 configEvent.mParam = param; 1074 mConfigEvents.add(configEvent); 1075 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1076 mWaitWorkCV.signal(); 1077} 1078 1079void AudioFlinger::ThreadBase::processConfigEvents() 1080{ 1081 mLock.lock(); 1082 while(!mConfigEvents.isEmpty()) { 1083 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1084 ConfigEvent configEvent = mConfigEvents[0]; 1085 mConfigEvents.removeAt(0); 1086 // release mLock before locking AudioFlinger mLock: lock order is always 1087 // AudioFlinger then ThreadBase to avoid cross deadlock 1088 mLock.unlock(); 1089 mAudioFlinger->mLock.lock(); 1090 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1091 mAudioFlinger->mLock.unlock(); 1092 mLock.lock(); 1093 } 1094 mLock.unlock(); 1095} 1096 1097status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1098{ 1099 const size_t SIZE = 256; 1100 char buffer[SIZE]; 1101 String8 result; 1102 1103 bool locked = tryLock(mLock); 1104 if (!locked) { 1105 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1106 write(fd, buffer, strlen(buffer)); 1107 } 1108 1109 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1110 result.append(buffer); 1111 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1112 result.append(buffer); 1113 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1114 result.append(buffer); 1115 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1116 result.append(buffer); 1117 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 1122 result.append(buffer); 1123 1124 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1125 result.append(buffer); 1126 result.append(" Index Command"); 1127 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1128 snprintf(buffer, SIZE, "\n %02d ", i); 1129 result.append(buffer); 1130 result.append(mNewParameters[i]); 1131 } 1132 1133 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1134 result.append(buffer); 1135 snprintf(buffer, SIZE, " Index event param\n"); 1136 result.append(buffer); 1137 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1138 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1139 result.append(buffer); 1140 } 1141 result.append("\n"); 1142 1143 write(fd, result.string(), result.size()); 1144 1145 if (locked) { 1146 mLock.unlock(); 1147 } 1148 return NO_ERROR; 1149} 1150 1151status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1152{ 1153 const size_t SIZE = 256; 1154 char buffer[SIZE]; 1155 String8 result; 1156 1157 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1158 write(fd, buffer, strlen(buffer)); 1159 1160 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1161 sp<EffectChain> chain = mEffectChains[i]; 1162 if (chain != 0) { 1163 chain->dump(fd, args); 1164 } 1165 } 1166 return NO_ERROR; 1167} 1168 1169void AudioFlinger::ThreadBase::acquireWakeLock() 1170{ 1171 Mutex::Autolock _l(mLock); 1172 acquireWakeLock_l(); 1173} 1174 1175void AudioFlinger::ThreadBase::acquireWakeLock_l() 1176{ 1177 if (mPowerManager == 0) { 1178 // use checkService() to avoid blocking if power service is not up yet 1179 sp<IBinder> binder = 1180 defaultServiceManager()->checkService(String16("power")); 1181 if (binder == 0) { 1182 LOGW("Thread %s cannot connect to the power manager service", mName); 1183 } else { 1184 mPowerManager = interface_cast<IPowerManager>(binder); 1185 binder->linkToDeath(mDeathRecipient); 1186 } 1187 } 1188 if (mPowerManager != 0) { 1189 sp<IBinder> binder = new BBinder(); 1190 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1191 binder, 1192 String16(mName)); 1193 if (status == NO_ERROR) { 1194 mWakeLockToken = binder; 1195 } 1196 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1197 } 1198} 1199 1200void AudioFlinger::ThreadBase::releaseWakeLock() 1201{ 1202 Mutex::Autolock _l(mLock); 1203 releaseWakeLock_l(); 1204} 1205 1206void AudioFlinger::ThreadBase::releaseWakeLock_l() 1207{ 1208 if (mWakeLockToken != 0) { 1209 ALOGV("releaseWakeLock_l() %s", mName); 1210 if (mPowerManager != 0) { 1211 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1212 } 1213 mWakeLockToken.clear(); 1214 } 1215} 1216 1217void AudioFlinger::ThreadBase::clearPowerManager() 1218{ 1219 Mutex::Autolock _l(mLock); 1220 releaseWakeLock_l(); 1221 mPowerManager.clear(); 1222} 1223 1224void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1225{ 1226 sp<ThreadBase> thread = mThread.promote(); 1227 if (thread != 0) { 1228 thread->clearPowerManager(); 1229 } 1230 LOGW("power manager service died !!!"); 1231} 1232 1233void AudioFlinger::ThreadBase::setEffectSuspended( 1234 const effect_uuid_t *type, bool suspend, int sessionId) 1235{ 1236 Mutex::Autolock _l(mLock); 1237 setEffectSuspended_l(type, suspend, sessionId); 1238} 1239 1240void AudioFlinger::ThreadBase::setEffectSuspended_l( 1241 const effect_uuid_t *type, bool suspend, int sessionId) 1242{ 1243 sp<EffectChain> chain; 1244 chain = getEffectChain_l(sessionId); 1245 if (chain != 0) { 1246 if (type != NULL) { 1247 chain->setEffectSuspended_l(type, suspend); 1248 } else { 1249 chain->setEffectSuspendedAll_l(suspend); 1250 } 1251 } 1252 1253 updateSuspendedSessions_l(type, suspend, sessionId); 1254} 1255 1256void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1257{ 1258 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1259 if (index < 0) { 1260 return; 1261 } 1262 1263 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1264 mSuspendedSessions.editValueAt(index); 1265 1266 for (size_t i = 0; i < sessionEffects.size(); i++) { 1267 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1268 for (int j = 0; j < desc->mRefCount; j++) { 1269 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1270 chain->setEffectSuspendedAll_l(true); 1271 } else { 1272 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1273 desc->mType.timeLow); 1274 chain->setEffectSuspended_l(&desc->mType, true); 1275 } 1276 } 1277 } 1278} 1279 1280void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1281 bool suspend, 1282 int sessionId) 1283{ 1284 int index = mSuspendedSessions.indexOfKey(sessionId); 1285 1286 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1287 1288 if (suspend) { 1289 if (index >= 0) { 1290 sessionEffects = mSuspendedSessions.editValueAt(index); 1291 } else { 1292 mSuspendedSessions.add(sessionId, sessionEffects); 1293 } 1294 } else { 1295 if (index < 0) { 1296 return; 1297 } 1298 sessionEffects = mSuspendedSessions.editValueAt(index); 1299 } 1300 1301 1302 int key = EffectChain::kKeyForSuspendAll; 1303 if (type != NULL) { 1304 key = type->timeLow; 1305 } 1306 index = sessionEffects.indexOfKey(key); 1307 1308 sp <SuspendedSessionDesc> desc; 1309 if (suspend) { 1310 if (index >= 0) { 1311 desc = sessionEffects.valueAt(index); 1312 } else { 1313 desc = new SuspendedSessionDesc(); 1314 if (type != NULL) { 1315 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1316 } 1317 sessionEffects.add(key, desc); 1318 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1319 } 1320 desc->mRefCount++; 1321 } else { 1322 if (index < 0) { 1323 return; 1324 } 1325 desc = sessionEffects.valueAt(index); 1326 if (--desc->mRefCount == 0) { 1327 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1328 sessionEffects.removeItemsAt(index); 1329 if (sessionEffects.isEmpty()) { 1330 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1331 sessionId); 1332 mSuspendedSessions.removeItem(sessionId); 1333 } 1334 } 1335 } 1336 if (!sessionEffects.isEmpty()) { 1337 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1338 } 1339} 1340 1341void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1342 bool enabled, 1343 int sessionId) 1344{ 1345 Mutex::Autolock _l(mLock); 1346 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1347} 1348 1349void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1350 bool enabled, 1351 int sessionId) 1352{ 1353 if (mType != RECORD) { 1354 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1355 // another session. This gives the priority to well behaved effect control panels 1356 // and applications not using global effects. 1357 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1358 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1359 } 1360 } 1361 1362 sp<EffectChain> chain = getEffectChain_l(sessionId); 1363 if (chain != 0) { 1364 chain->checkSuspendOnEffectEnabled(effect, enabled); 1365 } 1366} 1367 1368// ---------------------------------------------------------------------------- 1369 1370AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1371 AudioStreamOut* output, 1372 int id, 1373 uint32_t device) 1374 : ThreadBase(audioFlinger, id, device), 1375 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1376 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1377{ 1378 snprintf(mName, kNameLength, "AudioOut_%d", id); 1379 1380 readOutputParameters(); 1381 1382 mMasterVolume = mAudioFlinger->masterVolume(); 1383 mMasterMute = mAudioFlinger->masterMute(); 1384 1385 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 1386 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1387 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1388 mStreamTypes[stream].valid = true; 1389 } 1390} 1391 1392AudioFlinger::PlaybackThread::~PlaybackThread() 1393{ 1394 delete [] mMixBuffer; 1395} 1396 1397status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1398{ 1399 dumpInternals(fd, args); 1400 dumpTracks(fd, args); 1401 dumpEffectChains(fd, args); 1402 return NO_ERROR; 1403} 1404 1405status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1406{ 1407 const size_t SIZE = 256; 1408 char buffer[SIZE]; 1409 String8 result; 1410 1411 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1412 result.append(buffer); 1413 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1414 for (size_t i = 0; i < mTracks.size(); ++i) { 1415 sp<Track> track = mTracks[i]; 1416 if (track != 0) { 1417 track->dump(buffer, SIZE); 1418 result.append(buffer); 1419 } 1420 } 1421 1422 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1423 result.append(buffer); 1424 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1425 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1426 wp<Track> wTrack = mActiveTracks[i]; 1427 if (wTrack != 0) { 1428 sp<Track> track = wTrack.promote(); 1429 if (track != 0) { 1430 track->dump(buffer, SIZE); 1431 result.append(buffer); 1432 } 1433 } 1434 } 1435 write(fd, result.string(), result.size()); 1436 return NO_ERROR; 1437} 1438 1439status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1440{ 1441 const size_t SIZE = 256; 1442 char buffer[SIZE]; 1443 String8 result; 1444 1445 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1446 result.append(buffer); 1447 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1448 result.append(buffer); 1449 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1450 result.append(buffer); 1451 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1452 result.append(buffer); 1453 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1454 result.append(buffer); 1455 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1456 result.append(buffer); 1457 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1458 result.append(buffer); 1459 write(fd, result.string(), result.size()); 1460 1461 dumpBase(fd, args); 1462 1463 return NO_ERROR; 1464} 1465 1466// Thread virtuals 1467status_t AudioFlinger::PlaybackThread::readyToRun() 1468{ 1469 status_t status = initCheck(); 1470 if (status == NO_ERROR) { 1471 LOGI("AudioFlinger's thread %p ready to run", this); 1472 } else { 1473 LOGE("No working audio driver found."); 1474 } 1475 return status; 1476} 1477 1478void AudioFlinger::PlaybackThread::onFirstRef() 1479{ 1480 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1481} 1482 1483// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1484sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1485 const sp<AudioFlinger::Client>& client, 1486 int streamType, 1487 uint32_t sampleRate, 1488 uint32_t format, 1489 uint32_t channelMask, 1490 int frameCount, 1491 const sp<IMemory>& sharedBuffer, 1492 int sessionId, 1493 status_t *status) 1494{ 1495 sp<Track> track; 1496 status_t lStatus; 1497 1498 if (mType == DIRECT) { 1499 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1500 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1501 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1502 "for output %p with format %d", 1503 sampleRate, format, channelMask, mOutput, mFormat); 1504 lStatus = BAD_VALUE; 1505 goto Exit; 1506 } 1507 } 1508 } else { 1509 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1510 if (sampleRate > mSampleRate*2) { 1511 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1512 lStatus = BAD_VALUE; 1513 goto Exit; 1514 } 1515 } 1516 1517 lStatus = initCheck(); 1518 if (lStatus != NO_ERROR) { 1519 LOGE("Audio driver not initialized."); 1520 goto Exit; 1521 } 1522 1523 { // scope for mLock 1524 Mutex::Autolock _l(mLock); 1525 1526 // all tracks in same audio session must share the same routing strategy otherwise 1527 // conflicts will happen when tracks are moved from one output to another by audio policy 1528 // manager 1529 uint32_t strategy = 1530 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1531 for (size_t i = 0; i < mTracks.size(); ++i) { 1532 sp<Track> t = mTracks[i]; 1533 if (t != 0) { 1534 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1535 if (sessionId == t->sessionId() && strategy != actual) { 1536 LOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1537 strategy, actual); 1538 lStatus = BAD_VALUE; 1539 goto Exit; 1540 } 1541 } 1542 } 1543 1544 track = new Track(this, client, streamType, sampleRate, format, 1545 channelMask, frameCount, sharedBuffer, sessionId); 1546 if (track->getCblk() == NULL || track->name() < 0) { 1547 lStatus = NO_MEMORY; 1548 goto Exit; 1549 } 1550 mTracks.add(track); 1551 1552 sp<EffectChain> chain = getEffectChain_l(sessionId); 1553 if (chain != 0) { 1554 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1555 track->setMainBuffer(chain->inBuffer()); 1556 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1557 chain->incTrackCnt(); 1558 } 1559 1560 // invalidate track immediately if the stream type was moved to another thread since 1561 // createTrack() was called by the client process. 1562 if (!mStreamTypes[streamType].valid) { 1563 LOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1564 this, streamType); 1565 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1566 } 1567 } 1568 lStatus = NO_ERROR; 1569 1570Exit: 1571 if(status) { 1572 *status = lStatus; 1573 } 1574 return track; 1575} 1576 1577uint32_t AudioFlinger::PlaybackThread::latency() const 1578{ 1579 Mutex::Autolock _l(mLock); 1580 if (initCheck() == NO_ERROR) { 1581 return mOutput->stream->get_latency(mOutput->stream); 1582 } else { 1583 return 0; 1584 } 1585} 1586 1587status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1588{ 1589 mMasterVolume = value; 1590 return NO_ERROR; 1591} 1592 1593status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1594{ 1595 mMasterMute = muted; 1596 return NO_ERROR; 1597} 1598 1599float AudioFlinger::PlaybackThread::masterVolume() const 1600{ 1601 return mMasterVolume; 1602} 1603 1604bool AudioFlinger::PlaybackThread::masterMute() const 1605{ 1606 return mMasterMute; 1607} 1608 1609status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1610{ 1611 mStreamTypes[stream].volume = value; 1612 return NO_ERROR; 1613} 1614 1615status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1616{ 1617 mStreamTypes[stream].mute = muted; 1618 return NO_ERROR; 1619} 1620 1621float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1622{ 1623 return mStreamTypes[stream].volume; 1624} 1625 1626bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1627{ 1628 return mStreamTypes[stream].mute; 1629} 1630 1631// addTrack_l() must be called with ThreadBase::mLock held 1632status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1633{ 1634 status_t status = ALREADY_EXISTS; 1635 1636 // set retry count for buffer fill 1637 track->mRetryCount = kMaxTrackStartupRetries; 1638 if (mActiveTracks.indexOf(track) < 0) { 1639 // the track is newly added, make sure it fills up all its 1640 // buffers before playing. This is to ensure the client will 1641 // effectively get the latency it requested. 1642 track->mFillingUpStatus = Track::FS_FILLING; 1643 track->mResetDone = false; 1644 mActiveTracks.add(track); 1645 if (track->mainBuffer() != mMixBuffer) { 1646 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1647 if (chain != 0) { 1648 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1649 chain->incActiveTrackCnt(); 1650 } 1651 } 1652 1653 status = NO_ERROR; 1654 } 1655 1656 ALOGV("mWaitWorkCV.broadcast"); 1657 mWaitWorkCV.broadcast(); 1658 1659 return status; 1660} 1661 1662// destroyTrack_l() must be called with ThreadBase::mLock held 1663void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1664{ 1665 track->mState = TrackBase::TERMINATED; 1666 if (mActiveTracks.indexOf(track) < 0) { 1667 removeTrack_l(track); 1668 } 1669} 1670 1671void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1672{ 1673 mTracks.remove(track); 1674 deleteTrackName_l(track->name()); 1675 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1676 if (chain != 0) { 1677 chain->decTrackCnt(); 1678 } 1679} 1680 1681String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1682{ 1683 String8 out_s8 = String8(""); 1684 char *s; 1685 1686 Mutex::Autolock _l(mLock); 1687 if (initCheck() != NO_ERROR) { 1688 return out_s8; 1689 } 1690 1691 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1692 out_s8 = String8(s); 1693 free(s); 1694 return out_s8; 1695} 1696 1697// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1698void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1699 AudioSystem::OutputDescriptor desc; 1700 void *param2 = 0; 1701 1702 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1703 1704 switch (event) { 1705 case AudioSystem::OUTPUT_OPENED: 1706 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1707 desc.channels = mChannelMask; 1708 desc.samplingRate = mSampleRate; 1709 desc.format = mFormat; 1710 desc.frameCount = mFrameCount; 1711 desc.latency = latency(); 1712 param2 = &desc; 1713 break; 1714 1715 case AudioSystem::STREAM_CONFIG_CHANGED: 1716 param2 = ¶m; 1717 case AudioSystem::OUTPUT_CLOSED: 1718 default: 1719 break; 1720 } 1721 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1722} 1723 1724void AudioFlinger::PlaybackThread::readOutputParameters() 1725{ 1726 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1727 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1728 mChannelCount = (uint16_t)popcount(mChannelMask); 1729 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1730 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); 1731 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1732 1733 // FIXME - Current mixer implementation only supports stereo output: Always 1734 // Allocate a stereo buffer even if HW output is mono. 1735 if (mMixBuffer != NULL) delete[] mMixBuffer; 1736 mMixBuffer = new int16_t[mFrameCount * 2]; 1737 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1738 1739 // force reconfiguration of effect chains and engines to take new buffer size and audio 1740 // parameters into account 1741 // Note that mLock is not held when readOutputParameters() is called from the constructor 1742 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1743 // matter. 1744 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1745 Vector< sp<EffectChain> > effectChains = mEffectChains; 1746 for (size_t i = 0; i < effectChains.size(); i ++) { 1747 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1748 } 1749} 1750 1751status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1752{ 1753 if (halFrames == 0 || dspFrames == 0) { 1754 return BAD_VALUE; 1755 } 1756 Mutex::Autolock _l(mLock); 1757 if (initCheck() != NO_ERROR) { 1758 return INVALID_OPERATION; 1759 } 1760 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1761 1762 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1763} 1764 1765uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1766{ 1767 Mutex::Autolock _l(mLock); 1768 uint32_t result = 0; 1769 if (getEffectChain_l(sessionId) != 0) { 1770 result = EFFECT_SESSION; 1771 } 1772 1773 for (size_t i = 0; i < mTracks.size(); ++i) { 1774 sp<Track> track = mTracks[i]; 1775 if (sessionId == track->sessionId() && 1776 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1777 result |= TRACK_SESSION; 1778 break; 1779 } 1780 } 1781 1782 return result; 1783} 1784 1785uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1786{ 1787 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1788 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1789 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1790 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1791 } 1792 for (size_t i = 0; i < mTracks.size(); i++) { 1793 sp<Track> track = mTracks[i]; 1794 if (sessionId == track->sessionId() && 1795 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1796 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1797 } 1798 } 1799 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1800} 1801 1802 1803AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() 1804{ 1805 Mutex::Autolock _l(mLock); 1806 return mOutput; 1807} 1808 1809AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1810{ 1811 Mutex::Autolock _l(mLock); 1812 AudioStreamOut *output = mOutput; 1813 mOutput = NULL; 1814 return output; 1815} 1816 1817// this method must always be called either with ThreadBase mLock held or inside the thread loop 1818audio_stream_t* AudioFlinger::PlaybackThread::stream() 1819{ 1820 if (mOutput == NULL) { 1821 return NULL; 1822 } 1823 return &mOutput->stream->common; 1824} 1825 1826uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1827{ 1828 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1829 // decoding and transfer time. So sleeping for half of the latency would likely cause 1830 // underruns 1831 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1832 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1833 } else { 1834 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1835 } 1836} 1837 1838// ---------------------------------------------------------------------------- 1839 1840AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1841 : PlaybackThread(audioFlinger, output, id, device), 1842 mAudioMixer(NULL) 1843{ 1844 mType = ThreadBase::MIXER; 1845 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1846 1847 // FIXME - Current mixer implementation only supports stereo output 1848 if (mChannelCount == 1) { 1849 LOGE("Invalid audio hardware channel count"); 1850 } 1851} 1852 1853AudioFlinger::MixerThread::~MixerThread() 1854{ 1855 delete mAudioMixer; 1856} 1857 1858bool AudioFlinger::MixerThread::threadLoop() 1859{ 1860 Vector< sp<Track> > tracksToRemove; 1861 uint32_t mixerStatus = MIXER_IDLE; 1862 nsecs_t standbyTime = systemTime(); 1863 size_t mixBufferSize = mFrameCount * mFrameSize; 1864 // FIXME: Relaxed timing because of a certain device that can't meet latency 1865 // Should be reduced to 2x after the vendor fixes the driver issue 1866 // increase threshold again due to low power audio mode. The way this warning threshold is 1867 // calculated and its usefulness should be reconsidered anyway. 1868 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1869 nsecs_t lastWarning = 0; 1870 bool longStandbyExit = false; 1871 uint32_t activeSleepTime = activeSleepTimeUs(); 1872 uint32_t idleSleepTime = idleSleepTimeUs(); 1873 uint32_t sleepTime = idleSleepTime; 1874 uint32_t sleepTimeShift = 0; 1875 Vector< sp<EffectChain> > effectChains; 1876#ifdef DEBUG_CPU_USAGE 1877 ThreadCpuUsage cpu; 1878 const CentralTendencyStatistics& stats = cpu.statistics(); 1879#endif 1880 1881 acquireWakeLock(); 1882 1883 while (!exitPending()) 1884 { 1885#ifdef DEBUG_CPU_USAGE 1886 cpu.sampleAndEnable(); 1887 unsigned n = stats.n(); 1888 // cpu.elapsed() is expensive, so don't call it every loop 1889 if ((n & 127) == 1) { 1890 long long elapsed = cpu.elapsed(); 1891 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1892 double perLoop = elapsed / (double) n; 1893 double perLoop100 = perLoop * 0.01; 1894 double mean = stats.mean(); 1895 double stddev = stats.stddev(); 1896 double minimum = stats.minimum(); 1897 double maximum = stats.maximum(); 1898 cpu.resetStatistics(); 1899 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", 1900 elapsed * .000000001, n, perLoop * .000001, 1901 mean * .001, 1902 stddev * .001, 1903 minimum * .001, 1904 maximum * .001, 1905 mean / perLoop100, 1906 stddev / perLoop100, 1907 minimum / perLoop100, 1908 maximum / perLoop100); 1909 } 1910 } 1911#endif 1912 processConfigEvents(); 1913 1914 mixerStatus = MIXER_IDLE; 1915 { // scope for mLock 1916 1917 Mutex::Autolock _l(mLock); 1918 1919 if (checkForNewParameters_l()) { 1920 mixBufferSize = mFrameCount * mFrameSize; 1921 // FIXME: Relaxed timing because of a certain device that can't meet latency 1922 // Should be reduced to 2x after the vendor fixes the driver issue 1923 // increase threshold again due to low power audio mode. The way this warning 1924 // threshold is calculated and its usefulness should be reconsidered anyway. 1925 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1926 activeSleepTime = activeSleepTimeUs(); 1927 idleSleepTime = idleSleepTimeUs(); 1928 } 1929 1930 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1931 1932 // put audio hardware into standby after short delay 1933 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1934 mSuspended) { 1935 if (!mStandby) { 1936 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1937 mOutput->stream->common.standby(&mOutput->stream->common); 1938 mStandby = true; 1939 mBytesWritten = 0; 1940 } 1941 1942 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1943 // we're about to wait, flush the binder command buffer 1944 IPCThreadState::self()->flushCommands(); 1945 1946 if (exitPending()) break; 1947 1948 releaseWakeLock_l(); 1949 // wait until we have something to do... 1950 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1951 mWaitWorkCV.wait(mLock); 1952 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1953 acquireWakeLock_l(); 1954 1955 if (mMasterMute == false) { 1956 char value[PROPERTY_VALUE_MAX]; 1957 property_get("ro.audio.silent", value, "0"); 1958 if (atoi(value)) { 1959 ALOGD("Silence is golden"); 1960 setMasterMute(true); 1961 } 1962 } 1963 1964 standbyTime = systemTime() + kStandbyTimeInNsecs; 1965 sleepTime = idleSleepTime; 1966 sleepTimeShift = 0; 1967 continue; 1968 } 1969 } 1970 1971 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1972 1973 // prevent any changes in effect chain list and in each effect chain 1974 // during mixing and effect process as the audio buffers could be deleted 1975 // or modified if an effect is created or deleted 1976 lockEffectChains_l(effectChains); 1977 } 1978 1979 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1980 // mix buffers... 1981 mAudioMixer->process(); 1982 sleepTime = 0; 1983 // increase sleep time progressively when application underrun condition clears 1984 if (sleepTimeShift > 0) { 1985 sleepTimeShift--; 1986 } 1987 standbyTime = systemTime() + kStandbyTimeInNsecs; 1988 //TODO: delay standby when effects have a tail 1989 } else { 1990 // If no tracks are ready, sleep once for the duration of an output 1991 // buffer size, then write 0s to the output 1992 if (sleepTime == 0) { 1993 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1994 sleepTime = activeSleepTime >> sleepTimeShift; 1995 if (sleepTime < kMinThreadSleepTimeUs) { 1996 sleepTime = kMinThreadSleepTimeUs; 1997 } 1998 // reduce sleep time in case of consecutive application underruns to avoid 1999 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2000 // duration we would end up writing less data than needed by the audio HAL if 2001 // the condition persists. 2002 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2003 sleepTimeShift++; 2004 } 2005 } else { 2006 sleepTime = idleSleepTime; 2007 } 2008 } else if (mBytesWritten != 0 || 2009 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2010 memset (mMixBuffer, 0, mixBufferSize); 2011 sleepTime = 0; 2012 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2013 } 2014 // TODO add standby time extension fct of effect tail 2015 } 2016 2017 if (mSuspended) { 2018 sleepTime = suspendSleepTimeUs(); 2019 } 2020 // sleepTime == 0 means we must write to audio hardware 2021 if (sleepTime == 0) { 2022 for (size_t i = 0; i < effectChains.size(); i ++) { 2023 effectChains[i]->process_l(); 2024 } 2025 // enable changes in effect chain 2026 unlockEffectChains(effectChains); 2027 mLastWriteTime = systemTime(); 2028 mInWrite = true; 2029 mBytesWritten += mixBufferSize; 2030 2031 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2032 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2033 mNumWrites++; 2034 mInWrite = false; 2035 nsecs_t now = systemTime(); 2036 nsecs_t delta = now - mLastWriteTime; 2037 if (!mStandby && delta > maxPeriod) { 2038 mNumDelayedWrites++; 2039 if ((now - lastWarning) > kWarningThrottleNs) { 2040 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2041 ns2ms(delta), mNumDelayedWrites, this); 2042 lastWarning = now; 2043 } 2044 if (mStandby) { 2045 longStandbyExit = true; 2046 } 2047 } 2048 mStandby = false; 2049 } else { 2050 // enable changes in effect chain 2051 unlockEffectChains(effectChains); 2052 usleep(sleepTime); 2053 } 2054 2055 // finally let go of all our tracks, without the lock held 2056 // since we can't guarantee the destructors won't acquire that 2057 // same lock. 2058 tracksToRemove.clear(); 2059 2060 // Effect chains will be actually deleted here if they were removed from 2061 // mEffectChains list during mixing or effects processing 2062 effectChains.clear(); 2063 } 2064 2065 if (!mStandby) { 2066 mOutput->stream->common.standby(&mOutput->stream->common); 2067 } 2068 2069 releaseWakeLock(); 2070 2071 ALOGV("MixerThread %p exiting", this); 2072 return false; 2073} 2074 2075// prepareTracks_l() must be called with ThreadBase::mLock held 2076uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2077{ 2078 2079 uint32_t mixerStatus = MIXER_IDLE; 2080 // find out which tracks need to be processed 2081 size_t count = activeTracks.size(); 2082 size_t mixedTracks = 0; 2083 size_t tracksWithEffect = 0; 2084 2085 float masterVolume = mMasterVolume; 2086 bool masterMute = mMasterMute; 2087 2088 if (masterMute) { 2089 masterVolume = 0; 2090 } 2091 // Delegate master volume control to effect in output mix effect chain if needed 2092 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2093 if (chain != 0) { 2094 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2095 chain->setVolume_l(&v, &v); 2096 masterVolume = (float)((v + (1 << 23)) >> 24); 2097 chain.clear(); 2098 } 2099 2100 for (size_t i=0 ; i<count ; i++) { 2101 sp<Track> t = activeTracks[i].promote(); 2102 if (t == 0) continue; 2103 2104 Track* const track = t.get(); 2105 audio_track_cblk_t* cblk = track->cblk(); 2106 2107 // The first time a track is added we wait 2108 // for all its buffers to be filled before processing it 2109 mAudioMixer->setActiveTrack(track->name()); 2110 // make sure that we have enough frames to mix one full buffer. 2111 // enforce this condition only once to enable draining the buffer in case the client 2112 // app does not call stop() and relies on underrun to stop: 2113 // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed 2114 // during last round 2115 uint32_t minFrames = 1; 2116 if (!track->isStopped() && !track->isPausing() && 2117 (track->mRetryCount >= kMaxTrackRetries)) { 2118 if (t->sampleRate() == (int)mSampleRate) { 2119 minFrames = mFrameCount; 2120 } else { 2121 // +1 for rounding and +1 for additional sample needed for interpolation 2122 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2123 // add frames already consumed but not yet released by the resampler 2124 // because cblk->framesReady() will include these frames 2125 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2126 // the minimum track buffer size is normally twice the number of frames necessary 2127 // to fill one buffer and the resampler should not leave more than one buffer worth 2128 // of unreleased frames after each pass, but just in case... 2129 LOG_ASSERT(minFrames <= cblk->frameCount); 2130 } 2131 } 2132 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2133 !track->isPaused() && !track->isTerminated()) 2134 { 2135 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 2136 2137 mixedTracks++; 2138 2139 // track->mainBuffer() != mMixBuffer means there is an effect chain 2140 // connected to the track 2141 chain.clear(); 2142 if (track->mainBuffer() != mMixBuffer) { 2143 chain = getEffectChain_l(track->sessionId()); 2144 // Delegate volume control to effect in track effect chain if needed 2145 if (chain != 0) { 2146 tracksWithEffect++; 2147 } else { 2148 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 2149 track->name(), track->sessionId()); 2150 } 2151 } 2152 2153 2154 int param = AudioMixer::VOLUME; 2155 if (track->mFillingUpStatus == Track::FS_FILLED) { 2156 // no ramp for the first volume setting 2157 track->mFillingUpStatus = Track::FS_ACTIVE; 2158 if (track->mState == TrackBase::RESUMING) { 2159 track->mState = TrackBase::ACTIVE; 2160 param = AudioMixer::RAMP_VOLUME; 2161 } 2162 mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2163 } else if (cblk->server != 0) { 2164 // If the track is stopped before the first frame was mixed, 2165 // do not apply ramp 2166 param = AudioMixer::RAMP_VOLUME; 2167 } 2168 2169 // compute volume for this track 2170 uint32_t vl, vr, va; 2171 if (track->isMuted() || track->isPausing() || 2172 mStreamTypes[track->type()].mute) { 2173 vl = vr = va = 0; 2174 if (track->isPausing()) { 2175 track->setPaused(); 2176 } 2177 } else { 2178 2179 // read original volumes with volume control 2180 float typeVolume = mStreamTypes[track->type()].volume; 2181 float v = masterVolume * typeVolume; 2182 vl = (uint32_t)(v * cblk->volume[0]) << 12; 2183 vr = (uint32_t)(v * cblk->volume[1]) << 12; 2184 2185 va = (uint32_t)(v * cblk->sendLevel); 2186 } 2187 // Delegate volume control to effect in track effect chain if needed 2188 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2189 // Do not ramp volume if volume is controlled by effect 2190 param = AudioMixer::VOLUME; 2191 track->mHasVolumeController = true; 2192 } else { 2193 // force no volume ramp when volume controller was just disabled or removed 2194 // from effect chain to avoid volume spike 2195 if (track->mHasVolumeController) { 2196 param = AudioMixer::VOLUME; 2197 } 2198 track->mHasVolumeController = false; 2199 } 2200 2201 // Convert volumes from 8.24 to 4.12 format 2202 int16_t left, right, aux; 2203 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2204 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2205 left = int16_t(v_clamped); 2206 v_clamped = (vr + (1 << 11)) >> 12; 2207 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2208 right = int16_t(v_clamped); 2209 2210 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2211 aux = int16_t(va); 2212 2213 // XXX: these things DON'T need to be done each time 2214 mAudioMixer->setBufferProvider(track); 2215 mAudioMixer->enable(); 2216 2217 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 2218 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 2219 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 2220 mAudioMixer->setParameter( 2221 AudioMixer::TRACK, 2222 AudioMixer::FORMAT, (void *)track->format()); 2223 mAudioMixer->setParameter( 2224 AudioMixer::TRACK, 2225 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2226 mAudioMixer->setParameter( 2227 AudioMixer::RESAMPLE, 2228 AudioMixer::SAMPLE_RATE, 2229 (void *)(cblk->sampleRate)); 2230 mAudioMixer->setParameter( 2231 AudioMixer::TRACK, 2232 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2233 mAudioMixer->setParameter( 2234 AudioMixer::TRACK, 2235 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2236 2237 // reset retry count 2238 track->mRetryCount = kMaxTrackRetries; 2239 mixerStatus = MIXER_TRACKS_READY; 2240 } else { 2241 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 2242 if (track->isStopped()) { 2243 track->reset(); 2244 } 2245 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2246 // We have consumed all the buffers of this track. 2247 // Remove it from the list of active tracks. 2248 tracksToRemove->add(track); 2249 } else { 2250 // No buffers for this track. Give it a few chances to 2251 // fill a buffer, then remove it from active list. 2252 if (--(track->mRetryCount) <= 0) { 2253 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 2254 tracksToRemove->add(track); 2255 // indicate to client process that the track was disabled because of underrun 2256 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2257 } else if (mixerStatus != MIXER_TRACKS_READY) { 2258 mixerStatus = MIXER_TRACKS_ENABLED; 2259 } 2260 } 2261 mAudioMixer->disable(); 2262 } 2263 } 2264 2265 // remove all the tracks that need to be... 2266 count = tracksToRemove->size(); 2267 if (UNLIKELY(count)) { 2268 for (size_t i=0 ; i<count ; i++) { 2269 const sp<Track>& track = tracksToRemove->itemAt(i); 2270 mActiveTracks.remove(track); 2271 if (track->mainBuffer() != mMixBuffer) { 2272 chain = getEffectChain_l(track->sessionId()); 2273 if (chain != 0) { 2274 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2275 chain->decActiveTrackCnt(); 2276 } 2277 } 2278 if (track->isTerminated()) { 2279 removeTrack_l(track); 2280 } 2281 } 2282 } 2283 2284 // mix buffer must be cleared if all tracks are connected to an 2285 // effect chain as in this case the mixer will not write to 2286 // mix buffer and track effects will accumulate into it 2287 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2288 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2289 } 2290 2291 return mixerStatus; 2292} 2293 2294void AudioFlinger::MixerThread::invalidateTracks(int streamType) 2295{ 2296 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2297 this, streamType, mTracks.size()); 2298 Mutex::Autolock _l(mLock); 2299 2300 size_t size = mTracks.size(); 2301 for (size_t i = 0; i < size; i++) { 2302 sp<Track> t = mTracks[i]; 2303 if (t->type() == streamType) { 2304 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2305 t->mCblk->cv.signal(); 2306 } 2307 } 2308} 2309 2310void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid) 2311{ 2312 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2313 this, streamType, valid); 2314 Mutex::Autolock _l(mLock); 2315 2316 mStreamTypes[streamType].valid = valid; 2317} 2318 2319// getTrackName_l() must be called with ThreadBase::mLock held 2320int AudioFlinger::MixerThread::getTrackName_l() 2321{ 2322 return mAudioMixer->getTrackName(); 2323} 2324 2325// deleteTrackName_l() must be called with ThreadBase::mLock held 2326void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2327{ 2328 ALOGV("remove track (%d) and delete from mixer", name); 2329 mAudioMixer->deleteTrackName(name); 2330} 2331 2332// checkForNewParameters_l() must be called with ThreadBase::mLock held 2333bool AudioFlinger::MixerThread::checkForNewParameters_l() 2334{ 2335 bool reconfig = false; 2336 2337 while (!mNewParameters.isEmpty()) { 2338 status_t status = NO_ERROR; 2339 String8 keyValuePair = mNewParameters[0]; 2340 AudioParameter param = AudioParameter(keyValuePair); 2341 int value; 2342 2343 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2344 reconfig = true; 2345 } 2346 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2347 if (value != AUDIO_FORMAT_PCM_16_BIT) { 2348 status = BAD_VALUE; 2349 } else { 2350 reconfig = true; 2351 } 2352 } 2353 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2354 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2355 status = BAD_VALUE; 2356 } else { 2357 reconfig = true; 2358 } 2359 } 2360 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2361 // do not accept frame count changes if tracks are open as the track buffer 2362 // size depends on frame count and correct behavior would not be guaranteed 2363 // if frame count is changed after track creation 2364 if (!mTracks.isEmpty()) { 2365 status = INVALID_OPERATION; 2366 } else { 2367 reconfig = true; 2368 } 2369 } 2370 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2371 // when changing the audio output device, call addBatteryData to notify 2372 // the change 2373 if ((int)mDevice != value) { 2374 uint32_t params = 0; 2375 // check whether speaker is on 2376 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2377 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2378 } 2379 2380 int deviceWithoutSpeaker 2381 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2382 // check if any other device (except speaker) is on 2383 if (value & deviceWithoutSpeaker ) { 2384 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2385 } 2386 2387 if (params != 0) { 2388 addBatteryData(params); 2389 } 2390 } 2391 2392 // forward device change to effects that have requested to be 2393 // aware of attached audio device. 2394 mDevice = (uint32_t)value; 2395 for (size_t i = 0; i < mEffectChains.size(); i++) { 2396 mEffectChains[i]->setDevice_l(mDevice); 2397 } 2398 } 2399 2400 if (status == NO_ERROR) { 2401 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2402 keyValuePair.string()); 2403 if (!mStandby && status == INVALID_OPERATION) { 2404 mOutput->stream->common.standby(&mOutput->stream->common); 2405 mStandby = true; 2406 mBytesWritten = 0; 2407 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2408 keyValuePair.string()); 2409 } 2410 if (status == NO_ERROR && reconfig) { 2411 delete mAudioMixer; 2412 readOutputParameters(); 2413 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2414 for (size_t i = 0; i < mTracks.size() ; i++) { 2415 int name = getTrackName_l(); 2416 if (name < 0) break; 2417 mTracks[i]->mName = name; 2418 // limit track sample rate to 2 x new output sample rate 2419 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2420 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2421 } 2422 } 2423 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2424 } 2425 } 2426 2427 mNewParameters.removeAt(0); 2428 2429 mParamStatus = status; 2430 mParamCond.signal(); 2431 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2432 // already timed out waiting for the status and will never signal the condition. 2433 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2434 } 2435 return reconfig; 2436} 2437 2438status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2439{ 2440 const size_t SIZE = 256; 2441 char buffer[SIZE]; 2442 String8 result; 2443 2444 PlaybackThread::dumpInternals(fd, args); 2445 2446 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2447 result.append(buffer); 2448 write(fd, result.string(), result.size()); 2449 return NO_ERROR; 2450} 2451 2452uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2453{ 2454 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2455} 2456 2457uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2458{ 2459 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2460} 2461 2462// ---------------------------------------------------------------------------- 2463AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2464 : PlaybackThread(audioFlinger, output, id, device) 2465{ 2466 mType = ThreadBase::DIRECT; 2467} 2468 2469AudioFlinger::DirectOutputThread::~DirectOutputThread() 2470{ 2471} 2472 2473static inline 2474int32_t mul(int16_t in, int16_t v) 2475{ 2476#if defined(__arm__) && !defined(__thumb__) 2477 int32_t out; 2478 asm( "smulbb %[out], %[in], %[v] \n" 2479 : [out]"=r"(out) 2480 : [in]"%r"(in), [v]"r"(v) 2481 : ); 2482 return out; 2483#else 2484 return in * int32_t(v); 2485#endif 2486} 2487 2488void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2489{ 2490 // Do not apply volume on compressed audio 2491 if (!audio_is_linear_pcm(mFormat)) { 2492 return; 2493 } 2494 2495 // convert to signed 16 bit before volume calculation 2496 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2497 size_t count = mFrameCount * mChannelCount; 2498 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2499 int16_t *dst = mMixBuffer + count-1; 2500 while(count--) { 2501 *dst-- = (int16_t)(*src--^0x80) << 8; 2502 } 2503 } 2504 2505 size_t frameCount = mFrameCount; 2506 int16_t *out = mMixBuffer; 2507 if (ramp) { 2508 if (mChannelCount == 1) { 2509 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2510 int32_t vlInc = d / (int32_t)frameCount; 2511 int32_t vl = ((int32_t)mLeftVolShort << 16); 2512 do { 2513 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2514 out++; 2515 vl += vlInc; 2516 } while (--frameCount); 2517 2518 } else { 2519 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2520 int32_t vlInc = d / (int32_t)frameCount; 2521 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2522 int32_t vrInc = d / (int32_t)frameCount; 2523 int32_t vl = ((int32_t)mLeftVolShort << 16); 2524 int32_t vr = ((int32_t)mRightVolShort << 16); 2525 do { 2526 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2527 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2528 out += 2; 2529 vl += vlInc; 2530 vr += vrInc; 2531 } while (--frameCount); 2532 } 2533 } else { 2534 if (mChannelCount == 1) { 2535 do { 2536 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2537 out++; 2538 } while (--frameCount); 2539 } else { 2540 do { 2541 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2542 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2543 out += 2; 2544 } while (--frameCount); 2545 } 2546 } 2547 2548 // convert back to unsigned 8 bit after volume calculation 2549 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2550 size_t count = mFrameCount * mChannelCount; 2551 int16_t *src = mMixBuffer; 2552 uint8_t *dst = (uint8_t *)mMixBuffer; 2553 while(count--) { 2554 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2555 } 2556 } 2557 2558 mLeftVolShort = leftVol; 2559 mRightVolShort = rightVol; 2560} 2561 2562bool AudioFlinger::DirectOutputThread::threadLoop() 2563{ 2564 uint32_t mixerStatus = MIXER_IDLE; 2565 sp<Track> trackToRemove; 2566 sp<Track> activeTrack; 2567 nsecs_t standbyTime = systemTime(); 2568 int8_t *curBuf; 2569 size_t mixBufferSize = mFrameCount*mFrameSize; 2570 uint32_t activeSleepTime = activeSleepTimeUs(); 2571 uint32_t idleSleepTime = idleSleepTimeUs(); 2572 uint32_t sleepTime = idleSleepTime; 2573 // use shorter standby delay as on normal output to release 2574 // hardware resources as soon as possible 2575 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2576 2577 acquireWakeLock(); 2578 2579 while (!exitPending()) 2580 { 2581 bool rampVolume; 2582 uint16_t leftVol; 2583 uint16_t rightVol; 2584 Vector< sp<EffectChain> > effectChains; 2585 2586 processConfigEvents(); 2587 2588 mixerStatus = MIXER_IDLE; 2589 2590 { // scope for the mLock 2591 2592 Mutex::Autolock _l(mLock); 2593 2594 if (checkForNewParameters_l()) { 2595 mixBufferSize = mFrameCount*mFrameSize; 2596 activeSleepTime = activeSleepTimeUs(); 2597 idleSleepTime = idleSleepTimeUs(); 2598 standbyDelay = microseconds(activeSleepTime*2); 2599 } 2600 2601 // put audio hardware into standby after short delay 2602 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2603 mSuspended) { 2604 // wait until we have something to do... 2605 if (!mStandby) { 2606 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2607 mOutput->stream->common.standby(&mOutput->stream->common); 2608 mStandby = true; 2609 mBytesWritten = 0; 2610 } 2611 2612 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2613 // we're about to wait, flush the binder command buffer 2614 IPCThreadState::self()->flushCommands(); 2615 2616 if (exitPending()) break; 2617 2618 releaseWakeLock_l(); 2619 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2620 mWaitWorkCV.wait(mLock); 2621 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2622 acquireWakeLock_l(); 2623 2624 if (mMasterMute == false) { 2625 char value[PROPERTY_VALUE_MAX]; 2626 property_get("ro.audio.silent", value, "0"); 2627 if (atoi(value)) { 2628 ALOGD("Silence is golden"); 2629 setMasterMute(true); 2630 } 2631 } 2632 2633 standbyTime = systemTime() + standbyDelay; 2634 sleepTime = idleSleepTime; 2635 continue; 2636 } 2637 } 2638 2639 effectChains = mEffectChains; 2640 2641 // find out which tracks need to be processed 2642 if (mActiveTracks.size() != 0) { 2643 sp<Track> t = mActiveTracks[0].promote(); 2644 if (t == 0) continue; 2645 2646 Track* const track = t.get(); 2647 audio_track_cblk_t* cblk = track->cblk(); 2648 2649 // The first time a track is added we wait 2650 // for all its buffers to be filled before processing it 2651 if (cblk->framesReady() && track->isReady() && 2652 !track->isPaused() && !track->isTerminated()) 2653 { 2654 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2655 2656 if (track->mFillingUpStatus == Track::FS_FILLED) { 2657 track->mFillingUpStatus = Track::FS_ACTIVE; 2658 mLeftVolFloat = mRightVolFloat = 0; 2659 mLeftVolShort = mRightVolShort = 0; 2660 if (track->mState == TrackBase::RESUMING) { 2661 track->mState = TrackBase::ACTIVE; 2662 rampVolume = true; 2663 } 2664 } else if (cblk->server != 0) { 2665 // If the track is stopped before the first frame was mixed, 2666 // do not apply ramp 2667 rampVolume = true; 2668 } 2669 // compute volume for this track 2670 float left, right; 2671 if (track->isMuted() || mMasterMute || track->isPausing() || 2672 mStreamTypes[track->type()].mute) { 2673 left = right = 0; 2674 if (track->isPausing()) { 2675 track->setPaused(); 2676 } 2677 } else { 2678 float typeVolume = mStreamTypes[track->type()].volume; 2679 float v = mMasterVolume * typeVolume; 2680 float v_clamped = v * cblk->volume[0]; 2681 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2682 left = v_clamped/MAX_GAIN; 2683 v_clamped = v * cblk->volume[1]; 2684 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2685 right = v_clamped/MAX_GAIN; 2686 } 2687 2688 if (left != mLeftVolFloat || right != mRightVolFloat) { 2689 mLeftVolFloat = left; 2690 mRightVolFloat = right; 2691 2692 // If audio HAL implements volume control, 2693 // force software volume to nominal value 2694 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2695 left = 1.0f; 2696 right = 1.0f; 2697 } 2698 2699 // Convert volumes from float to 8.24 2700 uint32_t vl = (uint32_t)(left * (1 << 24)); 2701 uint32_t vr = (uint32_t)(right * (1 << 24)); 2702 2703 // Delegate volume control to effect in track effect chain if needed 2704 // only one effect chain can be present on DirectOutputThread, so if 2705 // there is one, the track is connected to it 2706 if (!effectChains.isEmpty()) { 2707 // Do not ramp volume if volume is controlled by effect 2708 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2709 rampVolume = false; 2710 } 2711 } 2712 2713 // Convert volumes from 8.24 to 4.12 format 2714 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2715 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2716 leftVol = (uint16_t)v_clamped; 2717 v_clamped = (vr + (1 << 11)) >> 12; 2718 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2719 rightVol = (uint16_t)v_clamped; 2720 } else { 2721 leftVol = mLeftVolShort; 2722 rightVol = mRightVolShort; 2723 rampVolume = false; 2724 } 2725 2726 // reset retry count 2727 track->mRetryCount = kMaxTrackRetriesDirect; 2728 activeTrack = t; 2729 mixerStatus = MIXER_TRACKS_READY; 2730 } else { 2731 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2732 if (track->isStopped()) { 2733 track->reset(); 2734 } 2735 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2736 // We have consumed all the buffers of this track. 2737 // Remove it from the list of active tracks. 2738 trackToRemove = track; 2739 } else { 2740 // No buffers for this track. Give it a few chances to 2741 // fill a buffer, then remove it from active list. 2742 if (--(track->mRetryCount) <= 0) { 2743 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2744 trackToRemove = track; 2745 } else { 2746 mixerStatus = MIXER_TRACKS_ENABLED; 2747 } 2748 } 2749 } 2750 } 2751 2752 // remove all the tracks that need to be... 2753 if (UNLIKELY(trackToRemove != 0)) { 2754 mActiveTracks.remove(trackToRemove); 2755 if (!effectChains.isEmpty()) { 2756 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2757 trackToRemove->sessionId()); 2758 effectChains[0]->decActiveTrackCnt(); 2759 } 2760 if (trackToRemove->isTerminated()) { 2761 removeTrack_l(trackToRemove); 2762 } 2763 } 2764 2765 lockEffectChains_l(effectChains); 2766 } 2767 2768 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2769 AudioBufferProvider::Buffer buffer; 2770 size_t frameCount = mFrameCount; 2771 curBuf = (int8_t *)mMixBuffer; 2772 // output audio to hardware 2773 while (frameCount) { 2774 buffer.frameCount = frameCount; 2775 activeTrack->getNextBuffer(&buffer); 2776 if (UNLIKELY(buffer.raw == NULL)) { 2777 memset(curBuf, 0, frameCount * mFrameSize); 2778 break; 2779 } 2780 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2781 frameCount -= buffer.frameCount; 2782 curBuf += buffer.frameCount * mFrameSize; 2783 activeTrack->releaseBuffer(&buffer); 2784 } 2785 sleepTime = 0; 2786 standbyTime = systemTime() + standbyDelay; 2787 } else { 2788 if (sleepTime == 0) { 2789 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2790 sleepTime = activeSleepTime; 2791 } else { 2792 sleepTime = idleSleepTime; 2793 } 2794 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2795 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2796 sleepTime = 0; 2797 } 2798 } 2799 2800 if (mSuspended) { 2801 sleepTime = suspendSleepTimeUs(); 2802 } 2803 // sleepTime == 0 means we must write to audio hardware 2804 if (sleepTime == 0) { 2805 if (mixerStatus == MIXER_TRACKS_READY) { 2806 applyVolume(leftVol, rightVol, rampVolume); 2807 } 2808 for (size_t i = 0; i < effectChains.size(); i ++) { 2809 effectChains[i]->process_l(); 2810 } 2811 unlockEffectChains(effectChains); 2812 2813 mLastWriteTime = systemTime(); 2814 mInWrite = true; 2815 mBytesWritten += mixBufferSize; 2816 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2817 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2818 mNumWrites++; 2819 mInWrite = false; 2820 mStandby = false; 2821 } else { 2822 unlockEffectChains(effectChains); 2823 usleep(sleepTime); 2824 } 2825 2826 // finally let go of removed track, without the lock held 2827 // since we can't guarantee the destructors won't acquire that 2828 // same lock. 2829 trackToRemove.clear(); 2830 activeTrack.clear(); 2831 2832 // Effect chains will be actually deleted here if they were removed from 2833 // mEffectChains list during mixing or effects processing 2834 effectChains.clear(); 2835 } 2836 2837 if (!mStandby) { 2838 mOutput->stream->common.standby(&mOutput->stream->common); 2839 } 2840 2841 releaseWakeLock(); 2842 2843 ALOGV("DirectOutputThread %p exiting", this); 2844 return false; 2845} 2846 2847// getTrackName_l() must be called with ThreadBase::mLock held 2848int AudioFlinger::DirectOutputThread::getTrackName_l() 2849{ 2850 return 0; 2851} 2852 2853// deleteTrackName_l() must be called with ThreadBase::mLock held 2854void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2855{ 2856} 2857 2858// checkForNewParameters_l() must be called with ThreadBase::mLock held 2859bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2860{ 2861 bool reconfig = false; 2862 2863 while (!mNewParameters.isEmpty()) { 2864 status_t status = NO_ERROR; 2865 String8 keyValuePair = mNewParameters[0]; 2866 AudioParameter param = AudioParameter(keyValuePair); 2867 int value; 2868 2869 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2870 // do not accept frame count changes if tracks are open as the track buffer 2871 // size depends on frame count and correct behavior would not be garantied 2872 // if frame count is changed after track creation 2873 if (!mTracks.isEmpty()) { 2874 status = INVALID_OPERATION; 2875 } else { 2876 reconfig = true; 2877 } 2878 } 2879 if (status == NO_ERROR) { 2880 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2881 keyValuePair.string()); 2882 if (!mStandby && status == INVALID_OPERATION) { 2883 mOutput->stream->common.standby(&mOutput->stream->common); 2884 mStandby = true; 2885 mBytesWritten = 0; 2886 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2887 keyValuePair.string()); 2888 } 2889 if (status == NO_ERROR && reconfig) { 2890 readOutputParameters(); 2891 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2892 } 2893 } 2894 2895 mNewParameters.removeAt(0); 2896 2897 mParamStatus = status; 2898 mParamCond.signal(); 2899 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2900 // already timed out waiting for the status and will never signal the condition. 2901 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2902 } 2903 return reconfig; 2904} 2905 2906uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2907{ 2908 uint32_t time; 2909 if (audio_is_linear_pcm(mFormat)) { 2910 time = PlaybackThread::activeSleepTimeUs(); 2911 } else { 2912 time = 10000; 2913 } 2914 return time; 2915} 2916 2917uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2918{ 2919 uint32_t time; 2920 if (audio_is_linear_pcm(mFormat)) { 2921 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2922 } else { 2923 time = 10000; 2924 } 2925 return time; 2926} 2927 2928uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2929{ 2930 uint32_t time; 2931 if (audio_is_linear_pcm(mFormat)) { 2932 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2933 } else { 2934 time = 10000; 2935 } 2936 return time; 2937} 2938 2939 2940// ---------------------------------------------------------------------------- 2941 2942AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2943 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2944{ 2945 mType = ThreadBase::DUPLICATING; 2946 addOutputTrack(mainThread); 2947} 2948 2949AudioFlinger::DuplicatingThread::~DuplicatingThread() 2950{ 2951 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2952 mOutputTracks[i]->destroy(); 2953 } 2954 mOutputTracks.clear(); 2955} 2956 2957bool AudioFlinger::DuplicatingThread::threadLoop() 2958{ 2959 Vector< sp<Track> > tracksToRemove; 2960 uint32_t mixerStatus = MIXER_IDLE; 2961 nsecs_t standbyTime = systemTime(); 2962 size_t mixBufferSize = mFrameCount*mFrameSize; 2963 SortedVector< sp<OutputTrack> > outputTracks; 2964 uint32_t writeFrames = 0; 2965 uint32_t activeSleepTime = activeSleepTimeUs(); 2966 uint32_t idleSleepTime = idleSleepTimeUs(); 2967 uint32_t sleepTime = idleSleepTime; 2968 Vector< sp<EffectChain> > effectChains; 2969 2970 acquireWakeLock(); 2971 2972 while (!exitPending()) 2973 { 2974 processConfigEvents(); 2975 2976 mixerStatus = MIXER_IDLE; 2977 { // scope for the mLock 2978 2979 Mutex::Autolock _l(mLock); 2980 2981 if (checkForNewParameters_l()) { 2982 mixBufferSize = mFrameCount*mFrameSize; 2983 updateWaitTime(); 2984 activeSleepTime = activeSleepTimeUs(); 2985 idleSleepTime = idleSleepTimeUs(); 2986 } 2987 2988 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2989 2990 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2991 outputTracks.add(mOutputTracks[i]); 2992 } 2993 2994 // put audio hardware into standby after short delay 2995 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2996 mSuspended) { 2997 if (!mStandby) { 2998 for (size_t i = 0; i < outputTracks.size(); i++) { 2999 outputTracks[i]->stop(); 3000 } 3001 mStandby = true; 3002 mBytesWritten = 0; 3003 } 3004 3005 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3006 // we're about to wait, flush the binder command buffer 3007 IPCThreadState::self()->flushCommands(); 3008 outputTracks.clear(); 3009 3010 if (exitPending()) break; 3011 3012 releaseWakeLock_l(); 3013 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3014 mWaitWorkCV.wait(mLock); 3015 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3016 acquireWakeLock_l(); 3017 3018 if (mMasterMute == false) { 3019 char value[PROPERTY_VALUE_MAX]; 3020 property_get("ro.audio.silent", value, "0"); 3021 if (atoi(value)) { 3022 ALOGD("Silence is golden"); 3023 setMasterMute(true); 3024 } 3025 } 3026 3027 standbyTime = systemTime() + kStandbyTimeInNsecs; 3028 sleepTime = idleSleepTime; 3029 continue; 3030 } 3031 } 3032 3033 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3034 3035 // prevent any changes in effect chain list and in each effect chain 3036 // during mixing and effect process as the audio buffers could be deleted 3037 // or modified if an effect is created or deleted 3038 lockEffectChains_l(effectChains); 3039 } 3040 3041 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3042 // mix buffers... 3043 if (outputsReady(outputTracks)) { 3044 mAudioMixer->process(); 3045 } else { 3046 memset(mMixBuffer, 0, mixBufferSize); 3047 } 3048 sleepTime = 0; 3049 writeFrames = mFrameCount; 3050 } else { 3051 if (sleepTime == 0) { 3052 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3053 sleepTime = activeSleepTime; 3054 } else { 3055 sleepTime = idleSleepTime; 3056 } 3057 } else if (mBytesWritten != 0) { 3058 // flush remaining overflow buffers in output tracks 3059 for (size_t i = 0; i < outputTracks.size(); i++) { 3060 if (outputTracks[i]->isActive()) { 3061 sleepTime = 0; 3062 writeFrames = 0; 3063 memset(mMixBuffer, 0, mixBufferSize); 3064 break; 3065 } 3066 } 3067 } 3068 } 3069 3070 if (mSuspended) { 3071 sleepTime = suspendSleepTimeUs(); 3072 } 3073 // sleepTime == 0 means we must write to audio hardware 3074 if (sleepTime == 0) { 3075 for (size_t i = 0; i < effectChains.size(); i ++) { 3076 effectChains[i]->process_l(); 3077 } 3078 // enable changes in effect chain 3079 unlockEffectChains(effectChains); 3080 3081 standbyTime = systemTime() + kStandbyTimeInNsecs; 3082 for (size_t i = 0; i < outputTracks.size(); i++) { 3083 outputTracks[i]->write(mMixBuffer, writeFrames); 3084 } 3085 mStandby = false; 3086 mBytesWritten += mixBufferSize; 3087 } else { 3088 // enable changes in effect chain 3089 unlockEffectChains(effectChains); 3090 usleep(sleepTime); 3091 } 3092 3093 // finally let go of all our tracks, without the lock held 3094 // since we can't guarantee the destructors won't acquire that 3095 // same lock. 3096 tracksToRemove.clear(); 3097 outputTracks.clear(); 3098 3099 // Effect chains will be actually deleted here if they were removed from 3100 // mEffectChains list during mixing or effects processing 3101 effectChains.clear(); 3102 } 3103 3104 releaseWakeLock(); 3105 3106 return false; 3107} 3108 3109void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3110{ 3111 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3112 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3113 this, 3114 mSampleRate, 3115 mFormat, 3116 mChannelMask, 3117 frameCount); 3118 if (outputTrack->cblk() != NULL) { 3119 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3120 mOutputTracks.add(outputTrack); 3121 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3122 updateWaitTime(); 3123 } 3124} 3125 3126void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3127{ 3128 Mutex::Autolock _l(mLock); 3129 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3130 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3131 mOutputTracks[i]->destroy(); 3132 mOutputTracks.removeAt(i); 3133 updateWaitTime(); 3134 return; 3135 } 3136 } 3137 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3138} 3139 3140void AudioFlinger::DuplicatingThread::updateWaitTime() 3141{ 3142 mWaitTimeMs = UINT_MAX; 3143 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3144 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3145 if (strong != NULL) { 3146 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3147 if (waitTimeMs < mWaitTimeMs) { 3148 mWaitTimeMs = waitTimeMs; 3149 } 3150 } 3151 } 3152} 3153 3154 3155bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3156{ 3157 for (size_t i = 0; i < outputTracks.size(); i++) { 3158 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3159 if (thread == 0) { 3160 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3161 return false; 3162 } 3163 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3164 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3165 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3166 return false; 3167 } 3168 } 3169 return true; 3170} 3171 3172uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3173{ 3174 return (mWaitTimeMs * 1000) / 2; 3175} 3176 3177// ---------------------------------------------------------------------------- 3178 3179// TrackBase constructor must be called with AudioFlinger::mLock held 3180AudioFlinger::ThreadBase::TrackBase::TrackBase( 3181 const wp<ThreadBase>& thread, 3182 const sp<Client>& client, 3183 uint32_t sampleRate, 3184 uint32_t format, 3185 uint32_t channelMask, 3186 int frameCount, 3187 uint32_t flags, 3188 const sp<IMemory>& sharedBuffer, 3189 int sessionId) 3190 : RefBase(), 3191 mThread(thread), 3192 mClient(client), 3193 mCblk(0), 3194 mFrameCount(0), 3195 mState(IDLE), 3196 mClientTid(-1), 3197 mFormat(format), 3198 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3199 mSessionId(sessionId) 3200{ 3201 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3202 3203 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3204 size_t size = sizeof(audio_track_cblk_t); 3205 uint8_t channelCount = popcount(channelMask); 3206 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3207 if (sharedBuffer == 0) { 3208 size += bufferSize; 3209 } 3210 3211 if (client != NULL) { 3212 mCblkMemory = client->heap()->allocate(size); 3213 if (mCblkMemory != 0) { 3214 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3215 if (mCblk) { // construct the shared structure in-place. 3216 new(mCblk) audio_track_cblk_t(); 3217 // clear all buffers 3218 mCblk->frameCount = frameCount; 3219 mCblk->sampleRate = sampleRate; 3220 mChannelCount = channelCount; 3221 mChannelMask = channelMask; 3222 if (sharedBuffer == 0) { 3223 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3224 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3225 // Force underrun condition to avoid false underrun callback until first data is 3226 // written to buffer (other flags are cleared) 3227 mCblk->flags = CBLK_UNDERRUN_ON; 3228 } else { 3229 mBuffer = sharedBuffer->pointer(); 3230 } 3231 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3232 } 3233 } else { 3234 LOGE("not enough memory for AudioTrack size=%u", size); 3235 client->heap()->dump("AudioTrack"); 3236 return; 3237 } 3238 } else { 3239 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3240 if (mCblk) { // construct the shared structure in-place. 3241 new(mCblk) audio_track_cblk_t(); 3242 // clear all buffers 3243 mCblk->frameCount = frameCount; 3244 mCblk->sampleRate = sampleRate; 3245 mChannelCount = channelCount; 3246 mChannelMask = channelMask; 3247 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3248 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3249 // Force underrun condition to avoid false underrun callback until first data is 3250 // written to buffer (other flags are cleared) 3251 mCblk->flags = CBLK_UNDERRUN_ON; 3252 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3253 } 3254 } 3255} 3256 3257AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3258{ 3259 if (mCblk) { 3260 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3261 if (mClient == NULL) { 3262 delete mCblk; 3263 } 3264 } 3265 mCblkMemory.clear(); // and free the shared memory 3266 if (mClient != NULL) { 3267 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3268 mClient.clear(); 3269 } 3270} 3271 3272void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3273{ 3274 buffer->raw = NULL; 3275 mFrameCount = buffer->frameCount; 3276 step(); 3277 buffer->frameCount = 0; 3278} 3279 3280bool AudioFlinger::ThreadBase::TrackBase::step() { 3281 bool result; 3282 audio_track_cblk_t* cblk = this->cblk(); 3283 3284 result = cblk->stepServer(mFrameCount); 3285 if (!result) { 3286 ALOGV("stepServer failed acquiring cblk mutex"); 3287 mFlags |= STEPSERVER_FAILED; 3288 } 3289 return result; 3290} 3291 3292void AudioFlinger::ThreadBase::TrackBase::reset() { 3293 audio_track_cblk_t* cblk = this->cblk(); 3294 3295 cblk->user = 0; 3296 cblk->server = 0; 3297 cblk->userBase = 0; 3298 cblk->serverBase = 0; 3299 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3300 ALOGV("TrackBase::reset"); 3301} 3302 3303sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3304{ 3305 return mCblkMemory; 3306} 3307 3308int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3309 return (int)mCblk->sampleRate; 3310} 3311 3312int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3313 return (const int)mChannelCount; 3314} 3315 3316uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3317 return mChannelMask; 3318} 3319 3320void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3321 audio_track_cblk_t* cblk = this->cblk(); 3322 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 3323 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 3324 3325 // Check validity of returned pointer in case the track control block would have been corrupted. 3326 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3327 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 3328 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3329 server %d, serverBase %d, user %d, userBase %d", 3330 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3331 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3332 return 0; 3333 } 3334 3335 return bufferStart; 3336} 3337 3338// ---------------------------------------------------------------------------- 3339 3340// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3341AudioFlinger::PlaybackThread::Track::Track( 3342 const wp<ThreadBase>& thread, 3343 const sp<Client>& client, 3344 int streamType, 3345 uint32_t sampleRate, 3346 uint32_t format, 3347 uint32_t channelMask, 3348 int frameCount, 3349 const sp<IMemory>& sharedBuffer, 3350 int sessionId) 3351 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3352 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3353 mAuxEffectId(0), mHasVolumeController(false) 3354{ 3355 if (mCblk != NULL) { 3356 sp<ThreadBase> baseThread = thread.promote(); 3357 if (baseThread != 0) { 3358 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3359 mName = playbackThread->getTrackName_l(); 3360 mMainBuffer = playbackThread->mixBuffer(); 3361 } 3362 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3363 if (mName < 0) { 3364 LOGE("no more track names available"); 3365 } 3366 mVolume[0] = 1.0f; 3367 mVolume[1] = 1.0f; 3368 mStreamType = streamType; 3369 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3370 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3371 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3372 } 3373} 3374 3375AudioFlinger::PlaybackThread::Track::~Track() 3376{ 3377 ALOGV("PlaybackThread::Track destructor"); 3378 sp<ThreadBase> thread = mThread.promote(); 3379 if (thread != 0) { 3380 Mutex::Autolock _l(thread->mLock); 3381 mState = TERMINATED; 3382 } 3383} 3384 3385void AudioFlinger::PlaybackThread::Track::destroy() 3386{ 3387 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3388 // by removing it from mTracks vector, so there is a risk that this Tracks's 3389 // desctructor is called. As the destructor needs to lock mLock, 3390 // we must acquire a strong reference on this Track before locking mLock 3391 // here so that the destructor is called only when exiting this function. 3392 // On the other hand, as long as Track::destroy() is only called by 3393 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3394 // this Track with its member mTrack. 3395 sp<Track> keep(this); 3396 { // scope for mLock 3397 sp<ThreadBase> thread = mThread.promote(); 3398 if (thread != 0) { 3399 if (!isOutputTrack()) { 3400 if (mState == ACTIVE || mState == RESUMING) { 3401 AudioSystem::stopOutput(thread->id(), 3402 (audio_stream_type_t)mStreamType, 3403 mSessionId); 3404 3405 // to track the speaker usage 3406 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3407 } 3408 AudioSystem::releaseOutput(thread->id()); 3409 } 3410 Mutex::Autolock _l(thread->mLock); 3411 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3412 playbackThread->destroyTrack_l(this); 3413 } 3414 } 3415} 3416 3417void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3418{ 3419 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", 3420 mName - AudioMixer::TRACK0, 3421 (mClient == NULL) ? getpid() : mClient->pid(), 3422 mStreamType, 3423 mFormat, 3424 mChannelMask, 3425 mSessionId, 3426 mFrameCount, 3427 mState, 3428 mMute, 3429 mFillingUpStatus, 3430 mCblk->sampleRate, 3431 mCblk->volume[0], 3432 mCblk->volume[1], 3433 mCblk->server, 3434 mCblk->user, 3435 (int)mMainBuffer, 3436 (int)mAuxBuffer); 3437} 3438 3439status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3440{ 3441 audio_track_cblk_t* cblk = this->cblk(); 3442 uint32_t framesReady; 3443 uint32_t framesReq = buffer->frameCount; 3444 3445 // Check if last stepServer failed, try to step now 3446 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3447 if (!step()) goto getNextBuffer_exit; 3448 ALOGV("stepServer recovered"); 3449 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3450 } 3451 3452 framesReady = cblk->framesReady(); 3453 3454 if (LIKELY(framesReady)) { 3455 uint32_t s = cblk->server; 3456 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3457 3458 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3459 if (framesReq > framesReady) { 3460 framesReq = framesReady; 3461 } 3462 if (s + framesReq > bufferEnd) { 3463 framesReq = bufferEnd - s; 3464 } 3465 3466 buffer->raw = getBuffer(s, framesReq); 3467 if (buffer->raw == NULL) goto getNextBuffer_exit; 3468 3469 buffer->frameCount = framesReq; 3470 return NO_ERROR; 3471 } 3472 3473getNextBuffer_exit: 3474 buffer->raw = NULL; 3475 buffer->frameCount = 0; 3476 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3477 return NOT_ENOUGH_DATA; 3478} 3479 3480bool AudioFlinger::PlaybackThread::Track::isReady() const { 3481 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3482 3483 if (mCblk->framesReady() >= mCblk->frameCount || 3484 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3485 mFillingUpStatus = FS_FILLED; 3486 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3487 return true; 3488 } 3489 return false; 3490} 3491 3492status_t AudioFlinger::PlaybackThread::Track::start() 3493{ 3494 status_t status = NO_ERROR; 3495 ALOGV("start(%d), calling thread %d session %d", 3496 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3497 sp<ThreadBase> thread = mThread.promote(); 3498 if (thread != 0) { 3499 Mutex::Autolock _l(thread->mLock); 3500 int state = mState; 3501 // here the track could be either new, or restarted 3502 // in both cases "unstop" the track 3503 if (mState == PAUSED) { 3504 mState = TrackBase::RESUMING; 3505 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3506 } else { 3507 mState = TrackBase::ACTIVE; 3508 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3509 } 3510 3511 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3512 thread->mLock.unlock(); 3513 status = AudioSystem::startOutput(thread->id(), 3514 (audio_stream_type_t)mStreamType, 3515 mSessionId); 3516 thread->mLock.lock(); 3517 3518 // to track the speaker usage 3519 if (status == NO_ERROR) { 3520 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3521 } 3522 } 3523 if (status == NO_ERROR) { 3524 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3525 playbackThread->addTrack_l(this); 3526 } else { 3527 mState = state; 3528 } 3529 } else { 3530 status = BAD_VALUE; 3531 } 3532 return status; 3533} 3534 3535void AudioFlinger::PlaybackThread::Track::stop() 3536{ 3537 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3538 sp<ThreadBase> thread = mThread.promote(); 3539 if (thread != 0) { 3540 Mutex::Autolock _l(thread->mLock); 3541 int state = mState; 3542 if (mState > STOPPED) { 3543 mState = STOPPED; 3544 // If the track is not active (PAUSED and buffers full), flush buffers 3545 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3546 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3547 reset(); 3548 } 3549 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3550 } 3551 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3552 thread->mLock.unlock(); 3553 AudioSystem::stopOutput(thread->id(), 3554 (audio_stream_type_t)mStreamType, 3555 mSessionId); 3556 thread->mLock.lock(); 3557 3558 // to track the speaker usage 3559 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3560 } 3561 } 3562} 3563 3564void AudioFlinger::PlaybackThread::Track::pause() 3565{ 3566 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3567 sp<ThreadBase> thread = mThread.promote(); 3568 if (thread != 0) { 3569 Mutex::Autolock _l(thread->mLock); 3570 if (mState == ACTIVE || mState == RESUMING) { 3571 mState = PAUSING; 3572 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3573 if (!isOutputTrack()) { 3574 thread->mLock.unlock(); 3575 AudioSystem::stopOutput(thread->id(), 3576 (audio_stream_type_t)mStreamType, 3577 mSessionId); 3578 thread->mLock.lock(); 3579 3580 // to track the speaker usage 3581 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3582 } 3583 } 3584 } 3585} 3586 3587void AudioFlinger::PlaybackThread::Track::flush() 3588{ 3589 ALOGV("flush(%d)", mName); 3590 sp<ThreadBase> thread = mThread.promote(); 3591 if (thread != 0) { 3592 Mutex::Autolock _l(thread->mLock); 3593 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3594 return; 3595 } 3596 // No point remaining in PAUSED state after a flush => go to 3597 // STOPPED state 3598 mState = STOPPED; 3599 3600 // do not reset the track if it is still in the process of being stopped or paused. 3601 // this will be done by prepareTracks_l() when the track is stopped. 3602 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3603 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3604 reset(); 3605 } 3606 } 3607} 3608 3609void AudioFlinger::PlaybackThread::Track::reset() 3610{ 3611 // Do not reset twice to avoid discarding data written just after a flush and before 3612 // the audioflinger thread detects the track is stopped. 3613 if (!mResetDone) { 3614 TrackBase::reset(); 3615 // Force underrun condition to avoid false underrun callback until first data is 3616 // written to buffer 3617 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3618 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3619 mFillingUpStatus = FS_FILLING; 3620 mResetDone = true; 3621 } 3622} 3623 3624void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3625{ 3626 mMute = muted; 3627} 3628 3629void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3630{ 3631 mVolume[0] = left; 3632 mVolume[1] = right; 3633} 3634 3635status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3636{ 3637 status_t status = DEAD_OBJECT; 3638 sp<ThreadBase> thread = mThread.promote(); 3639 if (thread != 0) { 3640 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3641 status = playbackThread->attachAuxEffect(this, EffectId); 3642 } 3643 return status; 3644} 3645 3646void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3647{ 3648 mAuxEffectId = EffectId; 3649 mAuxBuffer = buffer; 3650} 3651 3652// ---------------------------------------------------------------------------- 3653 3654// RecordTrack constructor must be called with AudioFlinger::mLock held 3655AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3656 const wp<ThreadBase>& thread, 3657 const sp<Client>& client, 3658 uint32_t sampleRate, 3659 uint32_t format, 3660 uint32_t channelMask, 3661 int frameCount, 3662 uint32_t flags, 3663 int sessionId) 3664 : TrackBase(thread, client, sampleRate, format, 3665 channelMask, frameCount, flags, 0, sessionId), 3666 mOverflow(false) 3667{ 3668 if (mCblk != NULL) { 3669 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3670 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3671 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3672 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3673 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3674 } else { 3675 mCblk->frameSize = sizeof(int8_t); 3676 } 3677 } 3678} 3679 3680AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3681{ 3682 sp<ThreadBase> thread = mThread.promote(); 3683 if (thread != 0) { 3684 AudioSystem::releaseInput(thread->id()); 3685 } 3686} 3687 3688status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3689{ 3690 audio_track_cblk_t* cblk = this->cblk(); 3691 uint32_t framesAvail; 3692 uint32_t framesReq = buffer->frameCount; 3693 3694 // Check if last stepServer failed, try to step now 3695 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3696 if (!step()) goto getNextBuffer_exit; 3697 ALOGV("stepServer recovered"); 3698 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3699 } 3700 3701 framesAvail = cblk->framesAvailable_l(); 3702 3703 if (LIKELY(framesAvail)) { 3704 uint32_t s = cblk->server; 3705 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3706 3707 if (framesReq > framesAvail) { 3708 framesReq = framesAvail; 3709 } 3710 if (s + framesReq > bufferEnd) { 3711 framesReq = bufferEnd - s; 3712 } 3713 3714 buffer->raw = getBuffer(s, framesReq); 3715 if (buffer->raw == NULL) goto getNextBuffer_exit; 3716 3717 buffer->frameCount = framesReq; 3718 return NO_ERROR; 3719 } 3720 3721getNextBuffer_exit: 3722 buffer->raw = NULL; 3723 buffer->frameCount = 0; 3724 return NOT_ENOUGH_DATA; 3725} 3726 3727status_t AudioFlinger::RecordThread::RecordTrack::start() 3728{ 3729 sp<ThreadBase> thread = mThread.promote(); 3730 if (thread != 0) { 3731 RecordThread *recordThread = (RecordThread *)thread.get(); 3732 return recordThread->start(this); 3733 } else { 3734 return BAD_VALUE; 3735 } 3736} 3737 3738void AudioFlinger::RecordThread::RecordTrack::stop() 3739{ 3740 sp<ThreadBase> thread = mThread.promote(); 3741 if (thread != 0) { 3742 RecordThread *recordThread = (RecordThread *)thread.get(); 3743 recordThread->stop(this); 3744 TrackBase::reset(); 3745 // Force overerrun condition to avoid false overrun callback until first data is 3746 // read from buffer 3747 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3748 } 3749} 3750 3751void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3752{ 3753 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3754 (mClient == NULL) ? getpid() : mClient->pid(), 3755 mFormat, 3756 mChannelMask, 3757 mSessionId, 3758 mFrameCount, 3759 mState, 3760 mCblk->sampleRate, 3761 mCblk->server, 3762 mCblk->user); 3763} 3764 3765 3766// ---------------------------------------------------------------------------- 3767 3768AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3769 const wp<ThreadBase>& thread, 3770 DuplicatingThread *sourceThread, 3771 uint32_t sampleRate, 3772 uint32_t format, 3773 uint32_t channelMask, 3774 int frameCount) 3775 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3776 mActive(false), mSourceThread(sourceThread) 3777{ 3778 3779 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3780 if (mCblk != NULL) { 3781 mCblk->flags |= CBLK_DIRECTION_OUT; 3782 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3783 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3784 mOutBuffer.frameCount = 0; 3785 playbackThread->mTracks.add(this); 3786 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3787 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3788 mCblk, mBuffer, mCblk->buffers, 3789 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3790 } else { 3791 LOGW("Error creating output track on thread %p", playbackThread); 3792 } 3793} 3794 3795AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3796{ 3797 clearBufferQueue(); 3798} 3799 3800status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3801{ 3802 status_t status = Track::start(); 3803 if (status != NO_ERROR) { 3804 return status; 3805 } 3806 3807 mActive = true; 3808 mRetryCount = 127; 3809 return status; 3810} 3811 3812void AudioFlinger::PlaybackThread::OutputTrack::stop() 3813{ 3814 Track::stop(); 3815 clearBufferQueue(); 3816 mOutBuffer.frameCount = 0; 3817 mActive = false; 3818} 3819 3820bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3821{ 3822 Buffer *pInBuffer; 3823 Buffer inBuffer; 3824 uint32_t channelCount = mChannelCount; 3825 bool outputBufferFull = false; 3826 inBuffer.frameCount = frames; 3827 inBuffer.i16 = data; 3828 3829 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3830 3831 if (!mActive && frames != 0) { 3832 start(); 3833 sp<ThreadBase> thread = mThread.promote(); 3834 if (thread != 0) { 3835 MixerThread *mixerThread = (MixerThread *)thread.get(); 3836 if (mCblk->frameCount > frames){ 3837 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3838 uint32_t startFrames = (mCblk->frameCount - frames); 3839 pInBuffer = new Buffer; 3840 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3841 pInBuffer->frameCount = startFrames; 3842 pInBuffer->i16 = pInBuffer->mBuffer; 3843 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3844 mBufferQueue.add(pInBuffer); 3845 } else { 3846 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3847 } 3848 } 3849 } 3850 } 3851 3852 while (waitTimeLeftMs) { 3853 // First write pending buffers, then new data 3854 if (mBufferQueue.size()) { 3855 pInBuffer = mBufferQueue.itemAt(0); 3856 } else { 3857 pInBuffer = &inBuffer; 3858 } 3859 3860 if (pInBuffer->frameCount == 0) { 3861 break; 3862 } 3863 3864 if (mOutBuffer.frameCount == 0) { 3865 mOutBuffer.frameCount = pInBuffer->frameCount; 3866 nsecs_t startTime = systemTime(); 3867 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3868 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3869 outputBufferFull = true; 3870 break; 3871 } 3872 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3873 if (waitTimeLeftMs >= waitTimeMs) { 3874 waitTimeLeftMs -= waitTimeMs; 3875 } else { 3876 waitTimeLeftMs = 0; 3877 } 3878 } 3879 3880 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3881 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3882 mCblk->stepUser(outFrames); 3883 pInBuffer->frameCount -= outFrames; 3884 pInBuffer->i16 += outFrames * channelCount; 3885 mOutBuffer.frameCount -= outFrames; 3886 mOutBuffer.i16 += outFrames * channelCount; 3887 3888 if (pInBuffer->frameCount == 0) { 3889 if (mBufferQueue.size()) { 3890 mBufferQueue.removeAt(0); 3891 delete [] pInBuffer->mBuffer; 3892 delete pInBuffer; 3893 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3894 } else { 3895 break; 3896 } 3897 } 3898 } 3899 3900 // If we could not write all frames, allocate a buffer and queue it for next time. 3901 if (inBuffer.frameCount) { 3902 sp<ThreadBase> thread = mThread.promote(); 3903 if (thread != 0 && !thread->standby()) { 3904 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3905 pInBuffer = new Buffer; 3906 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3907 pInBuffer->frameCount = inBuffer.frameCount; 3908 pInBuffer->i16 = pInBuffer->mBuffer; 3909 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3910 mBufferQueue.add(pInBuffer); 3911 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3912 } else { 3913 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3914 } 3915 } 3916 } 3917 3918 // Calling write() with a 0 length buffer, means that no more data will be written: 3919 // If no more buffers are pending, fill output track buffer to make sure it is started 3920 // by output mixer. 3921 if (frames == 0 && mBufferQueue.size() == 0) { 3922 if (mCblk->user < mCblk->frameCount) { 3923 frames = mCblk->frameCount - mCblk->user; 3924 pInBuffer = new Buffer; 3925 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3926 pInBuffer->frameCount = frames; 3927 pInBuffer->i16 = pInBuffer->mBuffer; 3928 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3929 mBufferQueue.add(pInBuffer); 3930 } else if (mActive) { 3931 stop(); 3932 } 3933 } 3934 3935 return outputBufferFull; 3936} 3937 3938status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3939{ 3940 int active; 3941 status_t result; 3942 audio_track_cblk_t* cblk = mCblk; 3943 uint32_t framesReq = buffer->frameCount; 3944 3945// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3946 buffer->frameCount = 0; 3947 3948 uint32_t framesAvail = cblk->framesAvailable(); 3949 3950 3951 if (framesAvail == 0) { 3952 Mutex::Autolock _l(cblk->lock); 3953 goto start_loop_here; 3954 while (framesAvail == 0) { 3955 active = mActive; 3956 if (UNLIKELY(!active)) { 3957 ALOGV("Not active and NO_MORE_BUFFERS"); 3958 return AudioTrack::NO_MORE_BUFFERS; 3959 } 3960 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3961 if (result != NO_ERROR) { 3962 return AudioTrack::NO_MORE_BUFFERS; 3963 } 3964 // read the server count again 3965 start_loop_here: 3966 framesAvail = cblk->framesAvailable_l(); 3967 } 3968 } 3969 3970// if (framesAvail < framesReq) { 3971// return AudioTrack::NO_MORE_BUFFERS; 3972// } 3973 3974 if (framesReq > framesAvail) { 3975 framesReq = framesAvail; 3976 } 3977 3978 uint32_t u = cblk->user; 3979 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3980 3981 if (u + framesReq > bufferEnd) { 3982 framesReq = bufferEnd - u; 3983 } 3984 3985 buffer->frameCount = framesReq; 3986 buffer->raw = (void *)cblk->buffer(u); 3987 return NO_ERROR; 3988} 3989 3990 3991void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3992{ 3993 size_t size = mBufferQueue.size(); 3994 Buffer *pBuffer; 3995 3996 for (size_t i = 0; i < size; i++) { 3997 pBuffer = mBufferQueue.itemAt(i); 3998 delete [] pBuffer->mBuffer; 3999 delete pBuffer; 4000 } 4001 mBufferQueue.clear(); 4002} 4003 4004// ---------------------------------------------------------------------------- 4005 4006AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4007 : RefBase(), 4008 mAudioFlinger(audioFlinger), 4009 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4010 mPid(pid) 4011{ 4012 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4013} 4014 4015// Client destructor must be called with AudioFlinger::mLock held 4016AudioFlinger::Client::~Client() 4017{ 4018 mAudioFlinger->removeClient_l(mPid); 4019} 4020 4021const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4022{ 4023 return mMemoryDealer; 4024} 4025 4026// ---------------------------------------------------------------------------- 4027 4028AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4029 const sp<IAudioFlingerClient>& client, 4030 pid_t pid) 4031 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4032{ 4033} 4034 4035AudioFlinger::NotificationClient::~NotificationClient() 4036{ 4037 mClient.clear(); 4038} 4039 4040void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4041{ 4042 sp<NotificationClient> keep(this); 4043 { 4044 mAudioFlinger->removeNotificationClient(mPid); 4045 } 4046} 4047 4048// ---------------------------------------------------------------------------- 4049 4050AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4051 : BnAudioTrack(), 4052 mTrack(track) 4053{ 4054} 4055 4056AudioFlinger::TrackHandle::~TrackHandle() { 4057 // just stop the track on deletion, associated resources 4058 // will be freed from the main thread once all pending buffers have 4059 // been played. Unless it's not in the active track list, in which 4060 // case we free everything now... 4061 mTrack->destroy(); 4062} 4063 4064status_t AudioFlinger::TrackHandle::start() { 4065 return mTrack->start(); 4066} 4067 4068void AudioFlinger::TrackHandle::stop() { 4069 mTrack->stop(); 4070} 4071 4072void AudioFlinger::TrackHandle::flush() { 4073 mTrack->flush(); 4074} 4075 4076void AudioFlinger::TrackHandle::mute(bool e) { 4077 mTrack->mute(e); 4078} 4079 4080void AudioFlinger::TrackHandle::pause() { 4081 mTrack->pause(); 4082} 4083 4084void AudioFlinger::TrackHandle::setVolume(float left, float right) { 4085 mTrack->setVolume(left, right); 4086} 4087 4088sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4089 return mTrack->getCblk(); 4090} 4091 4092status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4093{ 4094 return mTrack->attachAuxEffect(EffectId); 4095} 4096 4097status_t AudioFlinger::TrackHandle::onTransact( 4098 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4099{ 4100 return BnAudioTrack::onTransact(code, data, reply, flags); 4101} 4102 4103// ---------------------------------------------------------------------------- 4104 4105sp<IAudioRecord> AudioFlinger::openRecord( 4106 pid_t pid, 4107 int input, 4108 uint32_t sampleRate, 4109 uint32_t format, 4110 uint32_t channelMask, 4111 int frameCount, 4112 uint32_t flags, 4113 int *sessionId, 4114 status_t *status) 4115{ 4116 sp<RecordThread::RecordTrack> recordTrack; 4117 sp<RecordHandle> recordHandle; 4118 sp<Client> client; 4119 wp<Client> wclient; 4120 status_t lStatus; 4121 RecordThread *thread; 4122 size_t inFrameCount; 4123 int lSessionId; 4124 4125 // check calling permissions 4126 if (!recordingAllowed()) { 4127 lStatus = PERMISSION_DENIED; 4128 goto Exit; 4129 } 4130 4131 // add client to list 4132 { // scope for mLock 4133 Mutex::Autolock _l(mLock); 4134 thread = checkRecordThread_l(input); 4135 if (thread == NULL) { 4136 lStatus = BAD_VALUE; 4137 goto Exit; 4138 } 4139 4140 wclient = mClients.valueFor(pid); 4141 if (wclient != NULL) { 4142 client = wclient.promote(); 4143 } else { 4144 client = new Client(this, pid); 4145 mClients.add(pid, client); 4146 } 4147 4148 // If no audio session id is provided, create one here 4149 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4150 lSessionId = *sessionId; 4151 } else { 4152 lSessionId = nextUniqueId(); 4153 if (sessionId != NULL) { 4154 *sessionId = lSessionId; 4155 } 4156 } 4157 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4158 recordTrack = thread->createRecordTrack_l(client, 4159 sampleRate, 4160 format, 4161 channelMask, 4162 frameCount, 4163 flags, 4164 lSessionId, 4165 &lStatus); 4166 } 4167 if (lStatus != NO_ERROR) { 4168 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4169 // destructor is called by the TrackBase destructor with mLock held 4170 client.clear(); 4171 recordTrack.clear(); 4172 goto Exit; 4173 } 4174 4175 // return to handle to client 4176 recordHandle = new RecordHandle(recordTrack); 4177 lStatus = NO_ERROR; 4178 4179Exit: 4180 if (status) { 4181 *status = lStatus; 4182 } 4183 return recordHandle; 4184} 4185 4186// ---------------------------------------------------------------------------- 4187 4188AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4189 : BnAudioRecord(), 4190 mRecordTrack(recordTrack) 4191{ 4192} 4193 4194AudioFlinger::RecordHandle::~RecordHandle() { 4195 stop(); 4196} 4197 4198status_t AudioFlinger::RecordHandle::start() { 4199 ALOGV("RecordHandle::start()"); 4200 return mRecordTrack->start(); 4201} 4202 4203void AudioFlinger::RecordHandle::stop() { 4204 ALOGV("RecordHandle::stop()"); 4205 mRecordTrack->stop(); 4206} 4207 4208sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4209 return mRecordTrack->getCblk(); 4210} 4211 4212status_t AudioFlinger::RecordHandle::onTransact( 4213 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4214{ 4215 return BnAudioRecord::onTransact(code, data, reply, flags); 4216} 4217 4218// ---------------------------------------------------------------------------- 4219 4220AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4221 AudioStreamIn *input, 4222 uint32_t sampleRate, 4223 uint32_t channels, 4224 int id, 4225 uint32_t device) : 4226 ThreadBase(audioFlinger, id, device), 4227 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4228{ 4229 mType = ThreadBase::RECORD; 4230 4231 snprintf(mName, kNameLength, "AudioIn_%d", id); 4232 4233 mReqChannelCount = popcount(channels); 4234 mReqSampleRate = sampleRate; 4235 readInputParameters(); 4236} 4237 4238 4239AudioFlinger::RecordThread::~RecordThread() 4240{ 4241 delete[] mRsmpInBuffer; 4242 if (mResampler != NULL) { 4243 delete mResampler; 4244 delete[] mRsmpOutBuffer; 4245 } 4246} 4247 4248void AudioFlinger::RecordThread::onFirstRef() 4249{ 4250 run(mName, PRIORITY_URGENT_AUDIO); 4251} 4252 4253status_t AudioFlinger::RecordThread::readyToRun() 4254{ 4255 status_t status = initCheck(); 4256 LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4257 return status; 4258} 4259 4260bool AudioFlinger::RecordThread::threadLoop() 4261{ 4262 AudioBufferProvider::Buffer buffer; 4263 sp<RecordTrack> activeTrack; 4264 Vector< sp<EffectChain> > effectChains; 4265 4266 nsecs_t lastWarning = 0; 4267 4268 acquireWakeLock(); 4269 4270 // start recording 4271 while (!exitPending()) { 4272 4273 processConfigEvents(); 4274 4275 { // scope for mLock 4276 Mutex::Autolock _l(mLock); 4277 checkForNewParameters_l(); 4278 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4279 if (!mStandby) { 4280 mInput->stream->common.standby(&mInput->stream->common); 4281 mStandby = true; 4282 } 4283 4284 if (exitPending()) break; 4285 4286 releaseWakeLock_l(); 4287 ALOGV("RecordThread: loop stopping"); 4288 // go to sleep 4289 mWaitWorkCV.wait(mLock); 4290 ALOGV("RecordThread: loop starting"); 4291 acquireWakeLock_l(); 4292 continue; 4293 } 4294 if (mActiveTrack != 0) { 4295 if (mActiveTrack->mState == TrackBase::PAUSING) { 4296 if (!mStandby) { 4297 mInput->stream->common.standby(&mInput->stream->common); 4298 mStandby = true; 4299 } 4300 mActiveTrack.clear(); 4301 mStartStopCond.broadcast(); 4302 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4303 if (mReqChannelCount != mActiveTrack->channelCount()) { 4304 mActiveTrack.clear(); 4305 mStartStopCond.broadcast(); 4306 } else if (mBytesRead != 0) { 4307 // record start succeeds only if first read from audio input 4308 // succeeds 4309 if (mBytesRead > 0) { 4310 mActiveTrack->mState = TrackBase::ACTIVE; 4311 } else { 4312 mActiveTrack.clear(); 4313 } 4314 mStartStopCond.broadcast(); 4315 } 4316 mStandby = false; 4317 } 4318 } 4319 lockEffectChains_l(effectChains); 4320 } 4321 4322 if (mActiveTrack != 0) { 4323 if (mActiveTrack->mState != TrackBase::ACTIVE && 4324 mActiveTrack->mState != TrackBase::RESUMING) { 4325 unlockEffectChains(effectChains); 4326 usleep(kRecordThreadSleepUs); 4327 continue; 4328 } 4329 for (size_t i = 0; i < effectChains.size(); i ++) { 4330 effectChains[i]->process_l(); 4331 } 4332 4333 buffer.frameCount = mFrameCount; 4334 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4335 size_t framesOut = buffer.frameCount; 4336 if (mResampler == NULL) { 4337 // no resampling 4338 while (framesOut) { 4339 size_t framesIn = mFrameCount - mRsmpInIndex; 4340 if (framesIn) { 4341 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4342 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4343 if (framesIn > framesOut) 4344 framesIn = framesOut; 4345 mRsmpInIndex += framesIn; 4346 framesOut -= framesIn; 4347 if ((int)mChannelCount == mReqChannelCount || 4348 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4349 memcpy(dst, src, framesIn * mFrameSize); 4350 } else { 4351 int16_t *src16 = (int16_t *)src; 4352 int16_t *dst16 = (int16_t *)dst; 4353 if (mChannelCount == 1) { 4354 while (framesIn--) { 4355 *dst16++ = *src16; 4356 *dst16++ = *src16++; 4357 } 4358 } else { 4359 while (framesIn--) { 4360 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4361 src16 += 2; 4362 } 4363 } 4364 } 4365 } 4366 if (framesOut && mFrameCount == mRsmpInIndex) { 4367 if (framesOut == mFrameCount && 4368 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4369 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4370 framesOut = 0; 4371 } else { 4372 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4373 mRsmpInIndex = 0; 4374 } 4375 if (mBytesRead < 0) { 4376 LOGE("Error reading audio input"); 4377 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4378 // Force input into standby so that it tries to 4379 // recover at next read attempt 4380 mInput->stream->common.standby(&mInput->stream->common); 4381 usleep(kRecordThreadSleepUs); 4382 } 4383 mRsmpInIndex = mFrameCount; 4384 framesOut = 0; 4385 buffer.frameCount = 0; 4386 } 4387 } 4388 } 4389 } else { 4390 // resampling 4391 4392 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4393 // alter output frame count as if we were expecting stereo samples 4394 if (mChannelCount == 1 && mReqChannelCount == 1) { 4395 framesOut >>= 1; 4396 } 4397 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4398 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4399 // are 32 bit aligned which should be always true. 4400 if (mChannelCount == 2 && mReqChannelCount == 1) { 4401 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4402 // the resampler always outputs stereo samples: do post stereo to mono conversion 4403 int16_t *src = (int16_t *)mRsmpOutBuffer; 4404 int16_t *dst = buffer.i16; 4405 while (framesOut--) { 4406 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4407 src += 2; 4408 } 4409 } else { 4410 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4411 } 4412 4413 } 4414 mActiveTrack->releaseBuffer(&buffer); 4415 mActiveTrack->overflow(); 4416 } 4417 // client isn't retrieving buffers fast enough 4418 else { 4419 if (!mActiveTrack->setOverflow()) { 4420 nsecs_t now = systemTime(); 4421 if ((now - lastWarning) > kWarningThrottleNs) { 4422 LOGW("RecordThread: buffer overflow"); 4423 lastWarning = now; 4424 } 4425 } 4426 // Release the processor for a while before asking for a new buffer. 4427 // This will give the application more chance to read from the buffer and 4428 // clear the overflow. 4429 usleep(kRecordThreadSleepUs); 4430 } 4431 } 4432 // enable changes in effect chain 4433 unlockEffectChains(effectChains); 4434 effectChains.clear(); 4435 } 4436 4437 if (!mStandby) { 4438 mInput->stream->common.standby(&mInput->stream->common); 4439 } 4440 mActiveTrack.clear(); 4441 4442 mStartStopCond.broadcast(); 4443 4444 releaseWakeLock(); 4445 4446 ALOGV("RecordThread %p exiting", this); 4447 return false; 4448} 4449 4450 4451sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4452 const sp<AudioFlinger::Client>& client, 4453 uint32_t sampleRate, 4454 int format, 4455 int channelMask, 4456 int frameCount, 4457 uint32_t flags, 4458 int sessionId, 4459 status_t *status) 4460{ 4461 sp<RecordTrack> track; 4462 status_t lStatus; 4463 4464 lStatus = initCheck(); 4465 if (lStatus != NO_ERROR) { 4466 LOGE("Audio driver not initialized."); 4467 goto Exit; 4468 } 4469 4470 { // scope for mLock 4471 Mutex::Autolock _l(mLock); 4472 4473 track = new RecordTrack(this, client, sampleRate, 4474 format, channelMask, frameCount, flags, sessionId); 4475 4476 if (track->getCblk() == NULL) { 4477 lStatus = NO_MEMORY; 4478 goto Exit; 4479 } 4480 4481 mTrack = track.get(); 4482 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4483 bool suspend = audio_is_bluetooth_sco_device( 4484 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4485 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4486 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4487 } 4488 lStatus = NO_ERROR; 4489 4490Exit: 4491 if (status) { 4492 *status = lStatus; 4493 } 4494 return track; 4495} 4496 4497status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4498{ 4499 ALOGV("RecordThread::start"); 4500 sp <ThreadBase> strongMe = this; 4501 status_t status = NO_ERROR; 4502 { 4503 AutoMutex lock(&mLock); 4504 if (mActiveTrack != 0) { 4505 if (recordTrack != mActiveTrack.get()) { 4506 status = -EBUSY; 4507 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4508 mActiveTrack->mState = TrackBase::ACTIVE; 4509 } 4510 return status; 4511 } 4512 4513 recordTrack->mState = TrackBase::IDLE; 4514 mActiveTrack = recordTrack; 4515 mLock.unlock(); 4516 status_t status = AudioSystem::startInput(mId); 4517 mLock.lock(); 4518 if (status != NO_ERROR) { 4519 mActiveTrack.clear(); 4520 return status; 4521 } 4522 mRsmpInIndex = mFrameCount; 4523 mBytesRead = 0; 4524 if (mResampler != NULL) { 4525 mResampler->reset(); 4526 } 4527 mActiveTrack->mState = TrackBase::RESUMING; 4528 // signal thread to start 4529 ALOGV("Signal record thread"); 4530 mWaitWorkCV.signal(); 4531 // do not wait for mStartStopCond if exiting 4532 if (mExiting) { 4533 mActiveTrack.clear(); 4534 status = INVALID_OPERATION; 4535 goto startError; 4536 } 4537 mStartStopCond.wait(mLock); 4538 if (mActiveTrack == 0) { 4539 ALOGV("Record failed to start"); 4540 status = BAD_VALUE; 4541 goto startError; 4542 } 4543 ALOGV("Record started OK"); 4544 return status; 4545 } 4546startError: 4547 AudioSystem::stopInput(mId); 4548 return status; 4549} 4550 4551void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4552 ALOGV("RecordThread::stop"); 4553 sp <ThreadBase> strongMe = this; 4554 { 4555 AutoMutex lock(&mLock); 4556 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4557 mActiveTrack->mState = TrackBase::PAUSING; 4558 // do not wait for mStartStopCond if exiting 4559 if (mExiting) { 4560 return; 4561 } 4562 mStartStopCond.wait(mLock); 4563 // if we have been restarted, recordTrack == mActiveTrack.get() here 4564 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4565 mLock.unlock(); 4566 AudioSystem::stopInput(mId); 4567 mLock.lock(); 4568 ALOGV("Record stopped OK"); 4569 } 4570 } 4571 } 4572} 4573 4574status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4575{ 4576 const size_t SIZE = 256; 4577 char buffer[SIZE]; 4578 String8 result; 4579 pid_t pid = 0; 4580 4581 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4582 result.append(buffer); 4583 4584 if (mActiveTrack != 0) { 4585 result.append("Active Track:\n"); 4586 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4587 mActiveTrack->dump(buffer, SIZE); 4588 result.append(buffer); 4589 4590 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4591 result.append(buffer); 4592 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4593 result.append(buffer); 4594 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4595 result.append(buffer); 4596 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4597 result.append(buffer); 4598 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4599 result.append(buffer); 4600 4601 4602 } else { 4603 result.append("No record client\n"); 4604 } 4605 write(fd, result.string(), result.size()); 4606 4607 dumpBase(fd, args); 4608 dumpEffectChains(fd, args); 4609 4610 return NO_ERROR; 4611} 4612 4613status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4614{ 4615 size_t framesReq = buffer->frameCount; 4616 size_t framesReady = mFrameCount - mRsmpInIndex; 4617 int channelCount; 4618 4619 if (framesReady == 0) { 4620 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4621 if (mBytesRead < 0) { 4622 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4623 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4624 // Force input into standby so that it tries to 4625 // recover at next read attempt 4626 mInput->stream->common.standby(&mInput->stream->common); 4627 usleep(kRecordThreadSleepUs); 4628 } 4629 buffer->raw = NULL; 4630 buffer->frameCount = 0; 4631 return NOT_ENOUGH_DATA; 4632 } 4633 mRsmpInIndex = 0; 4634 framesReady = mFrameCount; 4635 } 4636 4637 if (framesReq > framesReady) { 4638 framesReq = framesReady; 4639 } 4640 4641 if (mChannelCount == 1 && mReqChannelCount == 2) { 4642 channelCount = 1; 4643 } else { 4644 channelCount = 2; 4645 } 4646 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4647 buffer->frameCount = framesReq; 4648 return NO_ERROR; 4649} 4650 4651void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4652{ 4653 mRsmpInIndex += buffer->frameCount; 4654 buffer->frameCount = 0; 4655} 4656 4657bool AudioFlinger::RecordThread::checkForNewParameters_l() 4658{ 4659 bool reconfig = false; 4660 4661 while (!mNewParameters.isEmpty()) { 4662 status_t status = NO_ERROR; 4663 String8 keyValuePair = mNewParameters[0]; 4664 AudioParameter param = AudioParameter(keyValuePair); 4665 int value; 4666 int reqFormat = mFormat; 4667 int reqSamplingRate = mReqSampleRate; 4668 int reqChannelCount = mReqChannelCount; 4669 4670 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4671 reqSamplingRate = value; 4672 reconfig = true; 4673 } 4674 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4675 reqFormat = value; 4676 reconfig = true; 4677 } 4678 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4679 reqChannelCount = popcount(value); 4680 reconfig = true; 4681 } 4682 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4683 // do not accept frame count changes if tracks are open as the track buffer 4684 // size depends on frame count and correct behavior would not be garantied 4685 // if frame count is changed after track creation 4686 if (mActiveTrack != 0) { 4687 status = INVALID_OPERATION; 4688 } else { 4689 reconfig = true; 4690 } 4691 } 4692 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4693 // forward device change to effects that have requested to be 4694 // aware of attached audio device. 4695 for (size_t i = 0; i < mEffectChains.size(); i++) { 4696 mEffectChains[i]->setDevice_l(value); 4697 } 4698 // store input device and output device but do not forward output device to audio HAL. 4699 // Note that status is ignored by the caller for output device 4700 // (see AudioFlinger::setParameters() 4701 if (value & AUDIO_DEVICE_OUT_ALL) { 4702 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4703 status = BAD_VALUE; 4704 } else { 4705 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4706 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4707 if (mTrack != NULL) { 4708 bool suspend = audio_is_bluetooth_sco_device( 4709 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4710 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4711 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4712 } 4713 } 4714 mDevice |= (uint32_t)value; 4715 } 4716 if (status == NO_ERROR) { 4717 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4718 if (status == INVALID_OPERATION) { 4719 mInput->stream->common.standby(&mInput->stream->common); 4720 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4721 } 4722 if (reconfig) { 4723 if (status == BAD_VALUE && 4724 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4725 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4726 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4727 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4728 (reqChannelCount < 3)) { 4729 status = NO_ERROR; 4730 } 4731 if (status == NO_ERROR) { 4732 readInputParameters(); 4733 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4734 } 4735 } 4736 } 4737 4738 mNewParameters.removeAt(0); 4739 4740 mParamStatus = status; 4741 mParamCond.signal(); 4742 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4743 // already timed out waiting for the status and will never signal the condition. 4744 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4745 } 4746 return reconfig; 4747} 4748 4749String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4750{ 4751 char *s; 4752 String8 out_s8 = String8(); 4753 4754 Mutex::Autolock _l(mLock); 4755 if (initCheck() != NO_ERROR) { 4756 return out_s8; 4757 } 4758 4759 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4760 out_s8 = String8(s); 4761 free(s); 4762 return out_s8; 4763} 4764 4765void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4766 AudioSystem::OutputDescriptor desc; 4767 void *param2 = 0; 4768 4769 switch (event) { 4770 case AudioSystem::INPUT_OPENED: 4771 case AudioSystem::INPUT_CONFIG_CHANGED: 4772 desc.channels = mChannelMask; 4773 desc.samplingRate = mSampleRate; 4774 desc.format = mFormat; 4775 desc.frameCount = mFrameCount; 4776 desc.latency = 0; 4777 param2 = &desc; 4778 break; 4779 4780 case AudioSystem::INPUT_CLOSED: 4781 default: 4782 break; 4783 } 4784 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4785} 4786 4787void AudioFlinger::RecordThread::readInputParameters() 4788{ 4789 if (mRsmpInBuffer) delete mRsmpInBuffer; 4790 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4791 if (mResampler) delete mResampler; 4792 mResampler = NULL; 4793 4794 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4795 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4796 mChannelCount = (uint16_t)popcount(mChannelMask); 4797 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4798 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); 4799 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4800 mFrameCount = mInputBytes / mFrameSize; 4801 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4802 4803 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4804 { 4805 int channelCount; 4806 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4807 // stereo to mono post process as the resampler always outputs stereo. 4808 if (mChannelCount == 1 && mReqChannelCount == 2) { 4809 channelCount = 1; 4810 } else { 4811 channelCount = 2; 4812 } 4813 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4814 mResampler->setSampleRate(mSampleRate); 4815 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4816 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4817 4818 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4819 if (mChannelCount == 1 && mReqChannelCount == 1) { 4820 mFrameCount >>= 1; 4821 } 4822 4823 } 4824 mRsmpInIndex = mFrameCount; 4825} 4826 4827unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4828{ 4829 Mutex::Autolock _l(mLock); 4830 if (initCheck() != NO_ERROR) { 4831 return 0; 4832 } 4833 4834 return mInput->stream->get_input_frames_lost(mInput->stream); 4835} 4836 4837uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4838{ 4839 Mutex::Autolock _l(mLock); 4840 uint32_t result = 0; 4841 if (getEffectChain_l(sessionId) != 0) { 4842 result = EFFECT_SESSION; 4843 } 4844 4845 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4846 result |= TRACK_SESSION; 4847 } 4848 4849 return result; 4850} 4851 4852AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4853{ 4854 Mutex::Autolock _l(mLock); 4855 return mTrack; 4856} 4857 4858AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() 4859{ 4860 Mutex::Autolock _l(mLock); 4861 return mInput; 4862} 4863 4864AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4865{ 4866 Mutex::Autolock _l(mLock); 4867 AudioStreamIn *input = mInput; 4868 mInput = NULL; 4869 return input; 4870} 4871 4872// this method must always be called either with ThreadBase mLock held or inside the thread loop 4873audio_stream_t* AudioFlinger::RecordThread::stream() 4874{ 4875 if (mInput == NULL) { 4876 return NULL; 4877 } 4878 return &mInput->stream->common; 4879} 4880 4881 4882// ---------------------------------------------------------------------------- 4883 4884int AudioFlinger::openOutput(uint32_t *pDevices, 4885 uint32_t *pSamplingRate, 4886 uint32_t *pFormat, 4887 uint32_t *pChannels, 4888 uint32_t *pLatencyMs, 4889 uint32_t flags) 4890{ 4891 status_t status; 4892 PlaybackThread *thread = NULL; 4893 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4894 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4895 uint32_t format = pFormat ? *pFormat : 0; 4896 uint32_t channels = pChannels ? *pChannels : 0; 4897 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4898 audio_stream_out_t *outStream; 4899 audio_hw_device_t *outHwDev; 4900 4901 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4902 pDevices ? *pDevices : 0, 4903 samplingRate, 4904 format, 4905 channels, 4906 flags); 4907 4908 if (pDevices == NULL || *pDevices == 0) { 4909 return 0; 4910 } 4911 4912 Mutex::Autolock _l(mLock); 4913 4914 outHwDev = findSuitableHwDev_l(*pDevices); 4915 if (outHwDev == NULL) 4916 return 0; 4917 4918 status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, 4919 &channels, &samplingRate, &outStream); 4920 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4921 outStream, 4922 samplingRate, 4923 format, 4924 channels, 4925 status); 4926 4927 mHardwareStatus = AUDIO_HW_IDLE; 4928 if (outStream != NULL) { 4929 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4930 int id = nextUniqueId(); 4931 4932 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4933 (format != AUDIO_FORMAT_PCM_16_BIT) || 4934 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4935 thread = new DirectOutputThread(this, output, id, *pDevices); 4936 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4937 } else { 4938 thread = new MixerThread(this, output, id, *pDevices); 4939 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4940 } 4941 mPlaybackThreads.add(id, thread); 4942 4943 if (pSamplingRate) *pSamplingRate = samplingRate; 4944 if (pFormat) *pFormat = format; 4945 if (pChannels) *pChannels = channels; 4946 if (pLatencyMs) *pLatencyMs = thread->latency(); 4947 4948 // notify client processes of the new output creation 4949 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4950 return id; 4951 } 4952 4953 return 0; 4954} 4955 4956int AudioFlinger::openDuplicateOutput(int output1, int output2) 4957{ 4958 Mutex::Autolock _l(mLock); 4959 MixerThread *thread1 = checkMixerThread_l(output1); 4960 MixerThread *thread2 = checkMixerThread_l(output2); 4961 4962 if (thread1 == NULL || thread2 == NULL) { 4963 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4964 return 0; 4965 } 4966 4967 int id = nextUniqueId(); 4968 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4969 thread->addOutputTrack(thread2); 4970 mPlaybackThreads.add(id, thread); 4971 // notify client processes of the new output creation 4972 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4973 return id; 4974} 4975 4976status_t AudioFlinger::closeOutput(int output) 4977{ 4978 // keep strong reference on the playback thread so that 4979 // it is not destroyed while exit() is executed 4980 sp <PlaybackThread> thread; 4981 { 4982 Mutex::Autolock _l(mLock); 4983 thread = checkPlaybackThread_l(output); 4984 if (thread == NULL) { 4985 return BAD_VALUE; 4986 } 4987 4988 ALOGV("closeOutput() %d", output); 4989 4990 if (thread->type() == ThreadBase::MIXER) { 4991 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4992 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4993 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4994 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4995 } 4996 } 4997 } 4998 void *param2 = 0; 4999 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5000 mPlaybackThreads.removeItem(output); 5001 } 5002 thread->exit(); 5003 5004 if (thread->type() != ThreadBase::DUPLICATING) { 5005 AudioStreamOut *out = thread->clearOutput(); 5006 // from now on thread->mOutput is NULL 5007 out->hwDev->close_output_stream(out->hwDev, out->stream); 5008 delete out; 5009 } 5010 return NO_ERROR; 5011} 5012 5013status_t AudioFlinger::suspendOutput(int output) 5014{ 5015 Mutex::Autolock _l(mLock); 5016 PlaybackThread *thread = checkPlaybackThread_l(output); 5017 5018 if (thread == NULL) { 5019 return BAD_VALUE; 5020 } 5021 5022 ALOGV("suspendOutput() %d", output); 5023 thread->suspend(); 5024 5025 return NO_ERROR; 5026} 5027 5028status_t AudioFlinger::restoreOutput(int output) 5029{ 5030 Mutex::Autolock _l(mLock); 5031 PlaybackThread *thread = checkPlaybackThread_l(output); 5032 5033 if (thread == NULL) { 5034 return BAD_VALUE; 5035 } 5036 5037 ALOGV("restoreOutput() %d", output); 5038 5039 thread->restore(); 5040 5041 return NO_ERROR; 5042} 5043 5044int AudioFlinger::openInput(uint32_t *pDevices, 5045 uint32_t *pSamplingRate, 5046 uint32_t *pFormat, 5047 uint32_t *pChannels, 5048 uint32_t acoustics) 5049{ 5050 status_t status; 5051 RecordThread *thread = NULL; 5052 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5053 uint32_t format = pFormat ? *pFormat : 0; 5054 uint32_t channels = pChannels ? *pChannels : 0; 5055 uint32_t reqSamplingRate = samplingRate; 5056 uint32_t reqFormat = format; 5057 uint32_t reqChannels = channels; 5058 audio_stream_in_t *inStream; 5059 audio_hw_device_t *inHwDev; 5060 5061 if (pDevices == NULL || *pDevices == 0) { 5062 return 0; 5063 } 5064 5065 Mutex::Autolock _l(mLock); 5066 5067 inHwDev = findSuitableHwDev_l(*pDevices); 5068 if (inHwDev == NULL) 5069 return 0; 5070 5071 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5072 &channels, &samplingRate, 5073 (audio_in_acoustics_t)acoustics, 5074 &inStream); 5075 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5076 inStream, 5077 samplingRate, 5078 format, 5079 channels, 5080 acoustics, 5081 status); 5082 5083 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5084 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5085 // or stereo to mono conversions on 16 bit PCM inputs. 5086 if (inStream == NULL && status == BAD_VALUE && 5087 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5088 (samplingRate <= 2 * reqSamplingRate) && 5089 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5090 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5091 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5092 &channels, &samplingRate, 5093 (audio_in_acoustics_t)acoustics, 5094 &inStream); 5095 } 5096 5097 if (inStream != NULL) { 5098 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5099 5100 int id = nextUniqueId(); 5101 // Start record thread 5102 // RecorThread require both input and output device indication to forward to audio 5103 // pre processing modules 5104 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5105 thread = new RecordThread(this, 5106 input, 5107 reqSamplingRate, 5108 reqChannels, 5109 id, 5110 device); 5111 mRecordThreads.add(id, thread); 5112 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5113 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5114 if (pFormat) *pFormat = format; 5115 if (pChannels) *pChannels = reqChannels; 5116 5117 input->stream->common.standby(&input->stream->common); 5118 5119 // notify client processes of the new input creation 5120 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5121 return id; 5122 } 5123 5124 return 0; 5125} 5126 5127status_t AudioFlinger::closeInput(int input) 5128{ 5129 // keep strong reference on the record thread so that 5130 // it is not destroyed while exit() is executed 5131 sp <RecordThread> thread; 5132 { 5133 Mutex::Autolock _l(mLock); 5134 thread = checkRecordThread_l(input); 5135 if (thread == NULL) { 5136 return BAD_VALUE; 5137 } 5138 5139 ALOGV("closeInput() %d", input); 5140 void *param2 = 0; 5141 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5142 mRecordThreads.removeItem(input); 5143 } 5144 thread->exit(); 5145 5146 AudioStreamIn *in = thread->clearInput(); 5147 // from now on thread->mInput is NULL 5148 in->hwDev->close_input_stream(in->hwDev, in->stream); 5149 delete in; 5150 5151 return NO_ERROR; 5152} 5153 5154status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 5155{ 5156 Mutex::Autolock _l(mLock); 5157 MixerThread *dstThread = checkMixerThread_l(output); 5158 if (dstThread == NULL) { 5159 LOGW("setStreamOutput() bad output id %d", output); 5160 return BAD_VALUE; 5161 } 5162 5163 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5164 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5165 5166 dstThread->setStreamValid(stream, true); 5167 5168 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5169 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5170 if (thread != dstThread && 5171 thread->type() != ThreadBase::DIRECT) { 5172 MixerThread *srcThread = (MixerThread *)thread; 5173 srcThread->setStreamValid(stream, false); 5174 srcThread->invalidateTracks(stream); 5175 } 5176 } 5177 5178 return NO_ERROR; 5179} 5180 5181 5182int AudioFlinger::newAudioSessionId() 5183{ 5184 return nextUniqueId(); 5185} 5186 5187void AudioFlinger::acquireAudioSessionId(int audioSession) 5188{ 5189 Mutex::Autolock _l(mLock); 5190 int caller = IPCThreadState::self()->getCallingPid(); 5191 ALOGV("acquiring %d from %d", audioSession, caller); 5192 int num = mAudioSessionRefs.size(); 5193 for (int i = 0; i< num; i++) { 5194 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5195 if (ref->sessionid == audioSession && ref->pid == caller) { 5196 ref->cnt++; 5197 ALOGV(" incremented refcount to %d", ref->cnt); 5198 return; 5199 } 5200 } 5201 AudioSessionRef *ref = new AudioSessionRef(); 5202 ref->sessionid = audioSession; 5203 ref->pid = caller; 5204 ref->cnt = 1; 5205 mAudioSessionRefs.push(ref); 5206 ALOGV(" added new entry for %d", ref->sessionid); 5207} 5208 5209void AudioFlinger::releaseAudioSessionId(int audioSession) 5210{ 5211 Mutex::Autolock _l(mLock); 5212 int caller = IPCThreadState::self()->getCallingPid(); 5213 ALOGV("releasing %d from %d", audioSession, caller); 5214 int num = mAudioSessionRefs.size(); 5215 for (int i = 0; i< num; i++) { 5216 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5217 if (ref->sessionid == audioSession && ref->pid == caller) { 5218 ref->cnt--; 5219 ALOGV(" decremented refcount to %d", ref->cnt); 5220 if (ref->cnt == 0) { 5221 mAudioSessionRefs.removeAt(i); 5222 delete ref; 5223 purgeStaleEffects_l(); 5224 } 5225 return; 5226 } 5227 } 5228 LOGW("session id %d not found for pid %d", audioSession, caller); 5229} 5230 5231void AudioFlinger::purgeStaleEffects_l() { 5232 5233 ALOGV("purging stale effects"); 5234 5235 Vector< sp<EffectChain> > chains; 5236 5237 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5238 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5239 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5240 sp<EffectChain> ec = t->mEffectChains[j]; 5241 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5242 chains.push(ec); 5243 } 5244 } 5245 } 5246 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5247 sp<RecordThread> t = mRecordThreads.valueAt(i); 5248 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5249 sp<EffectChain> ec = t->mEffectChains[j]; 5250 chains.push(ec); 5251 } 5252 } 5253 5254 for (size_t i = 0; i < chains.size(); i++) { 5255 sp<EffectChain> ec = chains[i]; 5256 int sessionid = ec->sessionId(); 5257 sp<ThreadBase> t = ec->mThread.promote(); 5258 if (t == 0) { 5259 continue; 5260 } 5261 size_t numsessionrefs = mAudioSessionRefs.size(); 5262 bool found = false; 5263 for (size_t k = 0; k < numsessionrefs; k++) { 5264 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5265 if (ref->sessionid == sessionid) { 5266 ALOGV(" session %d still exists for %d with %d refs", 5267 sessionid, ref->pid, ref->cnt); 5268 found = true; 5269 break; 5270 } 5271 } 5272 if (!found) { 5273 // remove all effects from the chain 5274 while (ec->mEffects.size()) { 5275 sp<EffectModule> effect = ec->mEffects[0]; 5276 effect->unPin(); 5277 Mutex::Autolock _l (t->mLock); 5278 t->removeEffect_l(effect); 5279 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5280 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5281 if (handle != 0) { 5282 handle->mEffect.clear(); 5283 if (handle->mHasControl && handle->mEnabled) { 5284 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5285 } 5286 } 5287 } 5288 AudioSystem::unregisterEffect(effect->id()); 5289 } 5290 } 5291 } 5292 return; 5293} 5294 5295// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5296AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5297{ 5298 PlaybackThread *thread = NULL; 5299 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5300 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5301 } 5302 return thread; 5303} 5304 5305// checkMixerThread_l() must be called with AudioFlinger::mLock held 5306AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5307{ 5308 PlaybackThread *thread = checkPlaybackThread_l(output); 5309 if (thread != NULL) { 5310 if (thread->type() == ThreadBase::DIRECT) { 5311 thread = NULL; 5312 } 5313 } 5314 return (MixerThread *)thread; 5315} 5316 5317// checkRecordThread_l() must be called with AudioFlinger::mLock held 5318AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5319{ 5320 RecordThread *thread = NULL; 5321 if (mRecordThreads.indexOfKey(input) >= 0) { 5322 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5323 } 5324 return thread; 5325} 5326 5327uint32_t AudioFlinger::nextUniqueId() 5328{ 5329 return android_atomic_inc(&mNextUniqueId); 5330} 5331 5332AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5333{ 5334 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5335 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5336 AudioStreamOut *output = thread->getOutput(); 5337 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5338 return thread; 5339 } 5340 } 5341 return NULL; 5342} 5343 5344uint32_t AudioFlinger::primaryOutputDevice_l() 5345{ 5346 PlaybackThread *thread = primaryPlaybackThread_l(); 5347 5348 if (thread == NULL) { 5349 return 0; 5350 } 5351 5352 return thread->device(); 5353} 5354 5355 5356// ---------------------------------------------------------------------------- 5357// Effect management 5358// ---------------------------------------------------------------------------- 5359 5360 5361status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5362{ 5363 Mutex::Autolock _l(mLock); 5364 return EffectQueryNumberEffects(numEffects); 5365} 5366 5367status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5368{ 5369 Mutex::Autolock _l(mLock); 5370 return EffectQueryEffect(index, descriptor); 5371} 5372 5373status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5374{ 5375 Mutex::Autolock _l(mLock); 5376 return EffectGetDescriptor(pUuid, descriptor); 5377} 5378 5379 5380sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5381 effect_descriptor_t *pDesc, 5382 const sp<IEffectClient>& effectClient, 5383 int32_t priority, 5384 int io, 5385 int sessionId, 5386 status_t *status, 5387 int *id, 5388 int *enabled) 5389{ 5390 status_t lStatus = NO_ERROR; 5391 sp<EffectHandle> handle; 5392 effect_descriptor_t desc; 5393 sp<Client> client; 5394 wp<Client> wclient; 5395 5396 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5397 pid, effectClient.get(), priority, sessionId, io); 5398 5399 if (pDesc == NULL) { 5400 lStatus = BAD_VALUE; 5401 goto Exit; 5402 } 5403 5404 // check audio settings permission for global effects 5405 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5406 lStatus = PERMISSION_DENIED; 5407 goto Exit; 5408 } 5409 5410 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5411 // that can only be created by audio policy manager (running in same process) 5412 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5413 lStatus = PERMISSION_DENIED; 5414 goto Exit; 5415 } 5416 5417 if (io == 0) { 5418 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5419 // output must be specified by AudioPolicyManager when using session 5420 // AUDIO_SESSION_OUTPUT_STAGE 5421 lStatus = BAD_VALUE; 5422 goto Exit; 5423 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5424 // if the output returned by getOutputForEffect() is removed before we lock the 5425 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5426 // and we will exit safely 5427 io = AudioSystem::getOutputForEffect(&desc); 5428 } 5429 } 5430 5431 { 5432 Mutex::Autolock _l(mLock); 5433 5434 5435 if (!EffectIsNullUuid(&pDesc->uuid)) { 5436 // if uuid is specified, request effect descriptor 5437 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5438 if (lStatus < 0) { 5439 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5440 goto Exit; 5441 } 5442 } else { 5443 // if uuid is not specified, look for an available implementation 5444 // of the required type in effect factory 5445 if (EffectIsNullUuid(&pDesc->type)) { 5446 LOGW("createEffect() no effect type"); 5447 lStatus = BAD_VALUE; 5448 goto Exit; 5449 } 5450 uint32_t numEffects = 0; 5451 effect_descriptor_t d; 5452 d.flags = 0; // prevent compiler warning 5453 bool found = false; 5454 5455 lStatus = EffectQueryNumberEffects(&numEffects); 5456 if (lStatus < 0) { 5457 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5458 goto Exit; 5459 } 5460 for (uint32_t i = 0; i < numEffects; i++) { 5461 lStatus = EffectQueryEffect(i, &desc); 5462 if (lStatus < 0) { 5463 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5464 continue; 5465 } 5466 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5467 // If matching type found save effect descriptor. If the session is 5468 // 0 and the effect is not auxiliary, continue enumeration in case 5469 // an auxiliary version of this effect type is available 5470 found = true; 5471 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5472 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5473 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5474 break; 5475 } 5476 } 5477 } 5478 if (!found) { 5479 lStatus = BAD_VALUE; 5480 LOGW("createEffect() effect not found"); 5481 goto Exit; 5482 } 5483 // For same effect type, chose auxiliary version over insert version if 5484 // connect to output mix (Compliance to OpenSL ES) 5485 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5486 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5487 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5488 } 5489 } 5490 5491 // Do not allow auxiliary effects on a session different from 0 (output mix) 5492 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5493 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5494 lStatus = INVALID_OPERATION; 5495 goto Exit; 5496 } 5497 5498 // check recording permission for visualizer 5499 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5500 !recordingAllowed()) { 5501 lStatus = PERMISSION_DENIED; 5502 goto Exit; 5503 } 5504 5505 // return effect descriptor 5506 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5507 5508 // If output is not specified try to find a matching audio session ID in one of the 5509 // output threads. 5510 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5511 // because of code checking output when entering the function. 5512 // Note: io is never 0 when creating an effect on an input 5513 if (io == 0) { 5514 // look for the thread where the specified audio session is present 5515 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5516 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5517 io = mPlaybackThreads.keyAt(i); 5518 break; 5519 } 5520 } 5521 if (io == 0) { 5522 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5523 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5524 io = mRecordThreads.keyAt(i); 5525 break; 5526 } 5527 } 5528 } 5529 // If no output thread contains the requested session ID, default to 5530 // first output. The effect chain will be moved to the correct output 5531 // thread when a track with the same session ID is created 5532 if (io == 0 && mPlaybackThreads.size()) { 5533 io = mPlaybackThreads.keyAt(0); 5534 } 5535 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5536 } 5537 ThreadBase *thread = checkRecordThread_l(io); 5538 if (thread == NULL) { 5539 thread = checkPlaybackThread_l(io); 5540 if (thread == NULL) { 5541 LOGE("createEffect() unknown output thread"); 5542 lStatus = BAD_VALUE; 5543 goto Exit; 5544 } 5545 } 5546 5547 wclient = mClients.valueFor(pid); 5548 5549 if (wclient != NULL) { 5550 client = wclient.promote(); 5551 } else { 5552 client = new Client(this, pid); 5553 mClients.add(pid, client); 5554 } 5555 5556 // create effect on selected output thread 5557 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5558 &desc, enabled, &lStatus); 5559 if (handle != 0 && id != NULL) { 5560 *id = handle->id(); 5561 } 5562 } 5563 5564Exit: 5565 if(status) { 5566 *status = lStatus; 5567 } 5568 return handle; 5569} 5570 5571status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5572{ 5573 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5574 sessionId, srcOutput, dstOutput); 5575 Mutex::Autolock _l(mLock); 5576 if (srcOutput == dstOutput) { 5577 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 5578 return NO_ERROR; 5579 } 5580 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5581 if (srcThread == NULL) { 5582 LOGW("moveEffects() bad srcOutput %d", srcOutput); 5583 return BAD_VALUE; 5584 } 5585 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5586 if (dstThread == NULL) { 5587 LOGW("moveEffects() bad dstOutput %d", dstOutput); 5588 return BAD_VALUE; 5589 } 5590 5591 Mutex::Autolock _dl(dstThread->mLock); 5592 Mutex::Autolock _sl(srcThread->mLock); 5593 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5594 5595 return NO_ERROR; 5596} 5597 5598// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5599status_t AudioFlinger::moveEffectChain_l(int sessionId, 5600 AudioFlinger::PlaybackThread *srcThread, 5601 AudioFlinger::PlaybackThread *dstThread, 5602 bool reRegister) 5603{ 5604 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5605 sessionId, srcThread, dstThread); 5606 5607 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5608 if (chain == 0) { 5609 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5610 sessionId, srcThread); 5611 return INVALID_OPERATION; 5612 } 5613 5614 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5615 // so that a new chain is created with correct parameters when first effect is added. This is 5616 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5617 // removed. 5618 srcThread->removeEffectChain_l(chain); 5619 5620 // transfer all effects one by one so that new effect chain is created on new thread with 5621 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5622 int dstOutput = dstThread->id(); 5623 sp<EffectChain> dstChain; 5624 uint32_t strategy = 0; // prevent compiler warning 5625 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5626 while (effect != 0) { 5627 srcThread->removeEffect_l(effect); 5628 dstThread->addEffect_l(effect); 5629 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5630 if (effect->state() == EffectModule::ACTIVE || 5631 effect->state() == EffectModule::STOPPING) { 5632 effect->start(); 5633 } 5634 // if the move request is not received from audio policy manager, the effect must be 5635 // re-registered with the new strategy and output 5636 if (dstChain == 0) { 5637 dstChain = effect->chain().promote(); 5638 if (dstChain == 0) { 5639 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5640 srcThread->addEffect_l(effect); 5641 return NO_INIT; 5642 } 5643 strategy = dstChain->strategy(); 5644 } 5645 if (reRegister) { 5646 AudioSystem::unregisterEffect(effect->id()); 5647 AudioSystem::registerEffect(&effect->desc(), 5648 dstOutput, 5649 strategy, 5650 sessionId, 5651 effect->id()); 5652 } 5653 effect = chain->getEffectFromId_l(0); 5654 } 5655 5656 return NO_ERROR; 5657} 5658 5659 5660// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5661sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5662 const sp<AudioFlinger::Client>& client, 5663 const sp<IEffectClient>& effectClient, 5664 int32_t priority, 5665 int sessionId, 5666 effect_descriptor_t *desc, 5667 int *enabled, 5668 status_t *status 5669 ) 5670{ 5671 sp<EffectModule> effect; 5672 sp<EffectHandle> handle; 5673 status_t lStatus; 5674 sp<EffectChain> chain; 5675 bool chainCreated = false; 5676 bool effectCreated = false; 5677 bool effectRegistered = false; 5678 5679 lStatus = initCheck(); 5680 if (lStatus != NO_ERROR) { 5681 LOGW("createEffect_l() Audio driver not initialized."); 5682 goto Exit; 5683 } 5684 5685 // Do not allow effects with session ID 0 on direct output or duplicating threads 5686 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5687 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5688 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5689 desc->name, sessionId); 5690 lStatus = BAD_VALUE; 5691 goto Exit; 5692 } 5693 // Only Pre processor effects are allowed on input threads and only on input threads 5694 if ((mType == RECORD && 5695 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5696 (mType != RECORD && 5697 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5698 LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5699 desc->name, desc->flags, mType); 5700 lStatus = BAD_VALUE; 5701 goto Exit; 5702 } 5703 5704 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5705 5706 { // scope for mLock 5707 Mutex::Autolock _l(mLock); 5708 5709 // check for existing effect chain with the requested audio session 5710 chain = getEffectChain_l(sessionId); 5711 if (chain == 0) { 5712 // create a new chain for this session 5713 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5714 chain = new EffectChain(this, sessionId); 5715 addEffectChain_l(chain); 5716 chain->setStrategy(getStrategyForSession_l(sessionId)); 5717 chainCreated = true; 5718 } else { 5719 effect = chain->getEffectFromDesc_l(desc); 5720 } 5721 5722 ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 5723 5724 if (effect == 0) { 5725 int id = mAudioFlinger->nextUniqueId(); 5726 // Check CPU and memory usage 5727 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5728 if (lStatus != NO_ERROR) { 5729 goto Exit; 5730 } 5731 effectRegistered = true; 5732 // create a new effect module if none present in the chain 5733 effect = new EffectModule(this, chain, desc, id, sessionId); 5734 lStatus = effect->status(); 5735 if (lStatus != NO_ERROR) { 5736 goto Exit; 5737 } 5738 lStatus = chain->addEffect_l(effect); 5739 if (lStatus != NO_ERROR) { 5740 goto Exit; 5741 } 5742 effectCreated = true; 5743 5744 effect->setDevice(mDevice); 5745 effect->setMode(mAudioFlinger->getMode()); 5746 } 5747 // create effect handle and connect it to effect module 5748 handle = new EffectHandle(effect, client, effectClient, priority); 5749 lStatus = effect->addHandle(handle); 5750 if (enabled) { 5751 *enabled = (int)effect->isEnabled(); 5752 } 5753 } 5754 5755Exit: 5756 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5757 Mutex::Autolock _l(mLock); 5758 if (effectCreated) { 5759 chain->removeEffect_l(effect); 5760 } 5761 if (effectRegistered) { 5762 AudioSystem::unregisterEffect(effect->id()); 5763 } 5764 if (chainCreated) { 5765 removeEffectChain_l(chain); 5766 } 5767 handle.clear(); 5768 } 5769 5770 if(status) { 5771 *status = lStatus; 5772 } 5773 return handle; 5774} 5775 5776sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5777{ 5778 sp<EffectModule> effect; 5779 5780 sp<EffectChain> chain = getEffectChain_l(sessionId); 5781 if (chain != 0) { 5782 effect = chain->getEffectFromId_l(effectId); 5783 } 5784 return effect; 5785} 5786 5787// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5788// PlaybackThread::mLock held 5789status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5790{ 5791 // check for existing effect chain with the requested audio session 5792 int sessionId = effect->sessionId(); 5793 sp<EffectChain> chain = getEffectChain_l(sessionId); 5794 bool chainCreated = false; 5795 5796 if (chain == 0) { 5797 // create a new chain for this session 5798 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5799 chain = new EffectChain(this, sessionId); 5800 addEffectChain_l(chain); 5801 chain->setStrategy(getStrategyForSession_l(sessionId)); 5802 chainCreated = true; 5803 } 5804 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5805 5806 if (chain->getEffectFromId_l(effect->id()) != 0) { 5807 LOGW("addEffect_l() %p effect %s already present in chain %p", 5808 this, effect->desc().name, chain.get()); 5809 return BAD_VALUE; 5810 } 5811 5812 status_t status = chain->addEffect_l(effect); 5813 if (status != NO_ERROR) { 5814 if (chainCreated) { 5815 removeEffectChain_l(chain); 5816 } 5817 return status; 5818 } 5819 5820 effect->setDevice(mDevice); 5821 effect->setMode(mAudioFlinger->getMode()); 5822 return NO_ERROR; 5823} 5824 5825void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5826 5827 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5828 effect_descriptor_t desc = effect->desc(); 5829 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5830 detachAuxEffect_l(effect->id()); 5831 } 5832 5833 sp<EffectChain> chain = effect->chain().promote(); 5834 if (chain != 0) { 5835 // remove effect chain if removing last effect 5836 if (chain->removeEffect_l(effect) == 0) { 5837 removeEffectChain_l(chain); 5838 } 5839 } else { 5840 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5841 } 5842} 5843 5844void AudioFlinger::ThreadBase::lockEffectChains_l( 5845 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5846{ 5847 effectChains = mEffectChains; 5848 for (size_t i = 0; i < mEffectChains.size(); i++) { 5849 mEffectChains[i]->lock(); 5850 } 5851} 5852 5853void AudioFlinger::ThreadBase::unlockEffectChains( 5854 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5855{ 5856 for (size_t i = 0; i < effectChains.size(); i++) { 5857 effectChains[i]->unlock(); 5858 } 5859} 5860 5861sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5862{ 5863 Mutex::Autolock _l(mLock); 5864 return getEffectChain_l(sessionId); 5865} 5866 5867sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5868{ 5869 sp<EffectChain> chain; 5870 5871 size_t size = mEffectChains.size(); 5872 for (size_t i = 0; i < size; i++) { 5873 if (mEffectChains[i]->sessionId() == sessionId) { 5874 chain = mEffectChains[i]; 5875 break; 5876 } 5877 } 5878 return chain; 5879} 5880 5881void AudioFlinger::ThreadBase::setMode(uint32_t mode) 5882{ 5883 Mutex::Autolock _l(mLock); 5884 size_t size = mEffectChains.size(); 5885 for (size_t i = 0; i < size; i++) { 5886 mEffectChains[i]->setMode_l(mode); 5887 } 5888} 5889 5890void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5891 const wp<EffectHandle>& handle, 5892 bool unpiniflast) { 5893 5894 Mutex::Autolock _l(mLock); 5895 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5896 // delete the effect module if removing last handle on it 5897 if (effect->removeHandle(handle) == 0) { 5898 if (!effect->isPinned() || unpiniflast) { 5899 removeEffect_l(effect); 5900 AudioSystem::unregisterEffect(effect->id()); 5901 } 5902 } 5903} 5904 5905status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5906{ 5907 int session = chain->sessionId(); 5908 int16_t *buffer = mMixBuffer; 5909 bool ownsBuffer = false; 5910 5911 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5912 if (session > 0) { 5913 // Only one effect chain can be present in direct output thread and it uses 5914 // the mix buffer as input 5915 if (mType != DIRECT) { 5916 size_t numSamples = mFrameCount * mChannelCount; 5917 buffer = new int16_t[numSamples]; 5918 memset(buffer, 0, numSamples * sizeof(int16_t)); 5919 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5920 ownsBuffer = true; 5921 } 5922 5923 // Attach all tracks with same session ID to this chain. 5924 for (size_t i = 0; i < mTracks.size(); ++i) { 5925 sp<Track> track = mTracks[i]; 5926 if (session == track->sessionId()) { 5927 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5928 track->setMainBuffer(buffer); 5929 chain->incTrackCnt(); 5930 } 5931 } 5932 5933 // indicate all active tracks in the chain 5934 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5935 sp<Track> track = mActiveTracks[i].promote(); 5936 if (track == 0) continue; 5937 if (session == track->sessionId()) { 5938 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5939 chain->incActiveTrackCnt(); 5940 } 5941 } 5942 } 5943 5944 chain->setInBuffer(buffer, ownsBuffer); 5945 chain->setOutBuffer(mMixBuffer); 5946 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5947 // chains list in order to be processed last as it contains output stage effects 5948 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5949 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5950 // after track specific effects and before output stage 5951 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5952 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5953 // Effect chain for other sessions are inserted at beginning of effect 5954 // chains list to be processed before output mix effects. Relative order between other 5955 // sessions is not important 5956 size_t size = mEffectChains.size(); 5957 size_t i = 0; 5958 for (i = 0; i < size; i++) { 5959 if (mEffectChains[i]->sessionId() < session) break; 5960 } 5961 mEffectChains.insertAt(chain, i); 5962 checkSuspendOnAddEffectChain_l(chain); 5963 5964 return NO_ERROR; 5965} 5966 5967size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5968{ 5969 int session = chain->sessionId(); 5970 5971 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5972 5973 for (size_t i = 0; i < mEffectChains.size(); i++) { 5974 if (chain == mEffectChains[i]) { 5975 mEffectChains.removeAt(i); 5976 // detach all active tracks from the chain 5977 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5978 sp<Track> track = mActiveTracks[i].promote(); 5979 if (track == 0) continue; 5980 if (session == track->sessionId()) { 5981 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5982 chain.get(), session); 5983 chain->decActiveTrackCnt(); 5984 } 5985 } 5986 5987 // detach all tracks with same session ID from this chain 5988 for (size_t i = 0; i < mTracks.size(); ++i) { 5989 sp<Track> track = mTracks[i]; 5990 if (session == track->sessionId()) { 5991 track->setMainBuffer(mMixBuffer); 5992 chain->decTrackCnt(); 5993 } 5994 } 5995 break; 5996 } 5997 } 5998 return mEffectChains.size(); 5999} 6000 6001status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6002 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6003{ 6004 Mutex::Autolock _l(mLock); 6005 return attachAuxEffect_l(track, EffectId); 6006} 6007 6008status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6009 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6010{ 6011 status_t status = NO_ERROR; 6012 6013 if (EffectId == 0) { 6014 track->setAuxBuffer(0, NULL); 6015 } else { 6016 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6017 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6018 if (effect != 0) { 6019 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6020 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6021 } else { 6022 status = INVALID_OPERATION; 6023 } 6024 } else { 6025 status = BAD_VALUE; 6026 } 6027 } 6028 return status; 6029} 6030 6031void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6032{ 6033 for (size_t i = 0; i < mTracks.size(); ++i) { 6034 sp<Track> track = mTracks[i]; 6035 if (track->auxEffectId() == effectId) { 6036 attachAuxEffect_l(track, 0); 6037 } 6038 } 6039} 6040 6041status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6042{ 6043 // only one chain per input thread 6044 if (mEffectChains.size() != 0) { 6045 return INVALID_OPERATION; 6046 } 6047 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6048 6049 chain->setInBuffer(NULL); 6050 chain->setOutBuffer(NULL); 6051 6052 checkSuspendOnAddEffectChain_l(chain); 6053 6054 mEffectChains.add(chain); 6055 6056 return NO_ERROR; 6057} 6058 6059size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6060{ 6061 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6062 LOGW_IF(mEffectChains.size() != 1, 6063 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6064 chain.get(), mEffectChains.size(), this); 6065 if (mEffectChains.size() == 1) { 6066 mEffectChains.removeAt(0); 6067 } 6068 return 0; 6069} 6070 6071// ---------------------------------------------------------------------------- 6072// EffectModule implementation 6073// ---------------------------------------------------------------------------- 6074 6075#undef LOG_TAG 6076#define LOG_TAG "AudioFlinger::EffectModule" 6077 6078AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6079 const wp<AudioFlinger::EffectChain>& chain, 6080 effect_descriptor_t *desc, 6081 int id, 6082 int sessionId) 6083 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6084 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6085{ 6086 ALOGV("Constructor %p", this); 6087 int lStatus; 6088 sp<ThreadBase> thread = mThread.promote(); 6089 if (thread == 0) { 6090 return; 6091 } 6092 6093 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6094 6095 // create effect engine from effect factory 6096 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6097 6098 if (mStatus != NO_ERROR) { 6099 return; 6100 } 6101 lStatus = init(); 6102 if (lStatus < 0) { 6103 mStatus = lStatus; 6104 goto Error; 6105 } 6106 6107 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6108 mPinned = true; 6109 } 6110 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6111 return; 6112Error: 6113 EffectRelease(mEffectInterface); 6114 mEffectInterface = NULL; 6115 ALOGV("Constructor Error %d", mStatus); 6116} 6117 6118AudioFlinger::EffectModule::~EffectModule() 6119{ 6120 ALOGV("Destructor %p", this); 6121 if (mEffectInterface != NULL) { 6122 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6123 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6124 sp<ThreadBase> thread = mThread.promote(); 6125 if (thread != 0) { 6126 audio_stream_t *stream = thread->stream(); 6127 if (stream != NULL) { 6128 stream->remove_audio_effect(stream, mEffectInterface); 6129 } 6130 } 6131 } 6132 // release effect engine 6133 EffectRelease(mEffectInterface); 6134 } 6135} 6136 6137status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6138{ 6139 status_t status; 6140 6141 Mutex::Autolock _l(mLock); 6142 // First handle in mHandles has highest priority and controls the effect module 6143 int priority = handle->priority(); 6144 size_t size = mHandles.size(); 6145 sp<EffectHandle> h; 6146 size_t i; 6147 for (i = 0; i < size; i++) { 6148 h = mHandles[i].promote(); 6149 if (h == 0) continue; 6150 if (h->priority() <= priority) break; 6151 } 6152 // if inserted in first place, move effect control from previous owner to this handle 6153 if (i == 0) { 6154 bool enabled = false; 6155 if (h != 0) { 6156 enabled = h->enabled(); 6157 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6158 } 6159 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6160 status = NO_ERROR; 6161 } else { 6162 status = ALREADY_EXISTS; 6163 } 6164 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6165 mHandles.insertAt(handle, i); 6166 return status; 6167} 6168 6169size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6170{ 6171 Mutex::Autolock _l(mLock); 6172 size_t size = mHandles.size(); 6173 size_t i; 6174 for (i = 0; i < size; i++) { 6175 if (mHandles[i] == handle) break; 6176 } 6177 if (i == size) { 6178 return size; 6179 } 6180 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6181 6182 bool enabled = false; 6183 EffectHandle *hdl = handle.unsafe_get(); 6184 if (hdl) { 6185 ALOGV("removeHandle() unsafe_get OK"); 6186 enabled = hdl->enabled(); 6187 } 6188 mHandles.removeAt(i); 6189 size = mHandles.size(); 6190 // if removed from first place, move effect control from this handle to next in line 6191 if (i == 0 && size != 0) { 6192 sp<EffectHandle> h = mHandles[0].promote(); 6193 if (h != 0) { 6194 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6195 } 6196 } 6197 6198 // Prevent calls to process() and other functions on effect interface from now on. 6199 // The effect engine will be released by the destructor when the last strong reference on 6200 // this object is released which can happen after next process is called. 6201 if (size == 0 && !mPinned) { 6202 mState = DESTROYED; 6203 } 6204 6205 return size; 6206} 6207 6208sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6209{ 6210 Mutex::Autolock _l(mLock); 6211 sp<EffectHandle> handle; 6212 if (mHandles.size() != 0) { 6213 handle = mHandles[0].promote(); 6214 } 6215 return handle; 6216} 6217 6218void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6219{ 6220 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6221 // keep a strong reference on this EffectModule to avoid calling the 6222 // destructor before we exit 6223 sp<EffectModule> keep(this); 6224 { 6225 sp<ThreadBase> thread = mThread.promote(); 6226 if (thread != 0) { 6227 thread->disconnectEffect(keep, handle, unpiniflast); 6228 } 6229 } 6230} 6231 6232void AudioFlinger::EffectModule::updateState() { 6233 Mutex::Autolock _l(mLock); 6234 6235 switch (mState) { 6236 case RESTART: 6237 reset_l(); 6238 // FALL THROUGH 6239 6240 case STARTING: 6241 // clear auxiliary effect input buffer for next accumulation 6242 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6243 memset(mConfig.inputCfg.buffer.raw, 6244 0, 6245 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6246 } 6247 start_l(); 6248 mState = ACTIVE; 6249 break; 6250 case STOPPING: 6251 stop_l(); 6252 mDisableWaitCnt = mMaxDisableWaitCnt; 6253 mState = STOPPED; 6254 break; 6255 case STOPPED: 6256 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6257 // turn off sequence. 6258 if (--mDisableWaitCnt == 0) { 6259 reset_l(); 6260 mState = IDLE; 6261 } 6262 break; 6263 default: //IDLE , ACTIVE, DESTROYED 6264 break; 6265 } 6266} 6267 6268void AudioFlinger::EffectModule::process() 6269{ 6270 Mutex::Autolock _l(mLock); 6271 6272 if (mState == DESTROYED || mEffectInterface == NULL || 6273 mConfig.inputCfg.buffer.raw == NULL || 6274 mConfig.outputCfg.buffer.raw == NULL) { 6275 return; 6276 } 6277 6278 if (isProcessEnabled()) { 6279 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6280 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6281 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6282 mConfig.inputCfg.buffer.s32, 6283 mConfig.inputCfg.buffer.frameCount/2); 6284 } 6285 6286 // do the actual processing in the effect engine 6287 int ret = (*mEffectInterface)->process(mEffectInterface, 6288 &mConfig.inputCfg.buffer, 6289 &mConfig.outputCfg.buffer); 6290 6291 // force transition to IDLE state when engine is ready 6292 if (mState == STOPPED && ret == -ENODATA) { 6293 mDisableWaitCnt = 1; 6294 } 6295 6296 // clear auxiliary effect input buffer for next accumulation 6297 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6298 memset(mConfig.inputCfg.buffer.raw, 0, 6299 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6300 } 6301 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6302 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6303 // If an insert effect is idle and input buffer is different from output buffer, 6304 // accumulate input onto output 6305 sp<EffectChain> chain = mChain.promote(); 6306 if (chain != 0 && chain->activeTrackCnt() != 0) { 6307 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6308 int16_t *in = mConfig.inputCfg.buffer.s16; 6309 int16_t *out = mConfig.outputCfg.buffer.s16; 6310 for (size_t i = 0; i < frameCnt; i++) { 6311 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6312 } 6313 } 6314 } 6315} 6316 6317void AudioFlinger::EffectModule::reset_l() 6318{ 6319 if (mEffectInterface == NULL) { 6320 return; 6321 } 6322 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6323} 6324 6325status_t AudioFlinger::EffectModule::configure() 6326{ 6327 uint32_t channels; 6328 if (mEffectInterface == NULL) { 6329 return NO_INIT; 6330 } 6331 6332 sp<ThreadBase> thread = mThread.promote(); 6333 if (thread == 0) { 6334 return DEAD_OBJECT; 6335 } 6336 6337 // TODO: handle configuration of effects replacing track process 6338 if (thread->channelCount() == 1) { 6339 channels = AUDIO_CHANNEL_OUT_MONO; 6340 } else { 6341 channels = AUDIO_CHANNEL_OUT_STEREO; 6342 } 6343 6344 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6345 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6346 } else { 6347 mConfig.inputCfg.channels = channels; 6348 } 6349 mConfig.outputCfg.channels = channels; 6350 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6351 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6352 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6353 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6354 mConfig.inputCfg.bufferProvider.cookie = NULL; 6355 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6356 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6357 mConfig.outputCfg.bufferProvider.cookie = NULL; 6358 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6359 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6360 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6361 // Insert effect: 6362 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6363 // always overwrites output buffer: input buffer == output buffer 6364 // - in other sessions: 6365 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6366 // other effect: overwrites output buffer: input buffer == output buffer 6367 // Auxiliary effect: 6368 // accumulates in output buffer: input buffer != output buffer 6369 // Therefore: accumulate <=> input buffer != output buffer 6370 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6371 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6372 } else { 6373 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6374 } 6375 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6376 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6377 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6378 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6379 6380 ALOGV("configure() %p thread %p buffer %p framecount %d", 6381 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6382 6383 status_t cmdStatus; 6384 uint32_t size = sizeof(int); 6385 status_t status = (*mEffectInterface)->command(mEffectInterface, 6386 EFFECT_CMD_SET_CONFIG, 6387 sizeof(effect_config_t), 6388 &mConfig, 6389 &size, 6390 &cmdStatus); 6391 if (status == 0) { 6392 status = cmdStatus; 6393 } 6394 6395 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6396 (1000 * mConfig.outputCfg.buffer.frameCount); 6397 6398 return status; 6399} 6400 6401status_t AudioFlinger::EffectModule::init() 6402{ 6403 Mutex::Autolock _l(mLock); 6404 if (mEffectInterface == NULL) { 6405 return NO_INIT; 6406 } 6407 status_t cmdStatus; 6408 uint32_t size = sizeof(status_t); 6409 status_t status = (*mEffectInterface)->command(mEffectInterface, 6410 EFFECT_CMD_INIT, 6411 0, 6412 NULL, 6413 &size, 6414 &cmdStatus); 6415 if (status == 0) { 6416 status = cmdStatus; 6417 } 6418 return status; 6419} 6420 6421status_t AudioFlinger::EffectModule::start() 6422{ 6423 Mutex::Autolock _l(mLock); 6424 return start_l(); 6425} 6426 6427status_t AudioFlinger::EffectModule::start_l() 6428{ 6429 if (mEffectInterface == NULL) { 6430 return NO_INIT; 6431 } 6432 status_t cmdStatus; 6433 uint32_t size = sizeof(status_t); 6434 status_t status = (*mEffectInterface)->command(mEffectInterface, 6435 EFFECT_CMD_ENABLE, 6436 0, 6437 NULL, 6438 &size, 6439 &cmdStatus); 6440 if (status == 0) { 6441 status = cmdStatus; 6442 } 6443 if (status == 0 && 6444 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6445 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6446 sp<ThreadBase> thread = mThread.promote(); 6447 if (thread != 0) { 6448 audio_stream_t *stream = thread->stream(); 6449 if (stream != NULL) { 6450 stream->add_audio_effect(stream, mEffectInterface); 6451 } 6452 } 6453 } 6454 return status; 6455} 6456 6457status_t AudioFlinger::EffectModule::stop() 6458{ 6459 Mutex::Autolock _l(mLock); 6460 return stop_l(); 6461} 6462 6463status_t AudioFlinger::EffectModule::stop_l() 6464{ 6465 if (mEffectInterface == NULL) { 6466 return NO_INIT; 6467 } 6468 status_t cmdStatus; 6469 uint32_t size = sizeof(status_t); 6470 status_t status = (*mEffectInterface)->command(mEffectInterface, 6471 EFFECT_CMD_DISABLE, 6472 0, 6473 NULL, 6474 &size, 6475 &cmdStatus); 6476 if (status == 0) { 6477 status = cmdStatus; 6478 } 6479 if (status == 0 && 6480 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6481 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6482 sp<ThreadBase> thread = mThread.promote(); 6483 if (thread != 0) { 6484 audio_stream_t *stream = thread->stream(); 6485 if (stream != NULL) { 6486 stream->remove_audio_effect(stream, mEffectInterface); 6487 } 6488 } 6489 } 6490 return status; 6491} 6492 6493status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6494 uint32_t cmdSize, 6495 void *pCmdData, 6496 uint32_t *replySize, 6497 void *pReplyData) 6498{ 6499 Mutex::Autolock _l(mLock); 6500// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6501 6502 if (mState == DESTROYED || mEffectInterface == NULL) { 6503 return NO_INIT; 6504 } 6505 status_t status = (*mEffectInterface)->command(mEffectInterface, 6506 cmdCode, 6507 cmdSize, 6508 pCmdData, 6509 replySize, 6510 pReplyData); 6511 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6512 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6513 for (size_t i = 1; i < mHandles.size(); i++) { 6514 sp<EffectHandle> h = mHandles[i].promote(); 6515 if (h != 0) { 6516 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6517 } 6518 } 6519 } 6520 return status; 6521} 6522 6523status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6524{ 6525 6526 Mutex::Autolock _l(mLock); 6527 ALOGV("setEnabled %p enabled %d", this, enabled); 6528 6529 if (enabled != isEnabled()) { 6530 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6531 if (enabled && status != NO_ERROR) { 6532 return status; 6533 } 6534 6535 switch (mState) { 6536 // going from disabled to enabled 6537 case IDLE: 6538 mState = STARTING; 6539 break; 6540 case STOPPED: 6541 mState = RESTART; 6542 break; 6543 case STOPPING: 6544 mState = ACTIVE; 6545 break; 6546 6547 // going from enabled to disabled 6548 case RESTART: 6549 mState = STOPPED; 6550 break; 6551 case STARTING: 6552 mState = IDLE; 6553 break; 6554 case ACTIVE: 6555 mState = STOPPING; 6556 break; 6557 case DESTROYED: 6558 return NO_ERROR; // simply ignore as we are being destroyed 6559 } 6560 for (size_t i = 1; i < mHandles.size(); i++) { 6561 sp<EffectHandle> h = mHandles[i].promote(); 6562 if (h != 0) { 6563 h->setEnabled(enabled); 6564 } 6565 } 6566 } 6567 return NO_ERROR; 6568} 6569 6570bool AudioFlinger::EffectModule::isEnabled() 6571{ 6572 switch (mState) { 6573 case RESTART: 6574 case STARTING: 6575 case ACTIVE: 6576 return true; 6577 case IDLE: 6578 case STOPPING: 6579 case STOPPED: 6580 case DESTROYED: 6581 default: 6582 return false; 6583 } 6584} 6585 6586bool AudioFlinger::EffectModule::isProcessEnabled() 6587{ 6588 switch (mState) { 6589 case RESTART: 6590 case ACTIVE: 6591 case STOPPING: 6592 case STOPPED: 6593 return true; 6594 case IDLE: 6595 case STARTING: 6596 case DESTROYED: 6597 default: 6598 return false; 6599 } 6600} 6601 6602status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6603{ 6604 Mutex::Autolock _l(mLock); 6605 status_t status = NO_ERROR; 6606 6607 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6608 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6609 if (isProcessEnabled() && 6610 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6611 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6612 status_t cmdStatus; 6613 uint32_t volume[2]; 6614 uint32_t *pVolume = NULL; 6615 uint32_t size = sizeof(volume); 6616 volume[0] = *left; 6617 volume[1] = *right; 6618 if (controller) { 6619 pVolume = volume; 6620 } 6621 status = (*mEffectInterface)->command(mEffectInterface, 6622 EFFECT_CMD_SET_VOLUME, 6623 size, 6624 volume, 6625 &size, 6626 pVolume); 6627 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6628 *left = volume[0]; 6629 *right = volume[1]; 6630 } 6631 } 6632 return status; 6633} 6634 6635status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6636{ 6637 Mutex::Autolock _l(mLock); 6638 status_t status = NO_ERROR; 6639 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6640 // audio pre processing modules on RecordThread can receive both output and 6641 // input device indication in the same call 6642 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6643 if (dev) { 6644 status_t cmdStatus; 6645 uint32_t size = sizeof(status_t); 6646 6647 status = (*mEffectInterface)->command(mEffectInterface, 6648 EFFECT_CMD_SET_DEVICE, 6649 sizeof(uint32_t), 6650 &dev, 6651 &size, 6652 &cmdStatus); 6653 if (status == NO_ERROR) { 6654 status = cmdStatus; 6655 } 6656 } 6657 dev = device & AUDIO_DEVICE_IN_ALL; 6658 if (dev) { 6659 status_t cmdStatus; 6660 uint32_t size = sizeof(status_t); 6661 6662 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6663 EFFECT_CMD_SET_INPUT_DEVICE, 6664 sizeof(uint32_t), 6665 &dev, 6666 &size, 6667 &cmdStatus); 6668 if (status2 == NO_ERROR) { 6669 status2 = cmdStatus; 6670 } 6671 if (status == NO_ERROR) { 6672 status = status2; 6673 } 6674 } 6675 } 6676 return status; 6677} 6678 6679status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 6680{ 6681 Mutex::Autolock _l(mLock); 6682 status_t status = NO_ERROR; 6683 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6684 status_t cmdStatus; 6685 uint32_t size = sizeof(status_t); 6686 status = (*mEffectInterface)->command(mEffectInterface, 6687 EFFECT_CMD_SET_AUDIO_MODE, 6688 sizeof(int), 6689 &mode, 6690 &size, 6691 &cmdStatus); 6692 if (status == NO_ERROR) { 6693 status = cmdStatus; 6694 } 6695 } 6696 return status; 6697} 6698 6699void AudioFlinger::EffectModule::setSuspended(bool suspended) 6700{ 6701 Mutex::Autolock _l(mLock); 6702 mSuspended = suspended; 6703} 6704bool AudioFlinger::EffectModule::suspended() 6705{ 6706 Mutex::Autolock _l(mLock); 6707 return mSuspended; 6708} 6709 6710status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6711{ 6712 const size_t SIZE = 256; 6713 char buffer[SIZE]; 6714 String8 result; 6715 6716 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6717 result.append(buffer); 6718 6719 bool locked = tryLock(mLock); 6720 // failed to lock - AudioFlinger is probably deadlocked 6721 if (!locked) { 6722 result.append("\t\tCould not lock Fx mutex:\n"); 6723 } 6724 6725 result.append("\t\tSession Status State Engine:\n"); 6726 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6727 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6728 result.append(buffer); 6729 6730 result.append("\t\tDescriptor:\n"); 6731 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6732 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6733 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6734 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6735 result.append(buffer); 6736 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6737 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6738 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6739 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6740 result.append(buffer); 6741 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6742 mDescriptor.apiVersion, 6743 mDescriptor.flags); 6744 result.append(buffer); 6745 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6746 mDescriptor.name); 6747 result.append(buffer); 6748 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6749 mDescriptor.implementor); 6750 result.append(buffer); 6751 6752 result.append("\t\t- Input configuration:\n"); 6753 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6754 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6755 (uint32_t)mConfig.inputCfg.buffer.raw, 6756 mConfig.inputCfg.buffer.frameCount, 6757 mConfig.inputCfg.samplingRate, 6758 mConfig.inputCfg.channels, 6759 mConfig.inputCfg.format); 6760 result.append(buffer); 6761 6762 result.append("\t\t- Output configuration:\n"); 6763 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6764 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6765 (uint32_t)mConfig.outputCfg.buffer.raw, 6766 mConfig.outputCfg.buffer.frameCount, 6767 mConfig.outputCfg.samplingRate, 6768 mConfig.outputCfg.channels, 6769 mConfig.outputCfg.format); 6770 result.append(buffer); 6771 6772 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6773 result.append(buffer); 6774 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6775 for (size_t i = 0; i < mHandles.size(); ++i) { 6776 sp<EffectHandle> handle = mHandles[i].promote(); 6777 if (handle != 0) { 6778 handle->dump(buffer, SIZE); 6779 result.append(buffer); 6780 } 6781 } 6782 6783 result.append("\n"); 6784 6785 write(fd, result.string(), result.length()); 6786 6787 if (locked) { 6788 mLock.unlock(); 6789 } 6790 6791 return NO_ERROR; 6792} 6793 6794// ---------------------------------------------------------------------------- 6795// EffectHandle implementation 6796// ---------------------------------------------------------------------------- 6797 6798#undef LOG_TAG 6799#define LOG_TAG "AudioFlinger::EffectHandle" 6800 6801AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6802 const sp<AudioFlinger::Client>& client, 6803 const sp<IEffectClient>& effectClient, 6804 int32_t priority) 6805 : BnEffect(), 6806 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6807 mPriority(priority), mHasControl(false), mEnabled(false) 6808{ 6809 ALOGV("constructor %p", this); 6810 6811 if (client == 0) { 6812 return; 6813 } 6814 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6815 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6816 if (mCblkMemory != 0) { 6817 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6818 6819 if (mCblk) { 6820 new(mCblk) effect_param_cblk_t(); 6821 mBuffer = (uint8_t *)mCblk + bufOffset; 6822 } 6823 } else { 6824 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6825 return; 6826 } 6827} 6828 6829AudioFlinger::EffectHandle::~EffectHandle() 6830{ 6831 ALOGV("Destructor %p", this); 6832 disconnect(false); 6833 ALOGV("Destructor DONE %p", this); 6834} 6835 6836status_t AudioFlinger::EffectHandle::enable() 6837{ 6838 ALOGV("enable %p", this); 6839 if (!mHasControl) return INVALID_OPERATION; 6840 if (mEffect == 0) return DEAD_OBJECT; 6841 6842 if (mEnabled) { 6843 return NO_ERROR; 6844 } 6845 6846 mEnabled = true; 6847 6848 sp<ThreadBase> thread = mEffect->thread().promote(); 6849 if (thread != 0) { 6850 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6851 } 6852 6853 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6854 if (mEffect->suspended()) { 6855 return NO_ERROR; 6856 } 6857 6858 status_t status = mEffect->setEnabled(true); 6859 if (status != NO_ERROR) { 6860 if (thread != 0) { 6861 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6862 } 6863 mEnabled = false; 6864 } 6865 return status; 6866} 6867 6868status_t AudioFlinger::EffectHandle::disable() 6869{ 6870 ALOGV("disable %p", this); 6871 if (!mHasControl) return INVALID_OPERATION; 6872 if (mEffect == 0) return DEAD_OBJECT; 6873 6874 if (!mEnabled) { 6875 return NO_ERROR; 6876 } 6877 mEnabled = false; 6878 6879 if (mEffect->suspended()) { 6880 return NO_ERROR; 6881 } 6882 6883 status_t status = mEffect->setEnabled(false); 6884 6885 sp<ThreadBase> thread = mEffect->thread().promote(); 6886 if (thread != 0) { 6887 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6888 } 6889 6890 return status; 6891} 6892 6893void AudioFlinger::EffectHandle::disconnect() 6894{ 6895 disconnect(true); 6896} 6897 6898void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6899{ 6900 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6901 if (mEffect == 0) { 6902 return; 6903 } 6904 mEffect->disconnect(this, unpiniflast); 6905 6906 if (mHasControl && mEnabled) { 6907 sp<ThreadBase> thread = mEffect->thread().promote(); 6908 if (thread != 0) { 6909 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6910 } 6911 } 6912 6913 // release sp on module => module destructor can be called now 6914 mEffect.clear(); 6915 if (mClient != 0) { 6916 if (mCblk) { 6917 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6918 } 6919 mCblkMemory.clear(); // and free the shared memory 6920 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6921 mClient.clear(); 6922 } 6923} 6924 6925status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6926 uint32_t cmdSize, 6927 void *pCmdData, 6928 uint32_t *replySize, 6929 void *pReplyData) 6930{ 6931// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6932// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6933 6934 // only get parameter command is permitted for applications not controlling the effect 6935 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6936 return INVALID_OPERATION; 6937 } 6938 if (mEffect == 0) return DEAD_OBJECT; 6939 if (mClient == 0) return INVALID_OPERATION; 6940 6941 // handle commands that are not forwarded transparently to effect engine 6942 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6943 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6944 // no risk to block the whole media server process or mixer threads is we are stuck here 6945 Mutex::Autolock _l(mCblk->lock); 6946 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6947 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6948 mCblk->serverIndex = 0; 6949 mCblk->clientIndex = 0; 6950 return BAD_VALUE; 6951 } 6952 status_t status = NO_ERROR; 6953 while (mCblk->serverIndex < mCblk->clientIndex) { 6954 int reply; 6955 uint32_t rsize = sizeof(int); 6956 int *p = (int *)(mBuffer + mCblk->serverIndex); 6957 int size = *p++; 6958 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6959 LOGW("command(): invalid parameter block size"); 6960 break; 6961 } 6962 effect_param_t *param = (effect_param_t *)p; 6963 if (param->psize == 0 || param->vsize == 0) { 6964 LOGW("command(): null parameter or value size"); 6965 mCblk->serverIndex += size; 6966 continue; 6967 } 6968 uint32_t psize = sizeof(effect_param_t) + 6969 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6970 param->vsize; 6971 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6972 psize, 6973 p, 6974 &rsize, 6975 &reply); 6976 // stop at first error encountered 6977 if (ret != NO_ERROR) { 6978 status = ret; 6979 *(int *)pReplyData = reply; 6980 break; 6981 } else if (reply != NO_ERROR) { 6982 *(int *)pReplyData = reply; 6983 break; 6984 } 6985 mCblk->serverIndex += size; 6986 } 6987 mCblk->serverIndex = 0; 6988 mCblk->clientIndex = 0; 6989 return status; 6990 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6991 *(int *)pReplyData = NO_ERROR; 6992 return enable(); 6993 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6994 *(int *)pReplyData = NO_ERROR; 6995 return disable(); 6996 } 6997 6998 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6999} 7000 7001sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7002 return mCblkMemory; 7003} 7004 7005void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7006{ 7007 ALOGV("setControl %p control %d", this, hasControl); 7008 7009 mHasControl = hasControl; 7010 mEnabled = enabled; 7011 7012 if (signal && mEffectClient != 0) { 7013 mEffectClient->controlStatusChanged(hasControl); 7014 } 7015} 7016 7017void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7018 uint32_t cmdSize, 7019 void *pCmdData, 7020 uint32_t replySize, 7021 void *pReplyData) 7022{ 7023 if (mEffectClient != 0) { 7024 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7025 } 7026} 7027 7028 7029 7030void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7031{ 7032 if (mEffectClient != 0) { 7033 mEffectClient->enableStatusChanged(enabled); 7034 } 7035} 7036 7037status_t AudioFlinger::EffectHandle::onTransact( 7038 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7039{ 7040 return BnEffect::onTransact(code, data, reply, flags); 7041} 7042 7043 7044void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7045{ 7046 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7047 7048 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7049 (mClient == NULL) ? getpid() : mClient->pid(), 7050 mPriority, 7051 mHasControl, 7052 !locked, 7053 mCblk ? mCblk->clientIndex : 0, 7054 mCblk ? mCblk->serverIndex : 0 7055 ); 7056 7057 if (locked) { 7058 mCblk->lock.unlock(); 7059 } 7060} 7061 7062#undef LOG_TAG 7063#define LOG_TAG "AudioFlinger::EffectChain" 7064 7065AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7066 int sessionId) 7067 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7068 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7069 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7070{ 7071 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7072 sp<ThreadBase> thread = mThread.promote(); 7073 if (thread == 0) { 7074 return; 7075 } 7076 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7077 thread->frameCount(); 7078} 7079 7080AudioFlinger::EffectChain::~EffectChain() 7081{ 7082 if (mOwnInBuffer) { 7083 delete mInBuffer; 7084 } 7085 7086} 7087 7088// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7089sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7090{ 7091 sp<EffectModule> effect; 7092 size_t size = mEffects.size(); 7093 7094 for (size_t i = 0; i < size; i++) { 7095 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7096 effect = mEffects[i]; 7097 break; 7098 } 7099 } 7100 return effect; 7101} 7102 7103// getEffectFromId_l() must be called with ThreadBase::mLock held 7104sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7105{ 7106 sp<EffectModule> effect; 7107 size_t size = mEffects.size(); 7108 7109 for (size_t i = 0; i < size; i++) { 7110 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7111 if (id == 0 || mEffects[i]->id() == id) { 7112 effect = mEffects[i]; 7113 break; 7114 } 7115 } 7116 return effect; 7117} 7118 7119// getEffectFromType_l() must be called with ThreadBase::mLock held 7120sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7121 const effect_uuid_t *type) 7122{ 7123 sp<EffectModule> effect; 7124 size_t size = mEffects.size(); 7125 7126 for (size_t i = 0; i < size; i++) { 7127 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7128 effect = mEffects[i]; 7129 break; 7130 } 7131 } 7132 return effect; 7133} 7134 7135// Must be called with EffectChain::mLock locked 7136void AudioFlinger::EffectChain::process_l() 7137{ 7138 sp<ThreadBase> thread = mThread.promote(); 7139 if (thread == 0) { 7140 LOGW("process_l(): cannot promote mixer thread"); 7141 return; 7142 } 7143 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7144 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7145 // always process effects unless no more tracks are on the session and the effect tail 7146 // has been rendered 7147 bool doProcess = true; 7148 if (!isGlobalSession) { 7149 bool tracksOnSession = (trackCnt() != 0); 7150 7151 if (!tracksOnSession && mTailBufferCount == 0) { 7152 doProcess = false; 7153 } 7154 7155 if (activeTrackCnt() == 0) { 7156 // if no track is active and the effect tail has not been rendered, 7157 // the input buffer must be cleared here as the mixer process will not do it 7158 if (tracksOnSession || mTailBufferCount > 0) { 7159 size_t numSamples = thread->frameCount() * thread->channelCount(); 7160 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7161 if (mTailBufferCount > 0) { 7162 mTailBufferCount--; 7163 } 7164 } 7165 } 7166 } 7167 7168 size_t size = mEffects.size(); 7169 if (doProcess) { 7170 for (size_t i = 0; i < size; i++) { 7171 mEffects[i]->process(); 7172 } 7173 } 7174 for (size_t i = 0; i < size; i++) { 7175 mEffects[i]->updateState(); 7176 } 7177} 7178 7179// addEffect_l() must be called with PlaybackThread::mLock held 7180status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7181{ 7182 effect_descriptor_t desc = effect->desc(); 7183 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7184 7185 Mutex::Autolock _l(mLock); 7186 effect->setChain(this); 7187 sp<ThreadBase> thread = mThread.promote(); 7188 if (thread == 0) { 7189 return NO_INIT; 7190 } 7191 effect->setThread(thread); 7192 7193 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7194 // Auxiliary effects are inserted at the beginning of mEffects vector as 7195 // they are processed first and accumulated in chain input buffer 7196 mEffects.insertAt(effect, 0); 7197 7198 // the input buffer for auxiliary effect contains mono samples in 7199 // 32 bit format. This is to avoid saturation in AudoMixer 7200 // accumulation stage. Saturation is done in EffectModule::process() before 7201 // calling the process in effect engine 7202 size_t numSamples = thread->frameCount(); 7203 int32_t *buffer = new int32_t[numSamples]; 7204 memset(buffer, 0, numSamples * sizeof(int32_t)); 7205 effect->setInBuffer((int16_t *)buffer); 7206 // auxiliary effects output samples to chain input buffer for further processing 7207 // by insert effects 7208 effect->setOutBuffer(mInBuffer); 7209 } else { 7210 // Insert effects are inserted at the end of mEffects vector as they are processed 7211 // after track and auxiliary effects. 7212 // Insert effect order as a function of indicated preference: 7213 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7214 // another effect is present 7215 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7216 // last effect claiming first position 7217 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7218 // first effect claiming last position 7219 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7220 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7221 // already present 7222 7223 int size = (int)mEffects.size(); 7224 int idx_insert = size; 7225 int idx_insert_first = -1; 7226 int idx_insert_last = -1; 7227 7228 for (int i = 0; i < size; i++) { 7229 effect_descriptor_t d = mEffects[i]->desc(); 7230 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7231 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7232 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7233 // check invalid effect chaining combinations 7234 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7235 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7236 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7237 return INVALID_OPERATION; 7238 } 7239 // remember position of first insert effect and by default 7240 // select this as insert position for new effect 7241 if (idx_insert == size) { 7242 idx_insert = i; 7243 } 7244 // remember position of last insert effect claiming 7245 // first position 7246 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7247 idx_insert_first = i; 7248 } 7249 // remember position of first insert effect claiming 7250 // last position 7251 if (iPref == EFFECT_FLAG_INSERT_LAST && 7252 idx_insert_last == -1) { 7253 idx_insert_last = i; 7254 } 7255 } 7256 } 7257 7258 // modify idx_insert from first position if needed 7259 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7260 if (idx_insert_last != -1) { 7261 idx_insert = idx_insert_last; 7262 } else { 7263 idx_insert = size; 7264 } 7265 } else { 7266 if (idx_insert_first != -1) { 7267 idx_insert = idx_insert_first + 1; 7268 } 7269 } 7270 7271 // always read samples from chain input buffer 7272 effect->setInBuffer(mInBuffer); 7273 7274 // if last effect in the chain, output samples to chain 7275 // output buffer, otherwise to chain input buffer 7276 if (idx_insert == size) { 7277 if (idx_insert != 0) { 7278 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7279 mEffects[idx_insert-1]->configure(); 7280 } 7281 effect->setOutBuffer(mOutBuffer); 7282 } else { 7283 effect->setOutBuffer(mInBuffer); 7284 } 7285 mEffects.insertAt(effect, idx_insert); 7286 7287 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7288 } 7289 effect->configure(); 7290 return NO_ERROR; 7291} 7292 7293// removeEffect_l() must be called with PlaybackThread::mLock held 7294size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7295{ 7296 Mutex::Autolock _l(mLock); 7297 int size = (int)mEffects.size(); 7298 int i; 7299 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7300 7301 for (i = 0; i < size; i++) { 7302 if (effect == mEffects[i]) { 7303 // calling stop here will remove pre-processing effect from the audio HAL. 7304 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7305 // the middle of a read from audio HAL 7306 if (mEffects[i]->state() == EffectModule::ACTIVE || 7307 mEffects[i]->state() == EffectModule::STOPPING) { 7308 mEffects[i]->stop(); 7309 } 7310 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7311 delete[] effect->inBuffer(); 7312 } else { 7313 if (i == size - 1 && i != 0) { 7314 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7315 mEffects[i - 1]->configure(); 7316 } 7317 } 7318 mEffects.removeAt(i); 7319 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7320 break; 7321 } 7322 } 7323 7324 return mEffects.size(); 7325} 7326 7327// setDevice_l() must be called with PlaybackThread::mLock held 7328void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7329{ 7330 size_t size = mEffects.size(); 7331 for (size_t i = 0; i < size; i++) { 7332 mEffects[i]->setDevice(device); 7333 } 7334} 7335 7336// setMode_l() must be called with PlaybackThread::mLock held 7337void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 7338{ 7339 size_t size = mEffects.size(); 7340 for (size_t i = 0; i < size; i++) { 7341 mEffects[i]->setMode(mode); 7342 } 7343} 7344 7345// setVolume_l() must be called with PlaybackThread::mLock held 7346bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7347{ 7348 uint32_t newLeft = *left; 7349 uint32_t newRight = *right; 7350 bool hasControl = false; 7351 int ctrlIdx = -1; 7352 size_t size = mEffects.size(); 7353 7354 // first update volume controller 7355 for (size_t i = size; i > 0; i--) { 7356 if (mEffects[i - 1]->isProcessEnabled() && 7357 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7358 ctrlIdx = i - 1; 7359 hasControl = true; 7360 break; 7361 } 7362 } 7363 7364 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7365 if (hasControl) { 7366 *left = mNewLeftVolume; 7367 *right = mNewRightVolume; 7368 } 7369 return hasControl; 7370 } 7371 7372 mVolumeCtrlIdx = ctrlIdx; 7373 mLeftVolume = newLeft; 7374 mRightVolume = newRight; 7375 7376 // second get volume update from volume controller 7377 if (ctrlIdx >= 0) { 7378 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7379 mNewLeftVolume = newLeft; 7380 mNewRightVolume = newRight; 7381 } 7382 // then indicate volume to all other effects in chain. 7383 // Pass altered volume to effects before volume controller 7384 // and requested volume to effects after controller 7385 uint32_t lVol = newLeft; 7386 uint32_t rVol = newRight; 7387 7388 for (size_t i = 0; i < size; i++) { 7389 if ((int)i == ctrlIdx) continue; 7390 // this also works for ctrlIdx == -1 when there is no volume controller 7391 if ((int)i > ctrlIdx) { 7392 lVol = *left; 7393 rVol = *right; 7394 } 7395 mEffects[i]->setVolume(&lVol, &rVol, false); 7396 } 7397 *left = newLeft; 7398 *right = newRight; 7399 7400 return hasControl; 7401} 7402 7403status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7404{ 7405 const size_t SIZE = 256; 7406 char buffer[SIZE]; 7407 String8 result; 7408 7409 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7410 result.append(buffer); 7411 7412 bool locked = tryLock(mLock); 7413 // failed to lock - AudioFlinger is probably deadlocked 7414 if (!locked) { 7415 result.append("\tCould not lock mutex:\n"); 7416 } 7417 7418 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7419 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7420 mEffects.size(), 7421 (uint32_t)mInBuffer, 7422 (uint32_t)mOutBuffer, 7423 mActiveTrackCnt); 7424 result.append(buffer); 7425 write(fd, result.string(), result.size()); 7426 7427 for (size_t i = 0; i < mEffects.size(); ++i) { 7428 sp<EffectModule> effect = mEffects[i]; 7429 if (effect != 0) { 7430 effect->dump(fd, args); 7431 } 7432 } 7433 7434 if (locked) { 7435 mLock.unlock(); 7436 } 7437 7438 return NO_ERROR; 7439} 7440 7441// must be called with ThreadBase::mLock held 7442void AudioFlinger::EffectChain::setEffectSuspended_l( 7443 const effect_uuid_t *type, bool suspend) 7444{ 7445 sp<SuspendedEffectDesc> desc; 7446 // use effect type UUID timelow as key as there is no real risk of identical 7447 // timeLow fields among effect type UUIDs. 7448 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7449 if (suspend) { 7450 if (index >= 0) { 7451 desc = mSuspendedEffects.valueAt(index); 7452 } else { 7453 desc = new SuspendedEffectDesc(); 7454 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7455 mSuspendedEffects.add(type->timeLow, desc); 7456 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7457 } 7458 if (desc->mRefCount++ == 0) { 7459 sp<EffectModule> effect = getEffectIfEnabled(type); 7460 if (effect != 0) { 7461 desc->mEffect = effect; 7462 effect->setSuspended(true); 7463 effect->setEnabled(false); 7464 } 7465 } 7466 } else { 7467 if (index < 0) { 7468 return; 7469 } 7470 desc = mSuspendedEffects.valueAt(index); 7471 if (desc->mRefCount <= 0) { 7472 LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7473 desc->mRefCount = 1; 7474 } 7475 if (--desc->mRefCount == 0) { 7476 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7477 if (desc->mEffect != 0) { 7478 sp<EffectModule> effect = desc->mEffect.promote(); 7479 if (effect != 0) { 7480 effect->setSuspended(false); 7481 sp<EffectHandle> handle = effect->controlHandle(); 7482 if (handle != 0) { 7483 effect->setEnabled(handle->enabled()); 7484 } 7485 } 7486 desc->mEffect.clear(); 7487 } 7488 mSuspendedEffects.removeItemsAt(index); 7489 } 7490 } 7491} 7492 7493// must be called with ThreadBase::mLock held 7494void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7495{ 7496 sp<SuspendedEffectDesc> desc; 7497 7498 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7499 if (suspend) { 7500 if (index >= 0) { 7501 desc = mSuspendedEffects.valueAt(index); 7502 } else { 7503 desc = new SuspendedEffectDesc(); 7504 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7505 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7506 } 7507 if (desc->mRefCount++ == 0) { 7508 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7509 for (size_t i = 0; i < effects.size(); i++) { 7510 setEffectSuspended_l(&effects[i]->desc().type, true); 7511 } 7512 } 7513 } else { 7514 if (index < 0) { 7515 return; 7516 } 7517 desc = mSuspendedEffects.valueAt(index); 7518 if (desc->mRefCount <= 0) { 7519 LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7520 desc->mRefCount = 1; 7521 } 7522 if (--desc->mRefCount == 0) { 7523 Vector<const effect_uuid_t *> types; 7524 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7525 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7526 continue; 7527 } 7528 types.add(&mSuspendedEffects.valueAt(i)->mType); 7529 } 7530 for (size_t i = 0; i < types.size(); i++) { 7531 setEffectSuspended_l(types[i], false); 7532 } 7533 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7534 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7535 } 7536 } 7537} 7538 7539 7540// The volume effect is used for automated tests only 7541#ifndef OPENSL_ES_H_ 7542static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7543 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7544const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7545#endif //OPENSL_ES_H_ 7546 7547bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7548{ 7549 // auxiliary effects and visualizer are never suspended on output mix 7550 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7551 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7552 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7553 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7554 return false; 7555 } 7556 return true; 7557} 7558 7559Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7560{ 7561 Vector< sp<EffectModule> > effects; 7562 for (size_t i = 0; i < mEffects.size(); i++) { 7563 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7564 continue; 7565 } 7566 effects.add(mEffects[i]); 7567 } 7568 return effects; 7569} 7570 7571sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7572 const effect_uuid_t *type) 7573{ 7574 sp<EffectModule> effect; 7575 effect = getEffectFromType_l(type); 7576 if (effect != 0 && !effect->isEnabled()) { 7577 effect.clear(); 7578 } 7579 return effect; 7580} 7581 7582void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7583 bool enabled) 7584{ 7585 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7586 if (enabled) { 7587 if (index < 0) { 7588 // if the effect is not suspend check if all effects are suspended 7589 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7590 if (index < 0) { 7591 return; 7592 } 7593 if (!isEffectEligibleForSuspend(effect->desc())) { 7594 return; 7595 } 7596 setEffectSuspended_l(&effect->desc().type, enabled); 7597 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7598 if (index < 0) { 7599 LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7600 return; 7601 } 7602 } 7603 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7604 effect->desc().type.timeLow); 7605 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7606 // if effect is requested to suspended but was not yet enabled, supend it now. 7607 if (desc->mEffect == 0) { 7608 desc->mEffect = effect; 7609 effect->setEnabled(false); 7610 effect->setSuspended(true); 7611 } 7612 } else { 7613 if (index < 0) { 7614 return; 7615 } 7616 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7617 effect->desc().type.timeLow); 7618 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7619 desc->mEffect.clear(); 7620 effect->setSuspended(false); 7621 } 7622} 7623 7624#undef LOG_TAG 7625#define LOG_TAG "AudioFlinger" 7626 7627// ---------------------------------------------------------------------------- 7628 7629status_t AudioFlinger::onTransact( 7630 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7631{ 7632 return BnAudioFlinger::onTransact(code, data, reply, flags); 7633} 7634 7635}; // namespace android 7636