AudioFlinger.cpp revision b6333aa8317ce5162ab006c4baed6b0890936dc7
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#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.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 uint32_t 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) ALOGE("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) ALOGE("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<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 121 if (service == NULL) { 122 // it already logged 123 return; 124 } 125 126 service->addBatteryData(params); 127} 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 165 mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 166 mMode(AUDIO_MODE_INVALID), 167 mBtNrecIsOff(false) 168{ 169} 170 171void AudioFlinger::onFirstRef() 172{ 173 int rc = 0; 174 175 Mutex::Autolock _l(mLock); 176 177 /* TODO: move all this work into an Init() function */ 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 ALOGI("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 ALOGI("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 ALOGE("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} 249 250audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 251{ 252 /* first matching HW device is returned */ 253 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 254 audio_hw_device_t *dev = mAudioHwDevs[i]; 255 if ((dev->get_supported_devices(dev) & devices) == devices) 256 return dev; 257 } 258 return NULL; 259} 260 261status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 262{ 263 const size_t SIZE = 256; 264 char buffer[SIZE]; 265 String8 result; 266 267 result.append("Clients:\n"); 268 for (size_t i = 0; i < mClients.size(); ++i) { 269 sp<Client> client = mClients.valueAt(i).promote(); 270 if (client != 0) { 271 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 272 result.append(buffer); 273 } 274 } 275 276 result.append("Global session refs:\n"); 277 result.append(" session pid cnt\n"); 278 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 279 AudioSessionRef *r = mAudioSessionRefs[i]; 280 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 281 result.append(buffer); 282 } 283 write(fd, result.string(), result.size()); 284 return NO_ERROR; 285} 286 287 288status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 289{ 290 const size_t SIZE = 256; 291 char buffer[SIZE]; 292 String8 result; 293 hardware_call_state hardwareStatus = mHardwareStatus; 294 295 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 296 result.append(buffer); 297 write(fd, result.string(), result.size()); 298 return NO_ERROR; 299} 300 301status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 302{ 303 const size_t SIZE = 256; 304 char buffer[SIZE]; 305 String8 result; 306 snprintf(buffer, SIZE, "Permission Denial: " 307 "can't dump AudioFlinger from pid=%d, uid=%d\n", 308 IPCThreadState::self()->getCallingPid(), 309 IPCThreadState::self()->getCallingUid()); 310 result.append(buffer); 311 write(fd, result.string(), result.size()); 312 return NO_ERROR; 313} 314 315static bool tryLock(Mutex& mutex) 316{ 317 bool locked = false; 318 for (int i = 0; i < kDumpLockRetries; ++i) { 319 if (mutex.tryLock() == NO_ERROR) { 320 locked = true; 321 break; 322 } 323 usleep(kDumpLockSleepUs); 324 } 325 return locked; 326} 327 328status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 329{ 330 if (!checkCallingPermission(String16("android.permission.DUMP"))) { 331 dumpPermissionDenial(fd, args); 332 } else { 333 // get state of hardware lock 334 bool hardwareLocked = tryLock(mHardwareLock); 335 if (!hardwareLocked) { 336 String8 result(kHardwareLockedString); 337 write(fd, result.string(), result.size()); 338 } else { 339 mHardwareLock.unlock(); 340 } 341 342 bool locked = tryLock(mLock); 343 344 // failed to lock - AudioFlinger is probably deadlocked 345 if (!locked) { 346 String8 result(kDeadlockedString); 347 write(fd, result.string(), result.size()); 348 } 349 350 dumpClients(fd, args); 351 dumpInternals(fd, args); 352 353 // dump playback threads 354 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 355 mPlaybackThreads.valueAt(i)->dump(fd, args); 356 } 357 358 // dump record threads 359 for (size_t i = 0; i < mRecordThreads.size(); i++) { 360 mRecordThreads.valueAt(i)->dump(fd, args); 361 } 362 363 // dump all hardware devs 364 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 365 audio_hw_device_t *dev = mAudioHwDevs[i]; 366 dev->dump(dev, fd); 367 } 368 if (locked) mLock.unlock(); 369 } 370 return NO_ERROR; 371} 372 373sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 374{ 375 // If pid is already in the mClients wp<> map, then use that entry 376 // (for which promote() is always != 0), otherwise create a new entry and Client. 377 sp<Client> client = mClients.valueFor(pid).promote(); 378 if (client == 0) { 379 client = new Client(this, pid); 380 mClients.add(pid, client); 381 } 382 383 return client; 384} 385 386// IAudioFlinger interface 387 388 389sp<IAudioTrack> AudioFlinger::createTrack( 390 pid_t pid, 391 audio_stream_type_t streamType, 392 uint32_t sampleRate, 393 audio_format_t format, 394 uint32_t channelMask, 395 int frameCount, 396 uint32_t flags, 397 const sp<IMemory>& sharedBuffer, 398 audio_io_handle_t output, 399 int *sessionId, 400 status_t *status) 401{ 402 sp<PlaybackThread::Track> track; 403 sp<TrackHandle> trackHandle; 404 sp<Client> client; 405 status_t lStatus; 406 int lSessionId; 407 408 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 409 // but if someone uses binder directly they could bypass that and cause us to crash 410 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 411 ALOGE("createTrack() invalid stream type %d", streamType); 412 lStatus = BAD_VALUE; 413 goto Exit; 414 } 415 416 { 417 Mutex::Autolock _l(mLock); 418 PlaybackThread *thread = checkPlaybackThread_l(output); 419 PlaybackThread *effectThread = NULL; 420 if (thread == NULL) { 421 ALOGE("unknown output thread"); 422 lStatus = BAD_VALUE; 423 goto Exit; 424 } 425 426 client = registerPid_l(pid); 427 428 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 429 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 430 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 431 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 432 if (mPlaybackThreads.keyAt(i) != output) { 433 // prevent same audio session on different output threads 434 uint32_t sessions = t->hasAudioSession(*sessionId); 435 if (sessions & PlaybackThread::TRACK_SESSION) { 436 ALOGE("createTrack() session ID %d already in use", *sessionId); 437 lStatus = BAD_VALUE; 438 goto Exit; 439 } 440 // check if an effect with same session ID is waiting for a track to be created 441 if (sessions & PlaybackThread::EFFECT_SESSION) { 442 effectThread = t.get(); 443 } 444 } 445 } 446 lSessionId = *sessionId; 447 } else { 448 // if no audio session id is provided, create one here 449 lSessionId = nextUniqueId(); 450 if (sessionId != NULL) { 451 *sessionId = lSessionId; 452 } 453 } 454 ALOGV("createTrack() lSessionId: %d", lSessionId); 455 456 track = thread->createTrack_l(client, streamType, sampleRate, format, 457 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 458 459 // move effect chain to this output thread if an effect on same session was waiting 460 // for a track to be created 461 if (lStatus == NO_ERROR && effectThread != NULL) { 462 Mutex::Autolock _dl(thread->mLock); 463 Mutex::Autolock _sl(effectThread->mLock); 464 moveEffectChain_l(lSessionId, effectThread, thread, true); 465 } 466 } 467 if (lStatus == NO_ERROR) { 468 trackHandle = new TrackHandle(track); 469 } else { 470 // remove local strong reference to Client before deleting the Track so that the Client 471 // destructor is called by the TrackBase destructor with mLock held 472 client.clear(); 473 track.clear(); 474 } 475 476Exit: 477 if(status) { 478 *status = lStatus; 479 } 480 return trackHandle; 481} 482 483uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 484{ 485 Mutex::Autolock _l(mLock); 486 PlaybackThread *thread = checkPlaybackThread_l(output); 487 if (thread == NULL) { 488 ALOGW("sampleRate() unknown thread %d", output); 489 return 0; 490 } 491 return thread->sampleRate(); 492} 493 494int AudioFlinger::channelCount(audio_io_handle_t output) const 495{ 496 Mutex::Autolock _l(mLock); 497 PlaybackThread *thread = checkPlaybackThread_l(output); 498 if (thread == NULL) { 499 ALOGW("channelCount() unknown thread %d", output); 500 return 0; 501 } 502 return thread->channelCount(); 503} 504 505audio_format_t AudioFlinger::format(audio_io_handle_t output) const 506{ 507 Mutex::Autolock _l(mLock); 508 PlaybackThread *thread = checkPlaybackThread_l(output); 509 if (thread == NULL) { 510 ALOGW("format() unknown thread %d", output); 511 return AUDIO_FORMAT_INVALID; 512 } 513 return thread->format(); 514} 515 516size_t AudioFlinger::frameCount(audio_io_handle_t output) const 517{ 518 Mutex::Autolock _l(mLock); 519 PlaybackThread *thread = checkPlaybackThread_l(output); 520 if (thread == NULL) { 521 ALOGW("frameCount() unknown thread %d", output); 522 return 0; 523 } 524 return thread->frameCount(); 525} 526 527uint32_t AudioFlinger::latency(audio_io_handle_t output) const 528{ 529 Mutex::Autolock _l(mLock); 530 PlaybackThread *thread = checkPlaybackThread_l(output); 531 if (thread == NULL) { 532 ALOGW("latency() unknown thread %d", output); 533 return 0; 534 } 535 return thread->latency(); 536} 537 538status_t AudioFlinger::setMasterVolume(float value) 539{ 540 status_t ret = initCheck(); 541 if (ret != NO_ERROR) { 542 return ret; 543 } 544 545 // check calling permissions 546 if (!settingsAllowed()) { 547 return PERMISSION_DENIED; 548 } 549 550 // when hw supports master volume, don't scale in sw mixer 551 { // scope for the lock 552 AutoMutex lock(mHardwareLock); 553 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 554 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 555 value = 1.0f; 556 } 557 mHardwareStatus = AUDIO_HW_IDLE; 558 } 559 560 Mutex::Autolock _l(mLock); 561 mMasterVolume = value; 562 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 563 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 564 565 return NO_ERROR; 566} 567 568status_t AudioFlinger::setMode(audio_mode_t mode) 569{ 570 status_t ret = initCheck(); 571 if (ret != NO_ERROR) { 572 return ret; 573 } 574 575 // check calling permissions 576 if (!settingsAllowed()) { 577 return PERMISSION_DENIED; 578 } 579 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 580 ALOGW("Illegal value: setMode(%d)", mode); 581 return BAD_VALUE; 582 } 583 584 { // scope for the lock 585 AutoMutex lock(mHardwareLock); 586 mHardwareStatus = AUDIO_HW_SET_MODE; 587 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 588 mHardwareStatus = AUDIO_HW_IDLE; 589 } 590 591 if (NO_ERROR == ret) { 592 Mutex::Autolock _l(mLock); 593 mMode = mode; 594 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 595 mPlaybackThreads.valueAt(i)->setMode(mode); 596 } 597 598 return ret; 599} 600 601status_t AudioFlinger::setMicMute(bool state) 602{ 603 status_t ret = initCheck(); 604 if (ret != NO_ERROR) { 605 return ret; 606 } 607 608 // check calling permissions 609 if (!settingsAllowed()) { 610 return PERMISSION_DENIED; 611 } 612 613 AutoMutex lock(mHardwareLock); 614 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 615 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 616 mHardwareStatus = AUDIO_HW_IDLE; 617 return ret; 618} 619 620bool AudioFlinger::getMicMute() const 621{ 622 status_t ret = initCheck(); 623 if (ret != NO_ERROR) { 624 return false; 625 } 626 627 bool state = AUDIO_MODE_INVALID; 628 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 629 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 630 mHardwareStatus = AUDIO_HW_IDLE; 631 return state; 632} 633 634status_t AudioFlinger::setMasterMute(bool muted) 635{ 636 // check calling permissions 637 if (!settingsAllowed()) { 638 return PERMISSION_DENIED; 639 } 640 641 Mutex::Autolock _l(mLock); 642 mMasterMute = muted; 643 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 644 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 645 646 return NO_ERROR; 647} 648 649float AudioFlinger::masterVolume() const 650{ 651 Mutex::Autolock _l(mLock); 652 return masterVolume_l(); 653} 654 655bool AudioFlinger::masterMute() const 656{ 657 Mutex::Autolock _l(mLock); 658 return masterMute_l(); 659} 660 661status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 662 audio_io_handle_t output) 663{ 664 // check calling permissions 665 if (!settingsAllowed()) { 666 return PERMISSION_DENIED; 667 } 668 669 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 670 ALOGE("setStreamVolume() invalid stream %d", stream); 671 return BAD_VALUE; 672 } 673 674 AutoMutex lock(mLock); 675 PlaybackThread *thread = NULL; 676 if (output) { 677 thread = checkPlaybackThread_l(output); 678 if (thread == NULL) { 679 return BAD_VALUE; 680 } 681 } 682 683 mStreamTypes[stream].volume = value; 684 685 if (thread == NULL) { 686 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 687 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 688 } 689 } else { 690 thread->setStreamVolume(stream, value); 691 } 692 693 return NO_ERROR; 694} 695 696status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 697{ 698 // check calling permissions 699 if (!settingsAllowed()) { 700 return PERMISSION_DENIED; 701 } 702 703 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 704 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 705 ALOGE("setStreamMute() invalid stream %d", stream); 706 return BAD_VALUE; 707 } 708 709 AutoMutex lock(mLock); 710 mStreamTypes[stream].mute = muted; 711 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 712 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 713 714 return NO_ERROR; 715} 716 717float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 718{ 719 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 720 return 0.0f; 721 } 722 723 AutoMutex lock(mLock); 724 float volume; 725 if (output) { 726 PlaybackThread *thread = checkPlaybackThread_l(output); 727 if (thread == NULL) { 728 return 0.0f; 729 } 730 volume = thread->streamVolume(stream); 731 } else { 732 volume = mStreamTypes[stream].volume; 733 } 734 735 return volume; 736} 737 738bool AudioFlinger::streamMute(audio_stream_type_t stream) const 739{ 740 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 741 return true; 742 } 743 744 return mStreamTypes[stream].mute; 745} 746 747status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 748{ 749 status_t result; 750 751 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 752 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 753 // check calling permissions 754 if (!settingsAllowed()) { 755 return PERMISSION_DENIED; 756 } 757 758 // ioHandle == 0 means the parameters are global to the audio hardware interface 759 if (ioHandle == 0) { 760 AutoMutex lock(mHardwareLock); 761 mHardwareStatus = AUDIO_SET_PARAMETER; 762 status_t final_result = NO_ERROR; 763 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 764 audio_hw_device_t *dev = mAudioHwDevs[i]; 765 result = dev->set_parameters(dev, keyValuePairs.string()); 766 final_result = result ?: final_result; 767 } 768 mHardwareStatus = AUDIO_HW_IDLE; 769 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 770 AudioParameter param = AudioParameter(keyValuePairs); 771 String8 value; 772 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 773 Mutex::Autolock _l(mLock); 774 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 775 if (mBtNrecIsOff != btNrecIsOff) { 776 for (size_t i = 0; i < mRecordThreads.size(); i++) { 777 sp<RecordThread> thread = mRecordThreads.valueAt(i); 778 RecordThread::RecordTrack *track = thread->track(); 779 if (track != NULL) { 780 audio_devices_t device = (audio_devices_t)( 781 thread->device() & AUDIO_DEVICE_IN_ALL); 782 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 783 thread->setEffectSuspended(FX_IID_AEC, 784 suspend, 785 track->sessionId()); 786 thread->setEffectSuspended(FX_IID_NS, 787 suspend, 788 track->sessionId()); 789 } 790 } 791 mBtNrecIsOff = btNrecIsOff; 792 } 793 } 794 return final_result; 795 } 796 797 // hold a strong ref on thread in case closeOutput() or closeInput() is called 798 // and the thread is exited once the lock is released 799 sp<ThreadBase> thread; 800 { 801 Mutex::Autolock _l(mLock); 802 thread = checkPlaybackThread_l(ioHandle); 803 if (thread == NULL) { 804 thread = checkRecordThread_l(ioHandle); 805 } else if (thread == primaryPlaybackThread_l()) { 806 // indicate output device change to all input threads for pre processing 807 AudioParameter param = AudioParameter(keyValuePairs); 808 int value; 809 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 810 for (size_t i = 0; i < mRecordThreads.size(); i++) { 811 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 812 } 813 } 814 } 815 } 816 if (thread != 0) { 817 return thread->setParameters(keyValuePairs); 818 } 819 return BAD_VALUE; 820} 821 822String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 823{ 824// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 825// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 826 827 if (ioHandle == 0) { 828 String8 out_s8; 829 830 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 831 audio_hw_device_t *dev = mAudioHwDevs[i]; 832 char *s = dev->get_parameters(dev, keys.string()); 833 out_s8 += String8(s); 834 free(s); 835 } 836 return out_s8; 837 } 838 839 Mutex::Autolock _l(mLock); 840 841 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 842 if (playbackThread != NULL) { 843 return playbackThread->getParameters(keys); 844 } 845 RecordThread *recordThread = checkRecordThread_l(ioHandle); 846 if (recordThread != NULL) { 847 return recordThread->getParameters(keys); 848 } 849 return String8(""); 850} 851 852size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 853{ 854 status_t ret = initCheck(); 855 if (ret != NO_ERROR) { 856 return 0; 857 } 858 859 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 860} 861 862unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 863{ 864 if (ioHandle == 0) { 865 return 0; 866 } 867 868 Mutex::Autolock _l(mLock); 869 870 RecordThread *recordThread = checkRecordThread_l(ioHandle); 871 if (recordThread != NULL) { 872 return recordThread->getInputFramesLost(); 873 } 874 return 0; 875} 876 877status_t AudioFlinger::setVoiceVolume(float value) 878{ 879 status_t ret = initCheck(); 880 if (ret != NO_ERROR) { 881 return ret; 882 } 883 884 // check calling permissions 885 if (!settingsAllowed()) { 886 return PERMISSION_DENIED; 887 } 888 889 AutoMutex lock(mHardwareLock); 890 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 891 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 892 mHardwareStatus = AUDIO_HW_IDLE; 893 894 return ret; 895} 896 897status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 898 audio_io_handle_t output) const 899{ 900 status_t status; 901 902 Mutex::Autolock _l(mLock); 903 904 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 905 if (playbackThread != NULL) { 906 return playbackThread->getRenderPosition(halFrames, dspFrames); 907 } 908 909 return BAD_VALUE; 910} 911 912void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 913{ 914 915 Mutex::Autolock _l(mLock); 916 917 pid_t pid = IPCThreadState::self()->getCallingPid(); 918 if (mNotificationClients.indexOfKey(pid) < 0) { 919 sp<NotificationClient> notificationClient = new NotificationClient(this, 920 client, 921 pid); 922 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 923 924 mNotificationClients.add(pid, notificationClient); 925 926 sp<IBinder> binder = client->asBinder(); 927 binder->linkToDeath(notificationClient); 928 929 // the config change is always sent from playback or record threads to avoid deadlock 930 // with AudioSystem::gLock 931 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 932 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 933 } 934 935 for (size_t i = 0; i < mRecordThreads.size(); i++) { 936 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 937 } 938 } 939} 940 941void AudioFlinger::removeNotificationClient(pid_t pid) 942{ 943 Mutex::Autolock _l(mLock); 944 945 int index = mNotificationClients.indexOfKey(pid); 946 if (index >= 0) { 947 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 948 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 949 mNotificationClients.removeItem(pid); 950 } 951 952 ALOGV("%d died, releasing its sessions", pid); 953 int num = mAudioSessionRefs.size(); 954 bool removed = false; 955 for (int i = 0; i< num; i++) { 956 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 957 ALOGV(" pid %d @ %d", ref->pid, i); 958 if (ref->pid == pid) { 959 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 960 mAudioSessionRefs.removeAt(i); 961 delete ref; 962 removed = true; 963 i--; 964 num--; 965 } 966 } 967 if (removed) { 968 purgeStaleEffects_l(); 969 } 970} 971 972// audioConfigChanged_l() must be called with AudioFlinger::mLock held 973void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 974{ 975 size_t size = mNotificationClients.size(); 976 for (size_t i = 0; i < size; i++) { 977 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 978 param2); 979 } 980} 981 982// removeClient_l() must be called with AudioFlinger::mLock held 983void AudioFlinger::removeClient_l(pid_t pid) 984{ 985 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 986 mClients.removeItem(pid); 987} 988 989 990// ---------------------------------------------------------------------------- 991 992AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 993 uint32_t device, type_t type) 994 : Thread(false), 995 mType(type), 996 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 997 // mChannelMask 998 mChannelCount(0), 999 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1000 mParamStatus(NO_ERROR), 1001 mStandby(false), mId(id), 1002 mDevice(device), 1003 mDeathRecipient(new PMDeathRecipient(this)) 1004{ 1005} 1006 1007AudioFlinger::ThreadBase::~ThreadBase() 1008{ 1009 mParamCond.broadcast(); 1010 // do not lock the mutex in destructor 1011 releaseWakeLock_l(); 1012 if (mPowerManager != 0) { 1013 sp<IBinder> binder = mPowerManager->asBinder(); 1014 binder->unlinkToDeath(mDeathRecipient); 1015 } 1016} 1017 1018void AudioFlinger::ThreadBase::exit() 1019{ 1020 ALOGV("ThreadBase::exit"); 1021 { 1022 // This lock prevents the following race in thread (uniprocessor for illustration): 1023 // if (!exitPending()) { 1024 // // context switch from here to exit() 1025 // // exit() calls requestExit(), what exitPending() observes 1026 // // exit() calls signal(), which is dropped since no waiters 1027 // // context switch back from exit() to here 1028 // mWaitWorkCV.wait(...); 1029 // // now thread is hung 1030 // } 1031 AutoMutex lock(mLock); 1032 requestExit(); 1033 mWaitWorkCV.signal(); 1034 } 1035 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1036 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1037 requestExitAndWait(); 1038} 1039 1040status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1041{ 1042 status_t status; 1043 1044 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1045 Mutex::Autolock _l(mLock); 1046 1047 mNewParameters.add(keyValuePairs); 1048 mWaitWorkCV.signal(); 1049 // wait condition with timeout in case the thread loop has exited 1050 // before the request could be processed 1051 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1052 status = mParamStatus; 1053 mWaitWorkCV.signal(); 1054 } else { 1055 status = TIMED_OUT; 1056 } 1057 return status; 1058} 1059 1060void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1061{ 1062 Mutex::Autolock _l(mLock); 1063 sendConfigEvent_l(event, param); 1064} 1065 1066// sendConfigEvent_l() must be called with ThreadBase::mLock held 1067void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1068{ 1069 ConfigEvent configEvent; 1070 configEvent.mEvent = event; 1071 configEvent.mParam = param; 1072 mConfigEvents.add(configEvent); 1073 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1074 mWaitWorkCV.signal(); 1075} 1076 1077void AudioFlinger::ThreadBase::processConfigEvents() 1078{ 1079 mLock.lock(); 1080 while(!mConfigEvents.isEmpty()) { 1081 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1082 ConfigEvent configEvent = mConfigEvents[0]; 1083 mConfigEvents.removeAt(0); 1084 // release mLock before locking AudioFlinger mLock: lock order is always 1085 // AudioFlinger then ThreadBase to avoid cross deadlock 1086 mLock.unlock(); 1087 mAudioFlinger->mLock.lock(); 1088 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1089 mAudioFlinger->mLock.unlock(); 1090 mLock.lock(); 1091 } 1092 mLock.unlock(); 1093} 1094 1095status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1096{ 1097 const size_t SIZE = 256; 1098 char buffer[SIZE]; 1099 String8 result; 1100 1101 bool locked = tryLock(mLock); 1102 if (!locked) { 1103 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1104 write(fd, buffer, strlen(buffer)); 1105 } 1106 1107 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1108 result.append(buffer); 1109 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1110 result.append(buffer); 1111 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1112 result.append(buffer); 1113 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1114 result.append(buffer); 1115 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1116 result.append(buffer); 1117 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1120 result.append(buffer); 1121 1122 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1123 result.append(buffer); 1124 result.append(" Index Command"); 1125 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1126 snprintf(buffer, SIZE, "\n %02d ", i); 1127 result.append(buffer); 1128 result.append(mNewParameters[i]); 1129 } 1130 1131 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1132 result.append(buffer); 1133 snprintf(buffer, SIZE, " Index event param\n"); 1134 result.append(buffer); 1135 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1136 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1137 result.append(buffer); 1138 } 1139 result.append("\n"); 1140 1141 write(fd, result.string(), result.size()); 1142 1143 if (locked) { 1144 mLock.unlock(); 1145 } 1146 return NO_ERROR; 1147} 1148 1149status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1150{ 1151 const size_t SIZE = 256; 1152 char buffer[SIZE]; 1153 String8 result; 1154 1155 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1156 write(fd, buffer, strlen(buffer)); 1157 1158 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1159 sp<EffectChain> chain = mEffectChains[i]; 1160 if (chain != 0) { 1161 chain->dump(fd, args); 1162 } 1163 } 1164 return NO_ERROR; 1165} 1166 1167void AudioFlinger::ThreadBase::acquireWakeLock() 1168{ 1169 Mutex::Autolock _l(mLock); 1170 acquireWakeLock_l(); 1171} 1172 1173void AudioFlinger::ThreadBase::acquireWakeLock_l() 1174{ 1175 if (mPowerManager == 0) { 1176 // use checkService() to avoid blocking if power service is not up yet 1177 sp<IBinder> binder = 1178 defaultServiceManager()->checkService(String16("power")); 1179 if (binder == 0) { 1180 ALOGW("Thread %s cannot connect to the power manager service", mName); 1181 } else { 1182 mPowerManager = interface_cast<IPowerManager>(binder); 1183 binder->linkToDeath(mDeathRecipient); 1184 } 1185 } 1186 if (mPowerManager != 0) { 1187 sp<IBinder> binder = new BBinder(); 1188 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1189 binder, 1190 String16(mName)); 1191 if (status == NO_ERROR) { 1192 mWakeLockToken = binder; 1193 } 1194 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1195 } 1196} 1197 1198void AudioFlinger::ThreadBase::releaseWakeLock() 1199{ 1200 Mutex::Autolock _l(mLock); 1201 releaseWakeLock_l(); 1202} 1203 1204void AudioFlinger::ThreadBase::releaseWakeLock_l() 1205{ 1206 if (mWakeLockToken != 0) { 1207 ALOGV("releaseWakeLock_l() %s", mName); 1208 if (mPowerManager != 0) { 1209 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1210 } 1211 mWakeLockToken.clear(); 1212 } 1213} 1214 1215void AudioFlinger::ThreadBase::clearPowerManager() 1216{ 1217 Mutex::Autolock _l(mLock); 1218 releaseWakeLock_l(); 1219 mPowerManager.clear(); 1220} 1221 1222void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1223{ 1224 sp<ThreadBase> thread = mThread.promote(); 1225 if (thread != 0) { 1226 thread->clearPowerManager(); 1227 } 1228 ALOGW("power manager service died !!!"); 1229} 1230 1231void AudioFlinger::ThreadBase::setEffectSuspended( 1232 const effect_uuid_t *type, bool suspend, int sessionId) 1233{ 1234 Mutex::Autolock _l(mLock); 1235 setEffectSuspended_l(type, suspend, sessionId); 1236} 1237 1238void AudioFlinger::ThreadBase::setEffectSuspended_l( 1239 const effect_uuid_t *type, bool suspend, int sessionId) 1240{ 1241 sp<EffectChain> chain = getEffectChain_l(sessionId); 1242 if (chain != 0) { 1243 if (type != NULL) { 1244 chain->setEffectSuspended_l(type, suspend); 1245 } else { 1246 chain->setEffectSuspendedAll_l(suspend); 1247 } 1248 } 1249 1250 updateSuspendedSessions_l(type, suspend, sessionId); 1251} 1252 1253void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1254{ 1255 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1256 if (index < 0) { 1257 return; 1258 } 1259 1260 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1261 mSuspendedSessions.editValueAt(index); 1262 1263 for (size_t i = 0; i < sessionEffects.size(); i++) { 1264 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1265 for (int j = 0; j < desc->mRefCount; j++) { 1266 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1267 chain->setEffectSuspendedAll_l(true); 1268 } else { 1269 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1270 desc->mType.timeLow); 1271 chain->setEffectSuspended_l(&desc->mType, true); 1272 } 1273 } 1274 } 1275} 1276 1277void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1278 bool suspend, 1279 int sessionId) 1280{ 1281 int index = mSuspendedSessions.indexOfKey(sessionId); 1282 1283 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1284 1285 if (suspend) { 1286 if (index >= 0) { 1287 sessionEffects = mSuspendedSessions.editValueAt(index); 1288 } else { 1289 mSuspendedSessions.add(sessionId, sessionEffects); 1290 } 1291 } else { 1292 if (index < 0) { 1293 return; 1294 } 1295 sessionEffects = mSuspendedSessions.editValueAt(index); 1296 } 1297 1298 1299 int key = EffectChain::kKeyForSuspendAll; 1300 if (type != NULL) { 1301 key = type->timeLow; 1302 } 1303 index = sessionEffects.indexOfKey(key); 1304 1305 sp <SuspendedSessionDesc> desc; 1306 if (suspend) { 1307 if (index >= 0) { 1308 desc = sessionEffects.valueAt(index); 1309 } else { 1310 desc = new SuspendedSessionDesc(); 1311 if (type != NULL) { 1312 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1313 } 1314 sessionEffects.add(key, desc); 1315 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1316 } 1317 desc->mRefCount++; 1318 } else { 1319 if (index < 0) { 1320 return; 1321 } 1322 desc = sessionEffects.valueAt(index); 1323 if (--desc->mRefCount == 0) { 1324 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1325 sessionEffects.removeItemsAt(index); 1326 if (sessionEffects.isEmpty()) { 1327 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1328 sessionId); 1329 mSuspendedSessions.removeItem(sessionId); 1330 } 1331 } 1332 } 1333 if (!sessionEffects.isEmpty()) { 1334 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1335 } 1336} 1337 1338void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1339 bool enabled, 1340 int sessionId) 1341{ 1342 Mutex::Autolock _l(mLock); 1343 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1344} 1345 1346void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1347 bool enabled, 1348 int sessionId) 1349{ 1350 if (mType != RECORD) { 1351 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1352 // another session. This gives the priority to well behaved effect control panels 1353 // and applications not using global effects. 1354 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1355 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1356 } 1357 } 1358 1359 sp<EffectChain> chain = getEffectChain_l(sessionId); 1360 if (chain != 0) { 1361 chain->checkSuspendOnEffectEnabled(effect, enabled); 1362 } 1363} 1364 1365// ---------------------------------------------------------------------------- 1366 1367AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1368 AudioStreamOut* output, 1369 audio_io_handle_t id, 1370 uint32_t device, 1371 type_t type) 1372 : ThreadBase(audioFlinger, id, device, type), 1373 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1374 // Assumes constructor is called by AudioFlinger with it's mLock held, 1375 // but it would be safer to explicitly pass initial masterMute as parameter 1376 mMasterMute(audioFlinger->masterMute_l()), 1377 // mStreamTypes[] initialized in constructor body 1378 mOutput(output), 1379 // Assumes constructor is called by AudioFlinger with it's mLock held, 1380 // but it would be safer to explicitly pass initial masterVolume as parameter 1381 mMasterVolume(audioFlinger->masterVolume_l()), 1382 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1383{ 1384 snprintf(mName, kNameLength, "AudioOut_%d", id); 1385 1386 readOutputParameters(); 1387 1388 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1389 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1390 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1391 stream = (audio_stream_type_t) (stream + 1)) { 1392 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1393 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1394 // initialized by stream_type_t default constructor 1395 // mStreamTypes[stream].valid = true; 1396 } 1397} 1398 1399AudioFlinger::PlaybackThread::~PlaybackThread() 1400{ 1401 delete [] mMixBuffer; 1402} 1403 1404status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1405{ 1406 dumpInternals(fd, args); 1407 dumpTracks(fd, args); 1408 dumpEffectChains(fd, args); 1409 return NO_ERROR; 1410} 1411 1412status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1413{ 1414 const size_t SIZE = 256; 1415 char buffer[SIZE]; 1416 String8 result; 1417 1418 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1419 result.append(buffer); 1420 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1421 for (size_t i = 0; i < mTracks.size(); ++i) { 1422 sp<Track> track = mTracks[i]; 1423 if (track != 0) { 1424 track->dump(buffer, SIZE); 1425 result.append(buffer); 1426 } 1427 } 1428 1429 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1430 result.append(buffer); 1431 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1432 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1433 sp<Track> track = mActiveTracks[i].promote(); 1434 if (track != 0) { 1435 track->dump(buffer, SIZE); 1436 result.append(buffer); 1437 } 1438 } 1439 write(fd, result.string(), result.size()); 1440 return NO_ERROR; 1441} 1442 1443status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1444{ 1445 const size_t SIZE = 256; 1446 char buffer[SIZE]; 1447 String8 result; 1448 1449 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1450 result.append(buffer); 1451 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1452 result.append(buffer); 1453 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1454 result.append(buffer); 1455 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1456 result.append(buffer); 1457 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1458 result.append(buffer); 1459 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1460 result.append(buffer); 1461 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1462 result.append(buffer); 1463 write(fd, result.string(), result.size()); 1464 1465 dumpBase(fd, args); 1466 1467 return NO_ERROR; 1468} 1469 1470// Thread virtuals 1471status_t AudioFlinger::PlaybackThread::readyToRun() 1472{ 1473 status_t status = initCheck(); 1474 if (status == NO_ERROR) { 1475 ALOGI("AudioFlinger's thread %p ready to run", this); 1476 } else { 1477 ALOGE("No working audio driver found."); 1478 } 1479 return status; 1480} 1481 1482void AudioFlinger::PlaybackThread::onFirstRef() 1483{ 1484 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1485} 1486 1487// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1488sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1489 const sp<AudioFlinger::Client>& client, 1490 audio_stream_type_t streamType, 1491 uint32_t sampleRate, 1492 audio_format_t format, 1493 uint32_t channelMask, 1494 int frameCount, 1495 const sp<IMemory>& sharedBuffer, 1496 int sessionId, 1497 status_t *status) 1498{ 1499 sp<Track> track; 1500 status_t lStatus; 1501 1502 if (mType == DIRECT) { 1503 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1504 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1505 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1506 "for output %p with format %d", 1507 sampleRate, format, channelMask, mOutput, mFormat); 1508 lStatus = BAD_VALUE; 1509 goto Exit; 1510 } 1511 } 1512 } else { 1513 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1514 if (sampleRate > mSampleRate*2) { 1515 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1516 lStatus = BAD_VALUE; 1517 goto Exit; 1518 } 1519 } 1520 1521 lStatus = initCheck(); 1522 if (lStatus != NO_ERROR) { 1523 ALOGE("Audio driver not initialized."); 1524 goto Exit; 1525 } 1526 1527 { // scope for mLock 1528 Mutex::Autolock _l(mLock); 1529 1530 // all tracks in same audio session must share the same routing strategy otherwise 1531 // conflicts will happen when tracks are moved from one output to another by audio policy 1532 // manager 1533 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1534 for (size_t i = 0; i < mTracks.size(); ++i) { 1535 sp<Track> t = mTracks[i]; 1536 if (t != 0) { 1537 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1538 if (sessionId == t->sessionId() && strategy != actual) { 1539 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1540 strategy, actual); 1541 lStatus = BAD_VALUE; 1542 goto Exit; 1543 } 1544 } 1545 } 1546 1547 track = new Track(this, client, streamType, sampleRate, format, 1548 channelMask, frameCount, sharedBuffer, sessionId); 1549 if (track->getCblk() == NULL || track->name() < 0) { 1550 lStatus = NO_MEMORY; 1551 goto Exit; 1552 } 1553 mTracks.add(track); 1554 1555 sp<EffectChain> chain = getEffectChain_l(sessionId); 1556 if (chain != 0) { 1557 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1558 track->setMainBuffer(chain->inBuffer()); 1559 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1560 chain->incTrackCnt(); 1561 } 1562 1563 // invalidate track immediately if the stream type was moved to another thread since 1564 // createTrack() was called by the client process. 1565 if (!mStreamTypes[streamType].valid) { 1566 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1567 this, streamType); 1568 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1569 } 1570 } 1571 lStatus = NO_ERROR; 1572 1573Exit: 1574 if(status) { 1575 *status = lStatus; 1576 } 1577 return track; 1578} 1579 1580uint32_t AudioFlinger::PlaybackThread::latency() const 1581{ 1582 Mutex::Autolock _l(mLock); 1583 if (initCheck() == NO_ERROR) { 1584 return mOutput->stream->get_latency(mOutput->stream); 1585 } else { 1586 return 0; 1587 } 1588} 1589 1590status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1591{ 1592 mMasterVolume = value; 1593 return NO_ERROR; 1594} 1595 1596status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1597{ 1598 mMasterMute = muted; 1599 return NO_ERROR; 1600} 1601 1602status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1603{ 1604 mStreamTypes[stream].volume = value; 1605 return NO_ERROR; 1606} 1607 1608status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1609{ 1610 mStreamTypes[stream].mute = muted; 1611 return NO_ERROR; 1612} 1613 1614float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1615{ 1616 return mStreamTypes[stream].volume; 1617} 1618 1619bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1620{ 1621 return mStreamTypes[stream].mute; 1622} 1623 1624// addTrack_l() must be called with ThreadBase::mLock held 1625status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1626{ 1627 status_t status = ALREADY_EXISTS; 1628 1629 // set retry count for buffer fill 1630 track->mRetryCount = kMaxTrackStartupRetries; 1631 if (mActiveTracks.indexOf(track) < 0) { 1632 // the track is newly added, make sure it fills up all its 1633 // buffers before playing. This is to ensure the client will 1634 // effectively get the latency it requested. 1635 track->mFillingUpStatus = Track::FS_FILLING; 1636 track->mResetDone = false; 1637 mActiveTracks.add(track); 1638 if (track->mainBuffer() != mMixBuffer) { 1639 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1640 if (chain != 0) { 1641 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1642 chain->incActiveTrackCnt(); 1643 } 1644 } 1645 1646 status = NO_ERROR; 1647 } 1648 1649 ALOGV("mWaitWorkCV.broadcast"); 1650 mWaitWorkCV.broadcast(); 1651 1652 return status; 1653} 1654 1655// destroyTrack_l() must be called with ThreadBase::mLock held 1656void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1657{ 1658 track->mState = TrackBase::TERMINATED; 1659 if (mActiveTracks.indexOf(track) < 0) { 1660 removeTrack_l(track); 1661 } 1662} 1663 1664void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1665{ 1666 mTracks.remove(track); 1667 deleteTrackName_l(track->name()); 1668 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1669 if (chain != 0) { 1670 chain->decTrackCnt(); 1671 } 1672} 1673 1674String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1675{ 1676 String8 out_s8 = String8(""); 1677 char *s; 1678 1679 Mutex::Autolock _l(mLock); 1680 if (initCheck() != NO_ERROR) { 1681 return out_s8; 1682 } 1683 1684 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1685 out_s8 = String8(s); 1686 free(s); 1687 return out_s8; 1688} 1689 1690// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1691void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1692 AudioSystem::OutputDescriptor desc; 1693 void *param2 = NULL; 1694 1695 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1696 1697 switch (event) { 1698 case AudioSystem::OUTPUT_OPENED: 1699 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1700 desc.channels = mChannelMask; 1701 desc.samplingRate = mSampleRate; 1702 desc.format = mFormat; 1703 desc.frameCount = mFrameCount; 1704 desc.latency = latency(); 1705 param2 = &desc; 1706 break; 1707 1708 case AudioSystem::STREAM_CONFIG_CHANGED: 1709 param2 = ¶m; 1710 case AudioSystem::OUTPUT_CLOSED: 1711 default: 1712 break; 1713 } 1714 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1715} 1716 1717void AudioFlinger::PlaybackThread::readOutputParameters() 1718{ 1719 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1720 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1721 mChannelCount = (uint16_t)popcount(mChannelMask); 1722 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1723 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1724 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1725 1726 // FIXME - Current mixer implementation only supports stereo output: Always 1727 // Allocate a stereo buffer even if HW output is mono. 1728 delete[] mMixBuffer; 1729 mMixBuffer = new int16_t[mFrameCount * 2]; 1730 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1731 1732 // force reconfiguration of effect chains and engines to take new buffer size and audio 1733 // parameters into account 1734 // Note that mLock is not held when readOutputParameters() is called from the constructor 1735 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1736 // matter. 1737 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1738 Vector< sp<EffectChain> > effectChains = mEffectChains; 1739 for (size_t i = 0; i < effectChains.size(); i ++) { 1740 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1741 } 1742} 1743 1744status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1745{ 1746 if (halFrames == NULL || dspFrames == NULL) { 1747 return BAD_VALUE; 1748 } 1749 Mutex::Autolock _l(mLock); 1750 if (initCheck() != NO_ERROR) { 1751 return INVALID_OPERATION; 1752 } 1753 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1754 1755 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1756} 1757 1758uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1759{ 1760 Mutex::Autolock _l(mLock); 1761 uint32_t result = 0; 1762 if (getEffectChain_l(sessionId) != 0) { 1763 result = EFFECT_SESSION; 1764 } 1765 1766 for (size_t i = 0; i < mTracks.size(); ++i) { 1767 sp<Track> track = mTracks[i]; 1768 if (sessionId == track->sessionId() && 1769 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1770 result |= TRACK_SESSION; 1771 break; 1772 } 1773 } 1774 1775 return result; 1776} 1777 1778uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1779{ 1780 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1781 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1782 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1783 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1784 } 1785 for (size_t i = 0; i < mTracks.size(); i++) { 1786 sp<Track> track = mTracks[i]; 1787 if (sessionId == track->sessionId() && 1788 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1789 return AudioSystem::getStrategyForStream(track->streamType()); 1790 } 1791 } 1792 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1793} 1794 1795 1796AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1797{ 1798 Mutex::Autolock _l(mLock); 1799 return mOutput; 1800} 1801 1802AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1803{ 1804 Mutex::Autolock _l(mLock); 1805 AudioStreamOut *output = mOutput; 1806 mOutput = NULL; 1807 return output; 1808} 1809 1810// this method must always be called either with ThreadBase mLock held or inside the thread loop 1811audio_stream_t* AudioFlinger::PlaybackThread::stream() 1812{ 1813 if (mOutput == NULL) { 1814 return NULL; 1815 } 1816 return &mOutput->stream->common; 1817} 1818 1819uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1820{ 1821 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1822 // decoding and transfer time. So sleeping for half of the latency would likely cause 1823 // underruns 1824 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1825 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1826 } else { 1827 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1828 } 1829} 1830 1831// ---------------------------------------------------------------------------- 1832 1833AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1834 audio_io_handle_t id, uint32_t device, type_t type) 1835 : PlaybackThread(audioFlinger, output, id, device, type), 1836 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1837 mPrevMixerStatus(MIXER_IDLE) 1838{ 1839 // FIXME - Current mixer implementation only supports stereo output 1840 if (mChannelCount == 1) { 1841 ALOGE("Invalid audio hardware channel count"); 1842 } 1843} 1844 1845AudioFlinger::MixerThread::~MixerThread() 1846{ 1847 delete mAudioMixer; 1848} 1849 1850bool AudioFlinger::MixerThread::threadLoop() 1851{ 1852 Vector< sp<Track> > tracksToRemove; 1853 mixer_state mixerStatus = MIXER_IDLE; 1854 nsecs_t standbyTime = systemTime(); 1855 size_t mixBufferSize = mFrameCount * mFrameSize; 1856 // FIXME: Relaxed timing because of a certain device that can't meet latency 1857 // Should be reduced to 2x after the vendor fixes the driver issue 1858 // increase threshold again due to low power audio mode. The way this warning threshold is 1859 // calculated and its usefulness should be reconsidered anyway. 1860 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1861 nsecs_t lastWarning = 0; 1862 bool longStandbyExit = false; 1863 uint32_t activeSleepTime = activeSleepTimeUs(); 1864 uint32_t idleSleepTime = idleSleepTimeUs(); 1865 uint32_t sleepTime = idleSleepTime; 1866 uint32_t sleepTimeShift = 0; 1867 Vector< sp<EffectChain> > effectChains; 1868#ifdef DEBUG_CPU_USAGE 1869 ThreadCpuUsage cpu; 1870 const CentralTendencyStatistics& stats = cpu.statistics(); 1871#endif 1872 1873 acquireWakeLock(); 1874 1875 while (!exitPending()) 1876 { 1877#ifdef DEBUG_CPU_USAGE 1878 cpu.sampleAndEnable(); 1879 unsigned n = stats.n(); 1880 // cpu.elapsed() is expensive, so don't call it every loop 1881 if ((n & 127) == 1) { 1882 long long elapsed = cpu.elapsed(); 1883 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1884 double perLoop = elapsed / (double) n; 1885 double perLoop100 = perLoop * 0.01; 1886 double mean = stats.mean(); 1887 double stddev = stats.stddev(); 1888 double minimum = stats.minimum(); 1889 double maximum = stats.maximum(); 1890 cpu.resetStatistics(); 1891 ALOGI("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", 1892 elapsed * .000000001, n, perLoop * .000001, 1893 mean * .001, 1894 stddev * .001, 1895 minimum * .001, 1896 maximum * .001, 1897 mean / perLoop100, 1898 stddev / perLoop100, 1899 minimum / perLoop100, 1900 maximum / perLoop100); 1901 } 1902 } 1903#endif 1904 processConfigEvents(); 1905 1906 mixerStatus = MIXER_IDLE; 1907 { // scope for mLock 1908 1909 Mutex::Autolock _l(mLock); 1910 1911 if (checkForNewParameters_l()) { 1912 mixBufferSize = mFrameCount * mFrameSize; 1913 // FIXME: Relaxed timing because of a certain device that can't meet latency 1914 // Should be reduced to 2x after the vendor fixes the driver issue 1915 // increase threshold again due to low power audio mode. The way this warning 1916 // threshold is calculated and its usefulness should be reconsidered anyway. 1917 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1918 activeSleepTime = activeSleepTimeUs(); 1919 idleSleepTime = idleSleepTimeUs(); 1920 } 1921 1922 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1923 1924 // put audio hardware into standby after short delay 1925 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1926 mSuspended)) { 1927 if (!mStandby) { 1928 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 1929 mOutput->stream->common.standby(&mOutput->stream->common); 1930 mStandby = true; 1931 mBytesWritten = 0; 1932 } 1933 1934 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1935 // we're about to wait, flush the binder command buffer 1936 IPCThreadState::self()->flushCommands(); 1937 1938 if (exitPending()) break; 1939 1940 releaseWakeLock_l(); 1941 // wait until we have something to do... 1942 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 1943 mWaitWorkCV.wait(mLock); 1944 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 1945 acquireWakeLock_l(); 1946 1947 mPrevMixerStatus = MIXER_IDLE; 1948 if (!mMasterMute) { 1949 char value[PROPERTY_VALUE_MAX]; 1950 property_get("ro.audio.silent", value, "0"); 1951 if (atoi(value)) { 1952 ALOGD("Silence is golden"); 1953 setMasterMute(true); 1954 } 1955 } 1956 1957 standbyTime = systemTime() + kStandbyTimeInNsecs; 1958 sleepTime = idleSleepTime; 1959 sleepTimeShift = 0; 1960 continue; 1961 } 1962 } 1963 1964 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1965 1966 // prevent any changes in effect chain list and in each effect chain 1967 // during mixing and effect process as the audio buffers could be deleted 1968 // or modified if an effect is created or deleted 1969 lockEffectChains_l(effectChains); 1970 } 1971 1972 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1973 // mix buffers... 1974 mAudioMixer->process(); 1975 // increase sleep time progressively when application underrun condition clears. 1976 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1977 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1978 // such that we would underrun the audio HAL. 1979 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 1980 sleepTimeShift--; 1981 } 1982 sleepTime = 0; 1983 standbyTime = systemTime() + kStandbyTimeInNsecs; 1984 //TODO: delay standby when effects have a tail 1985 } else { 1986 // If no tracks are ready, sleep once for the duration of an output 1987 // buffer size, then write 0s to the output 1988 if (sleepTime == 0) { 1989 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1990 sleepTime = activeSleepTime >> sleepTimeShift; 1991 if (sleepTime < kMinThreadSleepTimeUs) { 1992 sleepTime = kMinThreadSleepTimeUs; 1993 } 1994 // reduce sleep time in case of consecutive application underruns to avoid 1995 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 1996 // duration we would end up writing less data than needed by the audio HAL if 1997 // the condition persists. 1998 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 1999 sleepTimeShift++; 2000 } 2001 } else { 2002 sleepTime = idleSleepTime; 2003 } 2004 } else if (mBytesWritten != 0 || 2005 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2006 memset (mMixBuffer, 0, mixBufferSize); 2007 sleepTime = 0; 2008 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2009 } 2010 // TODO add standby time extension fct of effect tail 2011 } 2012 2013 if (mSuspended) { 2014 sleepTime = suspendSleepTimeUs(); 2015 } 2016 // sleepTime == 0 means we must write to audio hardware 2017 if (sleepTime == 0) { 2018 for (size_t i = 0; i < effectChains.size(); i ++) { 2019 effectChains[i]->process_l(); 2020 } 2021 // enable changes in effect chain 2022 unlockEffectChains(effectChains); 2023 mLastWriteTime = systemTime(); 2024 mInWrite = true; 2025 mBytesWritten += mixBufferSize; 2026 2027 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2028 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2029 mNumWrites++; 2030 mInWrite = false; 2031 nsecs_t now = systemTime(); 2032 nsecs_t delta = now - mLastWriteTime; 2033 if (!mStandby && delta > maxPeriod) { 2034 mNumDelayedWrites++; 2035 if ((now - lastWarning) > kWarningThrottleNs) { 2036 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2037 ns2ms(delta), mNumDelayedWrites, this); 2038 lastWarning = now; 2039 } 2040 if (mStandby) { 2041 longStandbyExit = true; 2042 } 2043 } 2044 mStandby = false; 2045 } else { 2046 // enable changes in effect chain 2047 unlockEffectChains(effectChains); 2048 usleep(sleepTime); 2049 } 2050 2051 // finally let go of all our tracks, without the lock held 2052 // since we can't guarantee the destructors won't acquire that 2053 // same lock. 2054 tracksToRemove.clear(); 2055 2056 // Effect chains will be actually deleted here if they were removed from 2057 // mEffectChains list during mixing or effects processing 2058 effectChains.clear(); 2059 } 2060 2061 if (!mStandby) { 2062 mOutput->stream->common.standby(&mOutput->stream->common); 2063 } 2064 2065 releaseWakeLock(); 2066 2067 ALOGV("MixerThread %p exiting", this); 2068 return false; 2069} 2070 2071// prepareTracks_l() must be called with ThreadBase::mLock held 2072AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2073 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2074{ 2075 2076 mixer_state mixerStatus = MIXER_IDLE; 2077 // find out which tracks need to be processed 2078 size_t count = activeTracks.size(); 2079 size_t mixedTracks = 0; 2080 size_t tracksWithEffect = 0; 2081 2082 float masterVolume = mMasterVolume; 2083 bool masterMute = mMasterMute; 2084 2085 if (masterMute) { 2086 masterVolume = 0; 2087 } 2088 // Delegate master volume control to effect in output mix effect chain if needed 2089 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2090 if (chain != 0) { 2091 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2092 chain->setVolume_l(&v, &v); 2093 masterVolume = (float)((v + (1 << 23)) >> 24); 2094 chain.clear(); 2095 } 2096 2097 for (size_t i=0 ; i<count ; i++) { 2098 sp<Track> t = activeTracks[i].promote(); 2099 if (t == 0) continue; 2100 2101 // this const just means the local variable doesn't change 2102 Track* const track = t.get(); 2103 audio_track_cblk_t* cblk = track->cblk(); 2104 2105 // The first time a track is added we wait 2106 // for all its buffers to be filled before processing it 2107 int name = track->name(); 2108 // make sure that we have enough frames to mix one full buffer. 2109 // enforce this condition only once to enable draining the buffer in case the client 2110 // app does not call stop() and relies on underrun to stop: 2111 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2112 // during last round 2113 uint32_t minFrames = 1; 2114 if (!track->isStopped() && !track->isPausing() && 2115 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2116 if (t->sampleRate() == (int)mSampleRate) { 2117 minFrames = mFrameCount; 2118 } else { 2119 // +1 for rounding and +1 for additional sample needed for interpolation 2120 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2121 // add frames already consumed but not yet released by the resampler 2122 // because cblk->framesReady() will include these frames 2123 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2124 // the minimum track buffer size is normally twice the number of frames necessary 2125 // to fill one buffer and the resampler should not leave more than one buffer worth 2126 // of unreleased frames after each pass, but just in case... 2127 ALOG_ASSERT(minFrames <= cblk->frameCount); 2128 } 2129 } 2130 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2131 !track->isPaused() && !track->isTerminated()) 2132 { 2133 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2134 2135 mixedTracks++; 2136 2137 // track->mainBuffer() != mMixBuffer means there is an effect chain 2138 // connected to the track 2139 chain.clear(); 2140 if (track->mainBuffer() != mMixBuffer) { 2141 chain = getEffectChain_l(track->sessionId()); 2142 // Delegate volume control to effect in track effect chain if needed 2143 if (chain != 0) { 2144 tracksWithEffect++; 2145 } else { 2146 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2147 name, track->sessionId()); 2148 } 2149 } 2150 2151 2152 int param = AudioMixer::VOLUME; 2153 if (track->mFillingUpStatus == Track::FS_FILLED) { 2154 // no ramp for the first volume setting 2155 track->mFillingUpStatus = Track::FS_ACTIVE; 2156 if (track->mState == TrackBase::RESUMING) { 2157 track->mState = TrackBase::ACTIVE; 2158 param = AudioMixer::RAMP_VOLUME; 2159 } 2160 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2161 } else if (cblk->server != 0) { 2162 // If the track is stopped before the first frame was mixed, 2163 // do not apply ramp 2164 param = AudioMixer::RAMP_VOLUME; 2165 } 2166 2167 // compute volume for this track 2168 uint32_t vl, vr, va; 2169 if (track->isMuted() || track->isPausing() || 2170 mStreamTypes[track->streamType()].mute) { 2171 vl = vr = va = 0; 2172 if (track->isPausing()) { 2173 track->setPaused(); 2174 } 2175 } else { 2176 2177 // read original volumes with volume control 2178 float typeVolume = mStreamTypes[track->streamType()].volume; 2179 float v = masterVolume * typeVolume; 2180 uint32_t vlr = cblk->getVolumeLR(); 2181 vl = vlr & 0xFFFF; 2182 vr = vlr >> 16; 2183 // track volumes come from shared memory, so can't be trusted and must be clamped 2184 if (vl > MAX_GAIN_INT) { 2185 ALOGV("Track left volume out of range: %04X", vl); 2186 vl = MAX_GAIN_INT; 2187 } 2188 if (vr > MAX_GAIN_INT) { 2189 ALOGV("Track right volume out of range: %04X", vr); 2190 vr = MAX_GAIN_INT; 2191 } 2192 // now apply the master volume and stream type volume 2193 vl = (uint32_t)(v * vl) << 12; 2194 vr = (uint32_t)(v * vr) << 12; 2195 // assuming master volume and stream type volume each go up to 1.0, 2196 // vl and vr are now in 8.24 format 2197 2198 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2199 // send level comes from shared memory and so may be corrupt 2200 if (sendLevel >= MAX_GAIN_INT) { 2201 ALOGV("Track send level out of range: %04X", sendLevel); 2202 sendLevel = MAX_GAIN_INT; 2203 } 2204 va = (uint32_t)(v * sendLevel); 2205 } 2206 // Delegate volume control to effect in track effect chain if needed 2207 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2208 // Do not ramp volume if volume is controlled by effect 2209 param = AudioMixer::VOLUME; 2210 track->mHasVolumeController = true; 2211 } else { 2212 // force no volume ramp when volume controller was just disabled or removed 2213 // from effect chain to avoid volume spike 2214 if (track->mHasVolumeController) { 2215 param = AudioMixer::VOLUME; 2216 } 2217 track->mHasVolumeController = false; 2218 } 2219 2220 // Convert volumes from 8.24 to 4.12 format 2221 int16_t left, right, aux; 2222 // This additional clamping is needed in case chain->setVolume_l() overshot 2223 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2224 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2225 left = int16_t(v_clamped); 2226 v_clamped = (vr + (1 << 11)) >> 12; 2227 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2228 right = int16_t(v_clamped); 2229 2230 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2231 aux = int16_t(va); 2232 2233 // XXX: these things DON'T need to be done each time 2234 mAudioMixer->setBufferProvider(name, track); 2235 mAudioMixer->enable(name); 2236 2237 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2238 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2239 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2240 mAudioMixer->setParameter( 2241 name, 2242 AudioMixer::TRACK, 2243 AudioMixer::FORMAT, (void *)track->format()); 2244 mAudioMixer->setParameter( 2245 name, 2246 AudioMixer::TRACK, 2247 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2248 mAudioMixer->setParameter( 2249 name, 2250 AudioMixer::RESAMPLE, 2251 AudioMixer::SAMPLE_RATE, 2252 (void *)(cblk->sampleRate)); 2253 mAudioMixer->setParameter( 2254 name, 2255 AudioMixer::TRACK, 2256 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2257 mAudioMixer->setParameter( 2258 name, 2259 AudioMixer::TRACK, 2260 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2261 2262 // reset retry count 2263 track->mRetryCount = kMaxTrackRetries; 2264 // If one track is ready, set the mixer ready if: 2265 // - the mixer was not ready during previous round OR 2266 // - no other track is not ready 2267 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2268 mixerStatus != MIXER_TRACKS_ENABLED) { 2269 mixerStatus = MIXER_TRACKS_READY; 2270 } 2271 } else { 2272 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2273 if (track->isStopped()) { 2274 track->reset(); 2275 } 2276 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2277 // We have consumed all the buffers of this track. 2278 // Remove it from the list of active tracks. 2279 tracksToRemove->add(track); 2280 } else { 2281 // No buffers for this track. Give it a few chances to 2282 // fill a buffer, then remove it from active list. 2283 if (--(track->mRetryCount) <= 0) { 2284 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2285 tracksToRemove->add(track); 2286 // indicate to client process that the track was disabled because of underrun 2287 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2288 // If one track is not ready, mark the mixer also not ready if: 2289 // - the mixer was ready during previous round OR 2290 // - no other track is ready 2291 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2292 mixerStatus != MIXER_TRACKS_READY) { 2293 mixerStatus = MIXER_TRACKS_ENABLED; 2294 } 2295 } 2296 mAudioMixer->disable(name); 2297 } 2298 } 2299 2300 // remove all the tracks that need to be... 2301 count = tracksToRemove->size(); 2302 if (CC_UNLIKELY(count)) { 2303 for (size_t i=0 ; i<count ; i++) { 2304 const sp<Track>& track = tracksToRemove->itemAt(i); 2305 mActiveTracks.remove(track); 2306 if (track->mainBuffer() != mMixBuffer) { 2307 chain = getEffectChain_l(track->sessionId()); 2308 if (chain != 0) { 2309 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2310 chain->decActiveTrackCnt(); 2311 } 2312 } 2313 if (track->isTerminated()) { 2314 removeTrack_l(track); 2315 } 2316 } 2317 } 2318 2319 // mix buffer must be cleared if all tracks are connected to an 2320 // effect chain as in this case the mixer will not write to 2321 // mix buffer and track effects will accumulate into it 2322 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2323 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2324 } 2325 2326 mPrevMixerStatus = mixerStatus; 2327 return mixerStatus; 2328} 2329 2330void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2331{ 2332 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2333 this, streamType, mTracks.size()); 2334 Mutex::Autolock _l(mLock); 2335 2336 size_t size = mTracks.size(); 2337 for (size_t i = 0; i < size; i++) { 2338 sp<Track> t = mTracks[i]; 2339 if (t->streamType() == streamType) { 2340 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2341 t->mCblk->cv.signal(); 2342 } 2343 } 2344} 2345 2346void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2347{ 2348 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2349 this, streamType, valid); 2350 Mutex::Autolock _l(mLock); 2351 2352 mStreamTypes[streamType].valid = valid; 2353} 2354 2355// getTrackName_l() must be called with ThreadBase::mLock held 2356int AudioFlinger::MixerThread::getTrackName_l() 2357{ 2358 return mAudioMixer->getTrackName(); 2359} 2360 2361// deleteTrackName_l() must be called with ThreadBase::mLock held 2362void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2363{ 2364 ALOGV("remove track (%d) and delete from mixer", name); 2365 mAudioMixer->deleteTrackName(name); 2366} 2367 2368// checkForNewParameters_l() must be called with ThreadBase::mLock held 2369bool AudioFlinger::MixerThread::checkForNewParameters_l() 2370{ 2371 bool reconfig = false; 2372 2373 while (!mNewParameters.isEmpty()) { 2374 status_t status = NO_ERROR; 2375 String8 keyValuePair = mNewParameters[0]; 2376 AudioParameter param = AudioParameter(keyValuePair); 2377 int value; 2378 2379 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2380 reconfig = true; 2381 } 2382 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2383 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2384 status = BAD_VALUE; 2385 } else { 2386 reconfig = true; 2387 } 2388 } 2389 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2390 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2391 status = BAD_VALUE; 2392 } else { 2393 reconfig = true; 2394 } 2395 } 2396 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2397 // do not accept frame count changes if tracks are open as the track buffer 2398 // size depends on frame count and correct behavior would not be guaranteed 2399 // if frame count is changed after track creation 2400 if (!mTracks.isEmpty()) { 2401 status = INVALID_OPERATION; 2402 } else { 2403 reconfig = true; 2404 } 2405 } 2406 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2407 // when changing the audio output device, call addBatteryData to notify 2408 // the change 2409 if ((int)mDevice != value) { 2410 uint32_t params = 0; 2411 // check whether speaker is on 2412 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2413 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2414 } 2415 2416 int deviceWithoutSpeaker 2417 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2418 // check if any other device (except speaker) is on 2419 if (value & deviceWithoutSpeaker ) { 2420 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2421 } 2422 2423 if (params != 0) { 2424 addBatteryData(params); 2425 } 2426 } 2427 2428 // forward device change to effects that have requested to be 2429 // aware of attached audio device. 2430 mDevice = (uint32_t)value; 2431 for (size_t i = 0; i < mEffectChains.size(); i++) { 2432 mEffectChains[i]->setDevice_l(mDevice); 2433 } 2434 } 2435 2436 if (status == NO_ERROR) { 2437 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2438 keyValuePair.string()); 2439 if (!mStandby && status == INVALID_OPERATION) { 2440 mOutput->stream->common.standby(&mOutput->stream->common); 2441 mStandby = true; 2442 mBytesWritten = 0; 2443 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2444 keyValuePair.string()); 2445 } 2446 if (status == NO_ERROR && reconfig) { 2447 delete mAudioMixer; 2448 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2449 mAudioMixer = NULL; 2450 readOutputParameters(); 2451 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2452 for (size_t i = 0; i < mTracks.size() ; i++) { 2453 int name = getTrackName_l(); 2454 if (name < 0) break; 2455 mTracks[i]->mName = name; 2456 // limit track sample rate to 2 x new output sample rate 2457 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2458 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2459 } 2460 } 2461 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2462 } 2463 } 2464 2465 mNewParameters.removeAt(0); 2466 2467 mParamStatus = status; 2468 mParamCond.signal(); 2469 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2470 // already timed out waiting for the status and will never signal the condition. 2471 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2472 } 2473 return reconfig; 2474} 2475 2476status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2477{ 2478 const size_t SIZE = 256; 2479 char buffer[SIZE]; 2480 String8 result; 2481 2482 PlaybackThread::dumpInternals(fd, args); 2483 2484 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2485 result.append(buffer); 2486 write(fd, result.string(), result.size()); 2487 return NO_ERROR; 2488} 2489 2490uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2491{ 2492 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2493} 2494 2495uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2496{ 2497 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2498} 2499 2500// ---------------------------------------------------------------------------- 2501AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2502 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2503 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2504 // mLeftVolFloat, mRightVolFloat 2505 // mLeftVolShort, mRightVolShort 2506{ 2507} 2508 2509AudioFlinger::DirectOutputThread::~DirectOutputThread() 2510{ 2511} 2512 2513void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2514{ 2515 // Do not apply volume on compressed audio 2516 if (!audio_is_linear_pcm(mFormat)) { 2517 return; 2518 } 2519 2520 // convert to signed 16 bit before volume calculation 2521 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2522 size_t count = mFrameCount * mChannelCount; 2523 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2524 int16_t *dst = mMixBuffer + count-1; 2525 while(count--) { 2526 *dst-- = (int16_t)(*src--^0x80) << 8; 2527 } 2528 } 2529 2530 size_t frameCount = mFrameCount; 2531 int16_t *out = mMixBuffer; 2532 if (ramp) { 2533 if (mChannelCount == 1) { 2534 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2535 int32_t vlInc = d / (int32_t)frameCount; 2536 int32_t vl = ((int32_t)mLeftVolShort << 16); 2537 do { 2538 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2539 out++; 2540 vl += vlInc; 2541 } while (--frameCount); 2542 2543 } else { 2544 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2545 int32_t vlInc = d / (int32_t)frameCount; 2546 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2547 int32_t vrInc = d / (int32_t)frameCount; 2548 int32_t vl = ((int32_t)mLeftVolShort << 16); 2549 int32_t vr = ((int32_t)mRightVolShort << 16); 2550 do { 2551 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2552 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2553 out += 2; 2554 vl += vlInc; 2555 vr += vrInc; 2556 } while (--frameCount); 2557 } 2558 } else { 2559 if (mChannelCount == 1) { 2560 do { 2561 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2562 out++; 2563 } while (--frameCount); 2564 } else { 2565 do { 2566 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2567 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2568 out += 2; 2569 } while (--frameCount); 2570 } 2571 } 2572 2573 // convert back to unsigned 8 bit after volume calculation 2574 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2575 size_t count = mFrameCount * mChannelCount; 2576 int16_t *src = mMixBuffer; 2577 uint8_t *dst = (uint8_t *)mMixBuffer; 2578 while(count--) { 2579 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2580 } 2581 } 2582 2583 mLeftVolShort = leftVol; 2584 mRightVolShort = rightVol; 2585} 2586 2587bool AudioFlinger::DirectOutputThread::threadLoop() 2588{ 2589 mixer_state mixerStatus = MIXER_IDLE; 2590 sp<Track> trackToRemove; 2591 sp<Track> activeTrack; 2592 nsecs_t standbyTime = systemTime(); 2593 int8_t *curBuf; 2594 size_t mixBufferSize = mFrameCount*mFrameSize; 2595 uint32_t activeSleepTime = activeSleepTimeUs(); 2596 uint32_t idleSleepTime = idleSleepTimeUs(); 2597 uint32_t sleepTime = idleSleepTime; 2598 // use shorter standby delay as on normal output to release 2599 // hardware resources as soon as possible 2600 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2601 2602 acquireWakeLock(); 2603 2604 while (!exitPending()) 2605 { 2606 bool rampVolume; 2607 uint16_t leftVol; 2608 uint16_t rightVol; 2609 Vector< sp<EffectChain> > effectChains; 2610 2611 processConfigEvents(); 2612 2613 mixerStatus = MIXER_IDLE; 2614 2615 { // scope for the mLock 2616 2617 Mutex::Autolock _l(mLock); 2618 2619 if (checkForNewParameters_l()) { 2620 mixBufferSize = mFrameCount*mFrameSize; 2621 activeSleepTime = activeSleepTimeUs(); 2622 idleSleepTime = idleSleepTimeUs(); 2623 standbyDelay = microseconds(activeSleepTime*2); 2624 } 2625 2626 // put audio hardware into standby after short delay 2627 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2628 mSuspended)) { 2629 // wait until we have something to do... 2630 if (!mStandby) { 2631 ALOGV("Audio hardware entering standby, mixer %p", this); 2632 mOutput->stream->common.standby(&mOutput->stream->common); 2633 mStandby = true; 2634 mBytesWritten = 0; 2635 } 2636 2637 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2638 // we're about to wait, flush the binder command buffer 2639 IPCThreadState::self()->flushCommands(); 2640 2641 if (exitPending()) break; 2642 2643 releaseWakeLock_l(); 2644 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2645 mWaitWorkCV.wait(mLock); 2646 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2647 acquireWakeLock_l(); 2648 2649 if (!mMasterMute) { 2650 char value[PROPERTY_VALUE_MAX]; 2651 property_get("ro.audio.silent", value, "0"); 2652 if (atoi(value)) { 2653 ALOGD("Silence is golden"); 2654 setMasterMute(true); 2655 } 2656 } 2657 2658 standbyTime = systemTime() + standbyDelay; 2659 sleepTime = idleSleepTime; 2660 continue; 2661 } 2662 } 2663 2664 effectChains = mEffectChains; 2665 2666 // find out which tracks need to be processed 2667 if (mActiveTracks.size() != 0) { 2668 sp<Track> t = mActiveTracks[0].promote(); 2669 if (t == 0) continue; 2670 2671 Track* const track = t.get(); 2672 audio_track_cblk_t* cblk = track->cblk(); 2673 2674 // The first time a track is added we wait 2675 // for all its buffers to be filled before processing it 2676 if (cblk->framesReady() && track->isReady() && 2677 !track->isPaused() && !track->isTerminated()) 2678 { 2679 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2680 2681 if (track->mFillingUpStatus == Track::FS_FILLED) { 2682 track->mFillingUpStatus = Track::FS_ACTIVE; 2683 mLeftVolFloat = mRightVolFloat = 0; 2684 mLeftVolShort = mRightVolShort = 0; 2685 if (track->mState == TrackBase::RESUMING) { 2686 track->mState = TrackBase::ACTIVE; 2687 rampVolume = true; 2688 } 2689 } else if (cblk->server != 0) { 2690 // If the track is stopped before the first frame was mixed, 2691 // do not apply ramp 2692 rampVolume = true; 2693 } 2694 // compute volume for this track 2695 float left, right; 2696 if (track->isMuted() || mMasterMute || track->isPausing() || 2697 mStreamTypes[track->streamType()].mute) { 2698 left = right = 0; 2699 if (track->isPausing()) { 2700 track->setPaused(); 2701 } 2702 } else { 2703 float typeVolume = mStreamTypes[track->streamType()].volume; 2704 float v = mMasterVolume * typeVolume; 2705 uint32_t vlr = cblk->getVolumeLR(); 2706 float v_clamped = v * (vlr & 0xFFFF); 2707 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2708 left = v_clamped/MAX_GAIN; 2709 v_clamped = v * (vlr >> 16); 2710 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2711 right = v_clamped/MAX_GAIN; 2712 } 2713 2714 if (left != mLeftVolFloat || right != mRightVolFloat) { 2715 mLeftVolFloat = left; 2716 mRightVolFloat = right; 2717 2718 // If audio HAL implements volume control, 2719 // force software volume to nominal value 2720 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2721 left = 1.0f; 2722 right = 1.0f; 2723 } 2724 2725 // Convert volumes from float to 8.24 2726 uint32_t vl = (uint32_t)(left * (1 << 24)); 2727 uint32_t vr = (uint32_t)(right * (1 << 24)); 2728 2729 // Delegate volume control to effect in track effect chain if needed 2730 // only one effect chain can be present on DirectOutputThread, so if 2731 // there is one, the track is connected to it 2732 if (!effectChains.isEmpty()) { 2733 // Do not ramp volume if volume is controlled by effect 2734 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2735 rampVolume = false; 2736 } 2737 } 2738 2739 // Convert volumes from 8.24 to 4.12 format 2740 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2741 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2742 leftVol = (uint16_t)v_clamped; 2743 v_clamped = (vr + (1 << 11)) >> 12; 2744 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2745 rightVol = (uint16_t)v_clamped; 2746 } else { 2747 leftVol = mLeftVolShort; 2748 rightVol = mRightVolShort; 2749 rampVolume = false; 2750 } 2751 2752 // reset retry count 2753 track->mRetryCount = kMaxTrackRetriesDirect; 2754 activeTrack = t; 2755 mixerStatus = MIXER_TRACKS_READY; 2756 } else { 2757 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2758 if (track->isStopped()) { 2759 track->reset(); 2760 } 2761 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2762 // We have consumed all the buffers of this track. 2763 // Remove it from the list of active tracks. 2764 trackToRemove = track; 2765 } else { 2766 // No buffers for this track. Give it a few chances to 2767 // fill a buffer, then remove it from active list. 2768 if (--(track->mRetryCount) <= 0) { 2769 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2770 trackToRemove = track; 2771 } else { 2772 mixerStatus = MIXER_TRACKS_ENABLED; 2773 } 2774 } 2775 } 2776 } 2777 2778 // remove all the tracks that need to be... 2779 if (CC_UNLIKELY(trackToRemove != 0)) { 2780 mActiveTracks.remove(trackToRemove); 2781 if (!effectChains.isEmpty()) { 2782 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2783 trackToRemove->sessionId()); 2784 effectChains[0]->decActiveTrackCnt(); 2785 } 2786 if (trackToRemove->isTerminated()) { 2787 removeTrack_l(trackToRemove); 2788 } 2789 } 2790 2791 lockEffectChains_l(effectChains); 2792 } 2793 2794 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2795 AudioBufferProvider::Buffer buffer; 2796 size_t frameCount = mFrameCount; 2797 curBuf = (int8_t *)mMixBuffer; 2798 // output audio to hardware 2799 while (frameCount) { 2800 buffer.frameCount = frameCount; 2801 activeTrack->getNextBuffer(&buffer); 2802 if (CC_UNLIKELY(buffer.raw == NULL)) { 2803 memset(curBuf, 0, frameCount * mFrameSize); 2804 break; 2805 } 2806 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2807 frameCount -= buffer.frameCount; 2808 curBuf += buffer.frameCount * mFrameSize; 2809 activeTrack->releaseBuffer(&buffer); 2810 } 2811 sleepTime = 0; 2812 standbyTime = systemTime() + standbyDelay; 2813 } else { 2814 if (sleepTime == 0) { 2815 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2816 sleepTime = activeSleepTime; 2817 } else { 2818 sleepTime = idleSleepTime; 2819 } 2820 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2821 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2822 sleepTime = 0; 2823 } 2824 } 2825 2826 if (mSuspended) { 2827 sleepTime = suspendSleepTimeUs(); 2828 } 2829 // sleepTime == 0 means we must write to audio hardware 2830 if (sleepTime == 0) { 2831 if (mixerStatus == MIXER_TRACKS_READY) { 2832 applyVolume(leftVol, rightVol, rampVolume); 2833 } 2834 for (size_t i = 0; i < effectChains.size(); i ++) { 2835 effectChains[i]->process_l(); 2836 } 2837 unlockEffectChains(effectChains); 2838 2839 mLastWriteTime = systemTime(); 2840 mInWrite = true; 2841 mBytesWritten += mixBufferSize; 2842 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2843 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2844 mNumWrites++; 2845 mInWrite = false; 2846 mStandby = false; 2847 } else { 2848 unlockEffectChains(effectChains); 2849 usleep(sleepTime); 2850 } 2851 2852 // finally let go of removed track, without the lock held 2853 // since we can't guarantee the destructors won't acquire that 2854 // same lock. 2855 trackToRemove.clear(); 2856 activeTrack.clear(); 2857 2858 // Effect chains will be actually deleted here if they were removed from 2859 // mEffectChains list during mixing or effects processing 2860 effectChains.clear(); 2861 } 2862 2863 if (!mStandby) { 2864 mOutput->stream->common.standby(&mOutput->stream->common); 2865 } 2866 2867 releaseWakeLock(); 2868 2869 ALOGV("DirectOutputThread %p exiting", this); 2870 return false; 2871} 2872 2873// getTrackName_l() must be called with ThreadBase::mLock held 2874int AudioFlinger::DirectOutputThread::getTrackName_l() 2875{ 2876 return 0; 2877} 2878 2879// deleteTrackName_l() must be called with ThreadBase::mLock held 2880void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2881{ 2882} 2883 2884// checkForNewParameters_l() must be called with ThreadBase::mLock held 2885bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2886{ 2887 bool reconfig = false; 2888 2889 while (!mNewParameters.isEmpty()) { 2890 status_t status = NO_ERROR; 2891 String8 keyValuePair = mNewParameters[0]; 2892 AudioParameter param = AudioParameter(keyValuePair); 2893 int value; 2894 2895 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2896 // do not accept frame count changes if tracks are open as the track buffer 2897 // size depends on frame count and correct behavior would not be garantied 2898 // if frame count is changed after track creation 2899 if (!mTracks.isEmpty()) { 2900 status = INVALID_OPERATION; 2901 } else { 2902 reconfig = true; 2903 } 2904 } 2905 if (status == NO_ERROR) { 2906 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2907 keyValuePair.string()); 2908 if (!mStandby && status == INVALID_OPERATION) { 2909 mOutput->stream->common.standby(&mOutput->stream->common); 2910 mStandby = true; 2911 mBytesWritten = 0; 2912 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2913 keyValuePair.string()); 2914 } 2915 if (status == NO_ERROR && reconfig) { 2916 readOutputParameters(); 2917 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2918 } 2919 } 2920 2921 mNewParameters.removeAt(0); 2922 2923 mParamStatus = status; 2924 mParamCond.signal(); 2925 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2926 // already timed out waiting for the status and will never signal the condition. 2927 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2928 } 2929 return reconfig; 2930} 2931 2932uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2933{ 2934 uint32_t time; 2935 if (audio_is_linear_pcm(mFormat)) { 2936 time = PlaybackThread::activeSleepTimeUs(); 2937 } else { 2938 time = 10000; 2939 } 2940 return time; 2941} 2942 2943uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2944{ 2945 uint32_t time; 2946 if (audio_is_linear_pcm(mFormat)) { 2947 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2948 } else { 2949 time = 10000; 2950 } 2951 return time; 2952} 2953 2954uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2955{ 2956 uint32_t time; 2957 if (audio_is_linear_pcm(mFormat)) { 2958 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2959 } else { 2960 time = 10000; 2961 } 2962 return time; 2963} 2964 2965 2966// ---------------------------------------------------------------------------- 2967 2968AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 2969 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 2970 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 2971 mWaitTimeMs(UINT_MAX) 2972{ 2973 addOutputTrack(mainThread); 2974} 2975 2976AudioFlinger::DuplicatingThread::~DuplicatingThread() 2977{ 2978 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2979 mOutputTracks[i]->destroy(); 2980 } 2981} 2982 2983bool AudioFlinger::DuplicatingThread::threadLoop() 2984{ 2985 Vector< sp<Track> > tracksToRemove; 2986 mixer_state mixerStatus = MIXER_IDLE; 2987 nsecs_t standbyTime = systemTime(); 2988 size_t mixBufferSize = mFrameCount*mFrameSize; 2989 SortedVector< sp<OutputTrack> > outputTracks; 2990 uint32_t writeFrames = 0; 2991 uint32_t activeSleepTime = activeSleepTimeUs(); 2992 uint32_t idleSleepTime = idleSleepTimeUs(); 2993 uint32_t sleepTime = idleSleepTime; 2994 Vector< sp<EffectChain> > effectChains; 2995 2996 acquireWakeLock(); 2997 2998 while (!exitPending()) 2999 { 3000 processConfigEvents(); 3001 3002 mixerStatus = MIXER_IDLE; 3003 { // scope for the mLock 3004 3005 Mutex::Autolock _l(mLock); 3006 3007 if (checkForNewParameters_l()) { 3008 mixBufferSize = mFrameCount*mFrameSize; 3009 updateWaitTime(); 3010 activeSleepTime = activeSleepTimeUs(); 3011 idleSleepTime = idleSleepTimeUs(); 3012 } 3013 3014 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3015 3016 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3017 outputTracks.add(mOutputTracks[i]); 3018 } 3019 3020 // put audio hardware into standby after short delay 3021 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3022 mSuspended)) { 3023 if (!mStandby) { 3024 for (size_t i = 0; i < outputTracks.size(); i++) { 3025 outputTracks[i]->stop(); 3026 } 3027 mStandby = true; 3028 mBytesWritten = 0; 3029 } 3030 3031 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3032 // we're about to wait, flush the binder command buffer 3033 IPCThreadState::self()->flushCommands(); 3034 outputTracks.clear(); 3035 3036 if (exitPending()) break; 3037 3038 releaseWakeLock_l(); 3039 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3040 mWaitWorkCV.wait(mLock); 3041 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3042 acquireWakeLock_l(); 3043 3044 mPrevMixerStatus = MIXER_IDLE; 3045 if (!mMasterMute) { 3046 char value[PROPERTY_VALUE_MAX]; 3047 property_get("ro.audio.silent", value, "0"); 3048 if (atoi(value)) { 3049 ALOGD("Silence is golden"); 3050 setMasterMute(true); 3051 } 3052 } 3053 3054 standbyTime = systemTime() + kStandbyTimeInNsecs; 3055 sleepTime = idleSleepTime; 3056 continue; 3057 } 3058 } 3059 3060 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3061 3062 // prevent any changes in effect chain list and in each effect chain 3063 // during mixing and effect process as the audio buffers could be deleted 3064 // or modified if an effect is created or deleted 3065 lockEffectChains_l(effectChains); 3066 } 3067 3068 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3069 // mix buffers... 3070 if (outputsReady(outputTracks)) { 3071 mAudioMixer->process(); 3072 } else { 3073 memset(mMixBuffer, 0, mixBufferSize); 3074 } 3075 sleepTime = 0; 3076 writeFrames = mFrameCount; 3077 } else { 3078 if (sleepTime == 0) { 3079 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3080 sleepTime = activeSleepTime; 3081 } else { 3082 sleepTime = idleSleepTime; 3083 } 3084 } else if (mBytesWritten != 0) { 3085 // flush remaining overflow buffers in output tracks 3086 for (size_t i = 0; i < outputTracks.size(); i++) { 3087 if (outputTracks[i]->isActive()) { 3088 sleepTime = 0; 3089 writeFrames = 0; 3090 memset(mMixBuffer, 0, mixBufferSize); 3091 break; 3092 } 3093 } 3094 } 3095 } 3096 3097 if (mSuspended) { 3098 sleepTime = suspendSleepTimeUs(); 3099 } 3100 // sleepTime == 0 means we must write to audio hardware 3101 if (sleepTime == 0) { 3102 for (size_t i = 0; i < effectChains.size(); i ++) { 3103 effectChains[i]->process_l(); 3104 } 3105 // enable changes in effect chain 3106 unlockEffectChains(effectChains); 3107 3108 standbyTime = systemTime() + kStandbyTimeInNsecs; 3109 for (size_t i = 0; i < outputTracks.size(); i++) { 3110 outputTracks[i]->write(mMixBuffer, writeFrames); 3111 } 3112 mStandby = false; 3113 mBytesWritten += mixBufferSize; 3114 } else { 3115 // enable changes in effect chain 3116 unlockEffectChains(effectChains); 3117 usleep(sleepTime); 3118 } 3119 3120 // finally let go of all our tracks, without the lock held 3121 // since we can't guarantee the destructors won't acquire that 3122 // same lock. 3123 tracksToRemove.clear(); 3124 outputTracks.clear(); 3125 3126 // Effect chains will be actually deleted here if they were removed from 3127 // mEffectChains list during mixing or effects processing 3128 effectChains.clear(); 3129 } 3130 3131 releaseWakeLock(); 3132 3133 return false; 3134} 3135 3136void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3137{ 3138 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3139 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3140 this, 3141 mSampleRate, 3142 mFormat, 3143 mChannelMask, 3144 frameCount); 3145 if (outputTrack->cblk() != NULL) { 3146 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3147 mOutputTracks.add(outputTrack); 3148 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3149 updateWaitTime(); 3150 } 3151} 3152 3153void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3154{ 3155 Mutex::Autolock _l(mLock); 3156 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3157 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3158 mOutputTracks[i]->destroy(); 3159 mOutputTracks.removeAt(i); 3160 updateWaitTime(); 3161 return; 3162 } 3163 } 3164 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3165} 3166 3167void AudioFlinger::DuplicatingThread::updateWaitTime() 3168{ 3169 mWaitTimeMs = UINT_MAX; 3170 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3171 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3172 if (strong != 0) { 3173 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3174 if (waitTimeMs < mWaitTimeMs) { 3175 mWaitTimeMs = waitTimeMs; 3176 } 3177 } 3178 } 3179} 3180 3181 3182bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3183{ 3184 for (size_t i = 0; i < outputTracks.size(); i++) { 3185 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3186 if (thread == 0) { 3187 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3188 return false; 3189 } 3190 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3191 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3192 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3193 return false; 3194 } 3195 } 3196 return true; 3197} 3198 3199uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3200{ 3201 return (mWaitTimeMs * 1000) / 2; 3202} 3203 3204// ---------------------------------------------------------------------------- 3205 3206// TrackBase constructor must be called with AudioFlinger::mLock held 3207AudioFlinger::ThreadBase::TrackBase::TrackBase( 3208 const wp<ThreadBase>& thread, 3209 const sp<Client>& client, 3210 uint32_t sampleRate, 3211 audio_format_t format, 3212 uint32_t channelMask, 3213 int frameCount, 3214 uint32_t flags, 3215 const sp<IMemory>& sharedBuffer, 3216 int sessionId) 3217 : RefBase(), 3218 mThread(thread), 3219 mClient(client), 3220 mCblk(NULL), 3221 // mBuffer 3222 // mBufferEnd 3223 mFrameCount(0), 3224 mState(IDLE), 3225 mFormat(format), 3226 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3227 mSessionId(sessionId) 3228 // mChannelCount 3229 // mChannelMask 3230{ 3231 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3232 3233 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3234 size_t size = sizeof(audio_track_cblk_t); 3235 uint8_t channelCount = popcount(channelMask); 3236 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3237 if (sharedBuffer == 0) { 3238 size += bufferSize; 3239 } 3240 3241 if (client != NULL) { 3242 mCblkMemory = client->heap()->allocate(size); 3243 if (mCblkMemory != 0) { 3244 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3245 if (mCblk != NULL) { // construct the shared structure in-place. 3246 new(mCblk) audio_track_cblk_t(); 3247 // clear all buffers 3248 mCblk->frameCount = frameCount; 3249 mCblk->sampleRate = sampleRate; 3250 mChannelCount = channelCount; 3251 mChannelMask = channelMask; 3252 if (sharedBuffer == 0) { 3253 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3254 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3255 // Force underrun condition to avoid false underrun callback until first data is 3256 // written to buffer (other flags are cleared) 3257 mCblk->flags = CBLK_UNDERRUN_ON; 3258 } else { 3259 mBuffer = sharedBuffer->pointer(); 3260 } 3261 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3262 } 3263 } else { 3264 ALOGE("not enough memory for AudioTrack size=%u", size); 3265 client->heap()->dump("AudioTrack"); 3266 return; 3267 } 3268 } else { 3269 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3270 // construct the shared structure in-place. 3271 new(mCblk) audio_track_cblk_t(); 3272 // clear all buffers 3273 mCblk->frameCount = frameCount; 3274 mCblk->sampleRate = sampleRate; 3275 mChannelCount = channelCount; 3276 mChannelMask = channelMask; 3277 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3278 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3279 // Force underrun condition to avoid false underrun callback until first data is 3280 // written to buffer (other flags are cleared) 3281 mCblk->flags = CBLK_UNDERRUN_ON; 3282 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3283 } 3284} 3285 3286AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3287{ 3288 if (mCblk != NULL) { 3289 if (mClient == 0) { 3290 delete mCblk; 3291 } else { 3292 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3293 } 3294 } 3295 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3296 if (mClient != 0) { 3297 // Client destructor must run with AudioFlinger mutex locked 3298 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3299 // If the client's reference count drops to zero, the associated destructor 3300 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3301 // relying on the automatic clear() at end of scope. 3302 mClient.clear(); 3303 } 3304} 3305 3306void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3307{ 3308 buffer->raw = NULL; 3309 mFrameCount = buffer->frameCount; 3310 step(); 3311 buffer->frameCount = 0; 3312} 3313 3314bool AudioFlinger::ThreadBase::TrackBase::step() { 3315 bool result; 3316 audio_track_cblk_t* cblk = this->cblk(); 3317 3318 result = cblk->stepServer(mFrameCount); 3319 if (!result) { 3320 ALOGV("stepServer failed acquiring cblk mutex"); 3321 mFlags |= STEPSERVER_FAILED; 3322 } 3323 return result; 3324} 3325 3326void AudioFlinger::ThreadBase::TrackBase::reset() { 3327 audio_track_cblk_t* cblk = this->cblk(); 3328 3329 cblk->user = 0; 3330 cblk->server = 0; 3331 cblk->userBase = 0; 3332 cblk->serverBase = 0; 3333 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3334 ALOGV("TrackBase::reset"); 3335} 3336 3337int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3338 return (int)mCblk->sampleRate; 3339} 3340 3341void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3342 audio_track_cblk_t* cblk = this->cblk(); 3343 size_t frameSize = cblk->frameSize; 3344 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3345 int8_t *bufferEnd = bufferStart + frames * frameSize; 3346 3347 // Check validity of returned pointer in case the track control block would have been corrupted. 3348 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3349 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3350 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3351 server %d, serverBase %d, user %d, userBase %d", 3352 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3353 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3354 return NULL; 3355 } 3356 3357 return bufferStart; 3358} 3359 3360// ---------------------------------------------------------------------------- 3361 3362// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3363AudioFlinger::PlaybackThread::Track::Track( 3364 const wp<ThreadBase>& thread, 3365 const sp<Client>& client, 3366 audio_stream_type_t streamType, 3367 uint32_t sampleRate, 3368 audio_format_t format, 3369 uint32_t channelMask, 3370 int frameCount, 3371 const sp<IMemory>& sharedBuffer, 3372 int sessionId) 3373 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3374 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3375 mAuxEffectId(0), mHasVolumeController(false) 3376{ 3377 if (mCblk != NULL) { 3378 sp<ThreadBase> baseThread = thread.promote(); 3379 if (baseThread != 0) { 3380 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3381 mName = playbackThread->getTrackName_l(); 3382 mMainBuffer = playbackThread->mixBuffer(); 3383 } 3384 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3385 if (mName < 0) { 3386 ALOGE("no more track names available"); 3387 } 3388 mStreamType = streamType; 3389 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3390 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3391 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3392 } 3393} 3394 3395AudioFlinger::PlaybackThread::Track::~Track() 3396{ 3397 ALOGV("PlaybackThread::Track destructor"); 3398 sp<ThreadBase> thread = mThread.promote(); 3399 if (thread != 0) { 3400 Mutex::Autolock _l(thread->mLock); 3401 mState = TERMINATED; 3402 } 3403} 3404 3405void AudioFlinger::PlaybackThread::Track::destroy() 3406{ 3407 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3408 // by removing it from mTracks vector, so there is a risk that this Tracks's 3409 // desctructor is called. As the destructor needs to lock mLock, 3410 // we must acquire a strong reference on this Track before locking mLock 3411 // here so that the destructor is called only when exiting this function. 3412 // On the other hand, as long as Track::destroy() is only called by 3413 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3414 // this Track with its member mTrack. 3415 sp<Track> keep(this); 3416 { // scope for mLock 3417 sp<ThreadBase> thread = mThread.promote(); 3418 if (thread != 0) { 3419 if (!isOutputTrack()) { 3420 if (mState == ACTIVE || mState == RESUMING) { 3421 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3422 3423 // to track the speaker usage 3424 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3425 } 3426 AudioSystem::releaseOutput(thread->id()); 3427 } 3428 Mutex::Autolock _l(thread->mLock); 3429 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3430 playbackThread->destroyTrack_l(this); 3431 } 3432 } 3433} 3434 3435void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3436{ 3437 uint32_t vlr = mCblk->getVolumeLR(); 3438 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", 3439 mName - AudioMixer::TRACK0, 3440 (mClient == 0) ? getpid() : mClient->pid(), 3441 mStreamType, 3442 mFormat, 3443 mChannelMask, 3444 mSessionId, 3445 mFrameCount, 3446 mState, 3447 mMute, 3448 mFillingUpStatus, 3449 mCblk->sampleRate, 3450 vlr & 0xFFFF, 3451 vlr >> 16, 3452 mCblk->server, 3453 mCblk->user, 3454 (int)mMainBuffer, 3455 (int)mAuxBuffer); 3456} 3457 3458status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3459{ 3460 audio_track_cblk_t* cblk = this->cblk(); 3461 uint32_t framesReady; 3462 uint32_t framesReq = buffer->frameCount; 3463 3464 // Check if last stepServer failed, try to step now 3465 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3466 if (!step()) goto getNextBuffer_exit; 3467 ALOGV("stepServer recovered"); 3468 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3469 } 3470 3471 framesReady = cblk->framesReady(); 3472 3473 if (CC_LIKELY(framesReady)) { 3474 uint32_t s = cblk->server; 3475 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3476 3477 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3478 if (framesReq > framesReady) { 3479 framesReq = framesReady; 3480 } 3481 if (s + framesReq > bufferEnd) { 3482 framesReq = bufferEnd - s; 3483 } 3484 3485 buffer->raw = getBuffer(s, framesReq); 3486 if (buffer->raw == NULL) goto getNextBuffer_exit; 3487 3488 buffer->frameCount = framesReq; 3489 return NO_ERROR; 3490 } 3491 3492getNextBuffer_exit: 3493 buffer->raw = NULL; 3494 buffer->frameCount = 0; 3495 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3496 return NOT_ENOUGH_DATA; 3497} 3498 3499bool AudioFlinger::PlaybackThread::Track::isReady() const { 3500 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3501 3502 if (mCblk->framesReady() >= mCblk->frameCount || 3503 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3504 mFillingUpStatus = FS_FILLED; 3505 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3506 return true; 3507 } 3508 return false; 3509} 3510 3511status_t AudioFlinger::PlaybackThread::Track::start() 3512{ 3513 status_t status = NO_ERROR; 3514 ALOGV("start(%d), calling pid %d session %d", 3515 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3516 sp<ThreadBase> thread = mThread.promote(); 3517 if (thread != 0) { 3518 Mutex::Autolock _l(thread->mLock); 3519 track_state state = mState; 3520 // here the track could be either new, or restarted 3521 // in both cases "unstop" the track 3522 if (mState == PAUSED) { 3523 mState = TrackBase::RESUMING; 3524 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3525 } else { 3526 mState = TrackBase::ACTIVE; 3527 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3528 } 3529 3530 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3531 thread->mLock.unlock(); 3532 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3533 thread->mLock.lock(); 3534 3535 // to track the speaker usage 3536 if (status == NO_ERROR) { 3537 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3538 } 3539 } 3540 if (status == NO_ERROR) { 3541 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3542 playbackThread->addTrack_l(this); 3543 } else { 3544 mState = state; 3545 } 3546 } else { 3547 status = BAD_VALUE; 3548 } 3549 return status; 3550} 3551 3552void AudioFlinger::PlaybackThread::Track::stop() 3553{ 3554 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3555 sp<ThreadBase> thread = mThread.promote(); 3556 if (thread != 0) { 3557 Mutex::Autolock _l(thread->mLock); 3558 track_state state = mState; 3559 if (mState > STOPPED) { 3560 mState = STOPPED; 3561 // If the track is not active (PAUSED and buffers full), flush buffers 3562 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3563 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3564 reset(); 3565 } 3566 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3567 } 3568 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3569 thread->mLock.unlock(); 3570 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3571 thread->mLock.lock(); 3572 3573 // to track the speaker usage 3574 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3575 } 3576 } 3577} 3578 3579void AudioFlinger::PlaybackThread::Track::pause() 3580{ 3581 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3582 sp<ThreadBase> thread = mThread.promote(); 3583 if (thread != 0) { 3584 Mutex::Autolock _l(thread->mLock); 3585 if (mState == ACTIVE || mState == RESUMING) { 3586 mState = PAUSING; 3587 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3588 if (!isOutputTrack()) { 3589 thread->mLock.unlock(); 3590 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3591 thread->mLock.lock(); 3592 3593 // to track the speaker usage 3594 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3595 } 3596 } 3597 } 3598} 3599 3600void AudioFlinger::PlaybackThread::Track::flush() 3601{ 3602 ALOGV("flush(%d)", mName); 3603 sp<ThreadBase> thread = mThread.promote(); 3604 if (thread != 0) { 3605 Mutex::Autolock _l(thread->mLock); 3606 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3607 return; 3608 } 3609 // No point remaining in PAUSED state after a flush => go to 3610 // STOPPED state 3611 mState = STOPPED; 3612 3613 // do not reset the track if it is still in the process of being stopped or paused. 3614 // this will be done by prepareTracks_l() when the track is stopped. 3615 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3616 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3617 reset(); 3618 } 3619 } 3620} 3621 3622void AudioFlinger::PlaybackThread::Track::reset() 3623{ 3624 // Do not reset twice to avoid discarding data written just after a flush and before 3625 // the audioflinger thread detects the track is stopped. 3626 if (!mResetDone) { 3627 TrackBase::reset(); 3628 // Force underrun condition to avoid false underrun callback until first data is 3629 // written to buffer 3630 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3631 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3632 mFillingUpStatus = FS_FILLING; 3633 mResetDone = true; 3634 } 3635} 3636 3637void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3638{ 3639 mMute = muted; 3640} 3641 3642status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3643{ 3644 status_t status = DEAD_OBJECT; 3645 sp<ThreadBase> thread = mThread.promote(); 3646 if (thread != 0) { 3647 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3648 status = playbackThread->attachAuxEffect(this, EffectId); 3649 } 3650 return status; 3651} 3652 3653void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3654{ 3655 mAuxEffectId = EffectId; 3656 mAuxBuffer = buffer; 3657} 3658 3659// ---------------------------------------------------------------------------- 3660 3661// RecordTrack constructor must be called with AudioFlinger::mLock held 3662AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3663 const wp<ThreadBase>& thread, 3664 const sp<Client>& client, 3665 uint32_t sampleRate, 3666 audio_format_t format, 3667 uint32_t channelMask, 3668 int frameCount, 3669 uint32_t flags, 3670 int sessionId) 3671 : TrackBase(thread, client, sampleRate, format, 3672 channelMask, frameCount, flags, 0, sessionId), 3673 mOverflow(false) 3674{ 3675 if (mCblk != NULL) { 3676 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3677 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3678 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3679 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3680 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3681 } else { 3682 mCblk->frameSize = sizeof(int8_t); 3683 } 3684 } 3685} 3686 3687AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3688{ 3689 sp<ThreadBase> thread = mThread.promote(); 3690 if (thread != 0) { 3691 AudioSystem::releaseInput(thread->id()); 3692 } 3693} 3694 3695status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3696{ 3697 audio_track_cblk_t* cblk = this->cblk(); 3698 uint32_t framesAvail; 3699 uint32_t framesReq = buffer->frameCount; 3700 3701 // Check if last stepServer failed, try to step now 3702 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3703 if (!step()) goto getNextBuffer_exit; 3704 ALOGV("stepServer recovered"); 3705 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3706 } 3707 3708 framesAvail = cblk->framesAvailable_l(); 3709 3710 if (CC_LIKELY(framesAvail)) { 3711 uint32_t s = cblk->server; 3712 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3713 3714 if (framesReq > framesAvail) { 3715 framesReq = framesAvail; 3716 } 3717 if (s + framesReq > bufferEnd) { 3718 framesReq = bufferEnd - s; 3719 } 3720 3721 buffer->raw = getBuffer(s, framesReq); 3722 if (buffer->raw == NULL) goto getNextBuffer_exit; 3723 3724 buffer->frameCount = framesReq; 3725 return NO_ERROR; 3726 } 3727 3728getNextBuffer_exit: 3729 buffer->raw = NULL; 3730 buffer->frameCount = 0; 3731 return NOT_ENOUGH_DATA; 3732} 3733 3734status_t AudioFlinger::RecordThread::RecordTrack::start() 3735{ 3736 sp<ThreadBase> thread = mThread.promote(); 3737 if (thread != 0) { 3738 RecordThread *recordThread = (RecordThread *)thread.get(); 3739 return recordThread->start(this); 3740 } else { 3741 return BAD_VALUE; 3742 } 3743} 3744 3745void AudioFlinger::RecordThread::RecordTrack::stop() 3746{ 3747 sp<ThreadBase> thread = mThread.promote(); 3748 if (thread != 0) { 3749 RecordThread *recordThread = (RecordThread *)thread.get(); 3750 recordThread->stop(this); 3751 TrackBase::reset(); 3752 // Force overerrun condition to avoid false overrun callback until first data is 3753 // read from buffer 3754 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3755 } 3756} 3757 3758void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3759{ 3760 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3761 (mClient == 0) ? getpid() : mClient->pid(), 3762 mFormat, 3763 mChannelMask, 3764 mSessionId, 3765 mFrameCount, 3766 mState, 3767 mCblk->sampleRate, 3768 mCblk->server, 3769 mCblk->user); 3770} 3771 3772 3773// ---------------------------------------------------------------------------- 3774 3775AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3776 const wp<ThreadBase>& thread, 3777 DuplicatingThread *sourceThread, 3778 uint32_t sampleRate, 3779 audio_format_t format, 3780 uint32_t channelMask, 3781 int frameCount) 3782 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3783 mActive(false), mSourceThread(sourceThread) 3784{ 3785 3786 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3787 if (mCblk != NULL) { 3788 mCblk->flags |= CBLK_DIRECTION_OUT; 3789 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3790 mOutBuffer.frameCount = 0; 3791 playbackThread->mTracks.add(this); 3792 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3793 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3794 mCblk, mBuffer, mCblk->buffers, 3795 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3796 } else { 3797 ALOGW("Error creating output track on thread %p", playbackThread); 3798 } 3799} 3800 3801AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3802{ 3803 clearBufferQueue(); 3804} 3805 3806status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3807{ 3808 status_t status = Track::start(); 3809 if (status != NO_ERROR) { 3810 return status; 3811 } 3812 3813 mActive = true; 3814 mRetryCount = 127; 3815 return status; 3816} 3817 3818void AudioFlinger::PlaybackThread::OutputTrack::stop() 3819{ 3820 Track::stop(); 3821 clearBufferQueue(); 3822 mOutBuffer.frameCount = 0; 3823 mActive = false; 3824} 3825 3826bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3827{ 3828 Buffer *pInBuffer; 3829 Buffer inBuffer; 3830 uint32_t channelCount = mChannelCount; 3831 bool outputBufferFull = false; 3832 inBuffer.frameCount = frames; 3833 inBuffer.i16 = data; 3834 3835 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3836 3837 if (!mActive && frames != 0) { 3838 start(); 3839 sp<ThreadBase> thread = mThread.promote(); 3840 if (thread != 0) { 3841 MixerThread *mixerThread = (MixerThread *)thread.get(); 3842 if (mCblk->frameCount > frames){ 3843 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3844 uint32_t startFrames = (mCblk->frameCount - frames); 3845 pInBuffer = new Buffer; 3846 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3847 pInBuffer->frameCount = startFrames; 3848 pInBuffer->i16 = pInBuffer->mBuffer; 3849 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3850 mBufferQueue.add(pInBuffer); 3851 } else { 3852 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3853 } 3854 } 3855 } 3856 } 3857 3858 while (waitTimeLeftMs) { 3859 // First write pending buffers, then new data 3860 if (mBufferQueue.size()) { 3861 pInBuffer = mBufferQueue.itemAt(0); 3862 } else { 3863 pInBuffer = &inBuffer; 3864 } 3865 3866 if (pInBuffer->frameCount == 0) { 3867 break; 3868 } 3869 3870 if (mOutBuffer.frameCount == 0) { 3871 mOutBuffer.frameCount = pInBuffer->frameCount; 3872 nsecs_t startTime = systemTime(); 3873 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3874 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3875 outputBufferFull = true; 3876 break; 3877 } 3878 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3879 if (waitTimeLeftMs >= waitTimeMs) { 3880 waitTimeLeftMs -= waitTimeMs; 3881 } else { 3882 waitTimeLeftMs = 0; 3883 } 3884 } 3885 3886 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3887 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3888 mCblk->stepUser(outFrames); 3889 pInBuffer->frameCount -= outFrames; 3890 pInBuffer->i16 += outFrames * channelCount; 3891 mOutBuffer.frameCount -= outFrames; 3892 mOutBuffer.i16 += outFrames * channelCount; 3893 3894 if (pInBuffer->frameCount == 0) { 3895 if (mBufferQueue.size()) { 3896 mBufferQueue.removeAt(0); 3897 delete [] pInBuffer->mBuffer; 3898 delete pInBuffer; 3899 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3900 } else { 3901 break; 3902 } 3903 } 3904 } 3905 3906 // If we could not write all frames, allocate a buffer and queue it for next time. 3907 if (inBuffer.frameCount) { 3908 sp<ThreadBase> thread = mThread.promote(); 3909 if (thread != 0 && !thread->standby()) { 3910 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3911 pInBuffer = new Buffer; 3912 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3913 pInBuffer->frameCount = inBuffer.frameCount; 3914 pInBuffer->i16 = pInBuffer->mBuffer; 3915 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3916 mBufferQueue.add(pInBuffer); 3917 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3918 } else { 3919 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3920 } 3921 } 3922 } 3923 3924 // Calling write() with a 0 length buffer, means that no more data will be written: 3925 // If no more buffers are pending, fill output track buffer to make sure it is started 3926 // by output mixer. 3927 if (frames == 0 && mBufferQueue.size() == 0) { 3928 if (mCblk->user < mCblk->frameCount) { 3929 frames = mCblk->frameCount - mCblk->user; 3930 pInBuffer = new Buffer; 3931 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3932 pInBuffer->frameCount = frames; 3933 pInBuffer->i16 = pInBuffer->mBuffer; 3934 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3935 mBufferQueue.add(pInBuffer); 3936 } else if (mActive) { 3937 stop(); 3938 } 3939 } 3940 3941 return outputBufferFull; 3942} 3943 3944status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3945{ 3946 int active; 3947 status_t result; 3948 audio_track_cblk_t* cblk = mCblk; 3949 uint32_t framesReq = buffer->frameCount; 3950 3951// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3952 buffer->frameCount = 0; 3953 3954 uint32_t framesAvail = cblk->framesAvailable(); 3955 3956 3957 if (framesAvail == 0) { 3958 Mutex::Autolock _l(cblk->lock); 3959 goto start_loop_here; 3960 while (framesAvail == 0) { 3961 active = mActive; 3962 if (CC_UNLIKELY(!active)) { 3963 ALOGV("Not active and NO_MORE_BUFFERS"); 3964 return NO_MORE_BUFFERS; 3965 } 3966 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3967 if (result != NO_ERROR) { 3968 return NO_MORE_BUFFERS; 3969 } 3970 // read the server count again 3971 start_loop_here: 3972 framesAvail = cblk->framesAvailable_l(); 3973 } 3974 } 3975 3976// if (framesAvail < framesReq) { 3977// return NO_MORE_BUFFERS; 3978// } 3979 3980 if (framesReq > framesAvail) { 3981 framesReq = framesAvail; 3982 } 3983 3984 uint32_t u = cblk->user; 3985 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3986 3987 if (u + framesReq > bufferEnd) { 3988 framesReq = bufferEnd - u; 3989 } 3990 3991 buffer->frameCount = framesReq; 3992 buffer->raw = (void *)cblk->buffer(u); 3993 return NO_ERROR; 3994} 3995 3996 3997void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3998{ 3999 size_t size = mBufferQueue.size(); 4000 Buffer *pBuffer; 4001 4002 for (size_t i = 0; i < size; i++) { 4003 pBuffer = mBufferQueue.itemAt(i); 4004 delete [] pBuffer->mBuffer; 4005 delete pBuffer; 4006 } 4007 mBufferQueue.clear(); 4008} 4009 4010// ---------------------------------------------------------------------------- 4011 4012AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4013 : RefBase(), 4014 mAudioFlinger(audioFlinger), 4015 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4016 mPid(pid) 4017{ 4018 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4019} 4020 4021// Client destructor must be called with AudioFlinger::mLock held 4022AudioFlinger::Client::~Client() 4023{ 4024 mAudioFlinger->removeClient_l(mPid); 4025} 4026 4027sp<MemoryDealer> AudioFlinger::Client::heap() const 4028{ 4029 return mMemoryDealer; 4030} 4031 4032// ---------------------------------------------------------------------------- 4033 4034AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4035 const sp<IAudioFlingerClient>& client, 4036 pid_t pid) 4037 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4038{ 4039} 4040 4041AudioFlinger::NotificationClient::~NotificationClient() 4042{ 4043} 4044 4045void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4046{ 4047 sp<NotificationClient> keep(this); 4048 { 4049 mAudioFlinger->removeNotificationClient(mPid); 4050 } 4051} 4052 4053// ---------------------------------------------------------------------------- 4054 4055AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4056 : BnAudioTrack(), 4057 mTrack(track) 4058{ 4059} 4060 4061AudioFlinger::TrackHandle::~TrackHandle() { 4062 // just stop the track on deletion, associated resources 4063 // will be freed from the main thread once all pending buffers have 4064 // been played. Unless it's not in the active track list, in which 4065 // case we free everything now... 4066 mTrack->destroy(); 4067} 4068 4069sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4070 return mTrack->getCblk(); 4071} 4072 4073status_t AudioFlinger::TrackHandle::start() { 4074 return mTrack->start(); 4075} 4076 4077void AudioFlinger::TrackHandle::stop() { 4078 mTrack->stop(); 4079} 4080 4081void AudioFlinger::TrackHandle::flush() { 4082 mTrack->flush(); 4083} 4084 4085void AudioFlinger::TrackHandle::mute(bool e) { 4086 mTrack->mute(e); 4087} 4088 4089void AudioFlinger::TrackHandle::pause() { 4090 mTrack->pause(); 4091} 4092 4093status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4094{ 4095 return mTrack->attachAuxEffect(EffectId); 4096} 4097 4098status_t AudioFlinger::TrackHandle::onTransact( 4099 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4100{ 4101 return BnAudioTrack::onTransact(code, data, reply, flags); 4102} 4103 4104// ---------------------------------------------------------------------------- 4105 4106sp<IAudioRecord> AudioFlinger::openRecord( 4107 pid_t pid, 4108 audio_io_handle_t input, 4109 uint32_t sampleRate, 4110 audio_format_t format, 4111 uint32_t channelMask, 4112 int frameCount, 4113 uint32_t flags, 4114 int *sessionId, 4115 status_t *status) 4116{ 4117 sp<RecordThread::RecordTrack> recordTrack; 4118 sp<RecordHandle> recordHandle; 4119 sp<Client> client; 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 client = registerPid_l(pid); 4141 4142 // If no audio session id is provided, create one here 4143 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4144 lSessionId = *sessionId; 4145 } else { 4146 lSessionId = nextUniqueId(); 4147 if (sessionId != NULL) { 4148 *sessionId = lSessionId; 4149 } 4150 } 4151 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4152 recordTrack = thread->createRecordTrack_l(client, 4153 sampleRate, 4154 format, 4155 channelMask, 4156 frameCount, 4157 flags, 4158 lSessionId, 4159 &lStatus); 4160 } 4161 if (lStatus != NO_ERROR) { 4162 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4163 // destructor is called by the TrackBase destructor with mLock held 4164 client.clear(); 4165 recordTrack.clear(); 4166 goto Exit; 4167 } 4168 4169 // return to handle to client 4170 recordHandle = new RecordHandle(recordTrack); 4171 lStatus = NO_ERROR; 4172 4173Exit: 4174 if (status) { 4175 *status = lStatus; 4176 } 4177 return recordHandle; 4178} 4179 4180// ---------------------------------------------------------------------------- 4181 4182AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4183 : BnAudioRecord(), 4184 mRecordTrack(recordTrack) 4185{ 4186} 4187 4188AudioFlinger::RecordHandle::~RecordHandle() { 4189 stop(); 4190} 4191 4192sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4193 return mRecordTrack->getCblk(); 4194} 4195 4196status_t AudioFlinger::RecordHandle::start() { 4197 ALOGV("RecordHandle::start()"); 4198 return mRecordTrack->start(); 4199} 4200 4201void AudioFlinger::RecordHandle::stop() { 4202 ALOGV("RecordHandle::stop()"); 4203 mRecordTrack->stop(); 4204} 4205 4206status_t AudioFlinger::RecordHandle::onTransact( 4207 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4208{ 4209 return BnAudioRecord::onTransact(code, data, reply, flags); 4210} 4211 4212// ---------------------------------------------------------------------------- 4213 4214AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4215 AudioStreamIn *input, 4216 uint32_t sampleRate, 4217 uint32_t channels, 4218 audio_io_handle_t id, 4219 uint32_t device) : 4220 ThreadBase(audioFlinger, id, device, RECORD), 4221 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4222 // mRsmpInIndex and mInputBytes set by readInputParameters() 4223 mReqChannelCount(popcount(channels)), 4224 mReqSampleRate(sampleRate) 4225 // mBytesRead is only meaningful while active, and so is cleared in start() 4226 // (but might be better to also clear here for dump?) 4227{ 4228 snprintf(mName, kNameLength, "AudioIn_%d", id); 4229 4230 readInputParameters(); 4231} 4232 4233 4234AudioFlinger::RecordThread::~RecordThread() 4235{ 4236 delete[] mRsmpInBuffer; 4237 delete mResampler; 4238 delete[] mRsmpOutBuffer; 4239} 4240 4241void AudioFlinger::RecordThread::onFirstRef() 4242{ 4243 run(mName, PRIORITY_URGENT_AUDIO); 4244} 4245 4246status_t AudioFlinger::RecordThread::readyToRun() 4247{ 4248 status_t status = initCheck(); 4249 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4250 return status; 4251} 4252 4253bool AudioFlinger::RecordThread::threadLoop() 4254{ 4255 AudioBufferProvider::Buffer buffer; 4256 sp<RecordTrack> activeTrack; 4257 Vector< sp<EffectChain> > effectChains; 4258 4259 nsecs_t lastWarning = 0; 4260 4261 acquireWakeLock(); 4262 4263 // start recording 4264 while (!exitPending()) { 4265 4266 processConfigEvents(); 4267 4268 { // scope for mLock 4269 Mutex::Autolock _l(mLock); 4270 checkForNewParameters_l(); 4271 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4272 if (!mStandby) { 4273 mInput->stream->common.standby(&mInput->stream->common); 4274 mStandby = true; 4275 } 4276 4277 if (exitPending()) break; 4278 4279 releaseWakeLock_l(); 4280 ALOGV("RecordThread: loop stopping"); 4281 // go to sleep 4282 mWaitWorkCV.wait(mLock); 4283 ALOGV("RecordThread: loop starting"); 4284 acquireWakeLock_l(); 4285 continue; 4286 } 4287 if (mActiveTrack != 0) { 4288 if (mActiveTrack->mState == TrackBase::PAUSING) { 4289 if (!mStandby) { 4290 mInput->stream->common.standby(&mInput->stream->common); 4291 mStandby = true; 4292 } 4293 mActiveTrack.clear(); 4294 mStartStopCond.broadcast(); 4295 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4296 if (mReqChannelCount != mActiveTrack->channelCount()) { 4297 mActiveTrack.clear(); 4298 mStartStopCond.broadcast(); 4299 } else if (mBytesRead != 0) { 4300 // record start succeeds only if first read from audio input 4301 // succeeds 4302 if (mBytesRead > 0) { 4303 mActiveTrack->mState = TrackBase::ACTIVE; 4304 } else { 4305 mActiveTrack.clear(); 4306 } 4307 mStartStopCond.broadcast(); 4308 } 4309 mStandby = false; 4310 } 4311 } 4312 lockEffectChains_l(effectChains); 4313 } 4314 4315 if (mActiveTrack != 0) { 4316 if (mActiveTrack->mState != TrackBase::ACTIVE && 4317 mActiveTrack->mState != TrackBase::RESUMING) { 4318 unlockEffectChains(effectChains); 4319 usleep(kRecordThreadSleepUs); 4320 continue; 4321 } 4322 for (size_t i = 0; i < effectChains.size(); i ++) { 4323 effectChains[i]->process_l(); 4324 } 4325 4326 buffer.frameCount = mFrameCount; 4327 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4328 size_t framesOut = buffer.frameCount; 4329 if (mResampler == NULL) { 4330 // no resampling 4331 while (framesOut) { 4332 size_t framesIn = mFrameCount - mRsmpInIndex; 4333 if (framesIn) { 4334 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4335 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4336 if (framesIn > framesOut) 4337 framesIn = framesOut; 4338 mRsmpInIndex += framesIn; 4339 framesOut -= framesIn; 4340 if ((int)mChannelCount == mReqChannelCount || 4341 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4342 memcpy(dst, src, framesIn * mFrameSize); 4343 } else { 4344 int16_t *src16 = (int16_t *)src; 4345 int16_t *dst16 = (int16_t *)dst; 4346 if (mChannelCount == 1) { 4347 while (framesIn--) { 4348 *dst16++ = *src16; 4349 *dst16++ = *src16++; 4350 } 4351 } else { 4352 while (framesIn--) { 4353 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4354 src16 += 2; 4355 } 4356 } 4357 } 4358 } 4359 if (framesOut && mFrameCount == mRsmpInIndex) { 4360 if (framesOut == mFrameCount && 4361 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4362 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4363 framesOut = 0; 4364 } else { 4365 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4366 mRsmpInIndex = 0; 4367 } 4368 if (mBytesRead < 0) { 4369 ALOGE("Error reading audio input"); 4370 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4371 // Force input into standby so that it tries to 4372 // recover at next read attempt 4373 mInput->stream->common.standby(&mInput->stream->common); 4374 usleep(kRecordThreadSleepUs); 4375 } 4376 mRsmpInIndex = mFrameCount; 4377 framesOut = 0; 4378 buffer.frameCount = 0; 4379 } 4380 } 4381 } 4382 } else { 4383 // resampling 4384 4385 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4386 // alter output frame count as if we were expecting stereo samples 4387 if (mChannelCount == 1 && mReqChannelCount == 1) { 4388 framesOut >>= 1; 4389 } 4390 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4391 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4392 // are 32 bit aligned which should be always true. 4393 if (mChannelCount == 2 && mReqChannelCount == 1) { 4394 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4395 // the resampler always outputs stereo samples: do post stereo to mono conversion 4396 int16_t *src = (int16_t *)mRsmpOutBuffer; 4397 int16_t *dst = buffer.i16; 4398 while (framesOut--) { 4399 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4400 src += 2; 4401 } 4402 } else { 4403 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4404 } 4405 4406 } 4407 mActiveTrack->releaseBuffer(&buffer); 4408 mActiveTrack->overflow(); 4409 } 4410 // client isn't retrieving buffers fast enough 4411 else { 4412 if (!mActiveTrack->setOverflow()) { 4413 nsecs_t now = systemTime(); 4414 if ((now - lastWarning) > kWarningThrottleNs) { 4415 ALOGW("RecordThread: buffer overflow"); 4416 lastWarning = now; 4417 } 4418 } 4419 // Release the processor for a while before asking for a new buffer. 4420 // This will give the application more chance to read from the buffer and 4421 // clear the overflow. 4422 usleep(kRecordThreadSleepUs); 4423 } 4424 } 4425 // enable changes in effect chain 4426 unlockEffectChains(effectChains); 4427 effectChains.clear(); 4428 } 4429 4430 if (!mStandby) { 4431 mInput->stream->common.standby(&mInput->stream->common); 4432 } 4433 mActiveTrack.clear(); 4434 4435 mStartStopCond.broadcast(); 4436 4437 releaseWakeLock(); 4438 4439 ALOGV("RecordThread %p exiting", this); 4440 return false; 4441} 4442 4443 4444sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4445 const sp<AudioFlinger::Client>& client, 4446 uint32_t sampleRate, 4447 audio_format_t format, 4448 int channelMask, 4449 int frameCount, 4450 uint32_t flags, 4451 int sessionId, 4452 status_t *status) 4453{ 4454 sp<RecordTrack> track; 4455 status_t lStatus; 4456 4457 lStatus = initCheck(); 4458 if (lStatus != NO_ERROR) { 4459 ALOGE("Audio driver not initialized."); 4460 goto Exit; 4461 } 4462 4463 { // scope for mLock 4464 Mutex::Autolock _l(mLock); 4465 4466 track = new RecordTrack(this, client, sampleRate, 4467 format, channelMask, frameCount, flags, sessionId); 4468 4469 if (track->getCblk() == 0) { 4470 lStatus = NO_MEMORY; 4471 goto Exit; 4472 } 4473 4474 mTrack = track.get(); 4475 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4476 bool suspend = audio_is_bluetooth_sco_device( 4477 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4478 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4479 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4480 } 4481 lStatus = NO_ERROR; 4482 4483Exit: 4484 if (status) { 4485 *status = lStatus; 4486 } 4487 return track; 4488} 4489 4490status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4491{ 4492 ALOGV("RecordThread::start"); 4493 sp <ThreadBase> strongMe = this; 4494 status_t status = NO_ERROR; 4495 { 4496 AutoMutex lock(mLock); 4497 if (mActiveTrack != 0) { 4498 if (recordTrack != mActiveTrack.get()) { 4499 status = -EBUSY; 4500 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4501 mActiveTrack->mState = TrackBase::ACTIVE; 4502 } 4503 return status; 4504 } 4505 4506 recordTrack->mState = TrackBase::IDLE; 4507 mActiveTrack = recordTrack; 4508 mLock.unlock(); 4509 status_t status = AudioSystem::startInput(mId); 4510 mLock.lock(); 4511 if (status != NO_ERROR) { 4512 mActiveTrack.clear(); 4513 return status; 4514 } 4515 mRsmpInIndex = mFrameCount; 4516 mBytesRead = 0; 4517 if (mResampler != NULL) { 4518 mResampler->reset(); 4519 } 4520 mActiveTrack->mState = TrackBase::RESUMING; 4521 // signal thread to start 4522 ALOGV("Signal record thread"); 4523 mWaitWorkCV.signal(); 4524 // do not wait for mStartStopCond if exiting 4525 if (exitPending()) { 4526 mActiveTrack.clear(); 4527 status = INVALID_OPERATION; 4528 goto startError; 4529 } 4530 mStartStopCond.wait(mLock); 4531 if (mActiveTrack == 0) { 4532 ALOGV("Record failed to start"); 4533 status = BAD_VALUE; 4534 goto startError; 4535 } 4536 ALOGV("Record started OK"); 4537 return status; 4538 } 4539startError: 4540 AudioSystem::stopInput(mId); 4541 return status; 4542} 4543 4544void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4545 ALOGV("RecordThread::stop"); 4546 sp <ThreadBase> strongMe = this; 4547 { 4548 AutoMutex lock(mLock); 4549 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4550 mActiveTrack->mState = TrackBase::PAUSING; 4551 // do not wait for mStartStopCond if exiting 4552 if (exitPending()) { 4553 return; 4554 } 4555 mStartStopCond.wait(mLock); 4556 // if we have been restarted, recordTrack == mActiveTrack.get() here 4557 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4558 mLock.unlock(); 4559 AudioSystem::stopInput(mId); 4560 mLock.lock(); 4561 ALOGV("Record stopped OK"); 4562 } 4563 } 4564 } 4565} 4566 4567status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4568{ 4569 const size_t SIZE = 256; 4570 char buffer[SIZE]; 4571 String8 result; 4572 4573 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4574 result.append(buffer); 4575 4576 if (mActiveTrack != 0) { 4577 result.append("Active Track:\n"); 4578 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4579 mActiveTrack->dump(buffer, SIZE); 4580 result.append(buffer); 4581 4582 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4583 result.append(buffer); 4584 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4585 result.append(buffer); 4586 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4587 result.append(buffer); 4588 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4589 result.append(buffer); 4590 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4591 result.append(buffer); 4592 4593 4594 } else { 4595 result.append("No record client\n"); 4596 } 4597 write(fd, result.string(), result.size()); 4598 4599 dumpBase(fd, args); 4600 dumpEffectChains(fd, args); 4601 4602 return NO_ERROR; 4603} 4604 4605status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4606{ 4607 size_t framesReq = buffer->frameCount; 4608 size_t framesReady = mFrameCount - mRsmpInIndex; 4609 int channelCount; 4610 4611 if (framesReady == 0) { 4612 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4613 if (mBytesRead < 0) { 4614 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4615 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4616 // Force input into standby so that it tries to 4617 // recover at next read attempt 4618 mInput->stream->common.standby(&mInput->stream->common); 4619 usleep(kRecordThreadSleepUs); 4620 } 4621 buffer->raw = NULL; 4622 buffer->frameCount = 0; 4623 return NOT_ENOUGH_DATA; 4624 } 4625 mRsmpInIndex = 0; 4626 framesReady = mFrameCount; 4627 } 4628 4629 if (framesReq > framesReady) { 4630 framesReq = framesReady; 4631 } 4632 4633 if (mChannelCount == 1 && mReqChannelCount == 2) { 4634 channelCount = 1; 4635 } else { 4636 channelCount = 2; 4637 } 4638 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4639 buffer->frameCount = framesReq; 4640 return NO_ERROR; 4641} 4642 4643void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4644{ 4645 mRsmpInIndex += buffer->frameCount; 4646 buffer->frameCount = 0; 4647} 4648 4649bool AudioFlinger::RecordThread::checkForNewParameters_l() 4650{ 4651 bool reconfig = false; 4652 4653 while (!mNewParameters.isEmpty()) { 4654 status_t status = NO_ERROR; 4655 String8 keyValuePair = mNewParameters[0]; 4656 AudioParameter param = AudioParameter(keyValuePair); 4657 int value; 4658 audio_format_t reqFormat = mFormat; 4659 int reqSamplingRate = mReqSampleRate; 4660 int reqChannelCount = mReqChannelCount; 4661 4662 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4663 reqSamplingRate = value; 4664 reconfig = true; 4665 } 4666 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4667 reqFormat = (audio_format_t) value; 4668 reconfig = true; 4669 } 4670 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4671 reqChannelCount = popcount(value); 4672 reconfig = true; 4673 } 4674 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4675 // do not accept frame count changes if tracks are open as the track buffer 4676 // size depends on frame count and correct behavior would not be garantied 4677 // if frame count is changed after track creation 4678 if (mActiveTrack != 0) { 4679 status = INVALID_OPERATION; 4680 } else { 4681 reconfig = true; 4682 } 4683 } 4684 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4685 // forward device change to effects that have requested to be 4686 // aware of attached audio device. 4687 for (size_t i = 0; i < mEffectChains.size(); i++) { 4688 mEffectChains[i]->setDevice_l(value); 4689 } 4690 // store input device and output device but do not forward output device to audio HAL. 4691 // Note that status is ignored by the caller for output device 4692 // (see AudioFlinger::setParameters() 4693 if (value & AUDIO_DEVICE_OUT_ALL) { 4694 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4695 status = BAD_VALUE; 4696 } else { 4697 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4698 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4699 if (mTrack != NULL) { 4700 bool suspend = audio_is_bluetooth_sco_device( 4701 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4702 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4703 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4704 } 4705 } 4706 mDevice |= (uint32_t)value; 4707 } 4708 if (status == NO_ERROR) { 4709 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4710 if (status == INVALID_OPERATION) { 4711 mInput->stream->common.standby(&mInput->stream->common); 4712 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4713 } 4714 if (reconfig) { 4715 if (status == BAD_VALUE && 4716 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4717 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4718 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4719 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4720 (reqChannelCount < 3)) { 4721 status = NO_ERROR; 4722 } 4723 if (status == NO_ERROR) { 4724 readInputParameters(); 4725 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4726 } 4727 } 4728 } 4729 4730 mNewParameters.removeAt(0); 4731 4732 mParamStatus = status; 4733 mParamCond.signal(); 4734 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4735 // already timed out waiting for the status and will never signal the condition. 4736 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4737 } 4738 return reconfig; 4739} 4740 4741String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4742{ 4743 char *s; 4744 String8 out_s8 = String8(); 4745 4746 Mutex::Autolock _l(mLock); 4747 if (initCheck() != NO_ERROR) { 4748 return out_s8; 4749 } 4750 4751 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4752 out_s8 = String8(s); 4753 free(s); 4754 return out_s8; 4755} 4756 4757void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4758 AudioSystem::OutputDescriptor desc; 4759 void *param2 = NULL; 4760 4761 switch (event) { 4762 case AudioSystem::INPUT_OPENED: 4763 case AudioSystem::INPUT_CONFIG_CHANGED: 4764 desc.channels = mChannelMask; 4765 desc.samplingRate = mSampleRate; 4766 desc.format = mFormat; 4767 desc.frameCount = mFrameCount; 4768 desc.latency = 0; 4769 param2 = &desc; 4770 break; 4771 4772 case AudioSystem::INPUT_CLOSED: 4773 default: 4774 break; 4775 } 4776 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4777} 4778 4779void AudioFlinger::RecordThread::readInputParameters() 4780{ 4781 delete mRsmpInBuffer; 4782 // mRsmpInBuffer is always assigned a new[] below 4783 delete mRsmpOutBuffer; 4784 mRsmpOutBuffer = NULL; 4785 delete mResampler; 4786 mResampler = NULL; 4787 4788 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4789 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4790 mChannelCount = (uint16_t)popcount(mChannelMask); 4791 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4792 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4793 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4794 mFrameCount = mInputBytes / mFrameSize; 4795 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4796 4797 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4798 { 4799 int channelCount; 4800 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4801 // stereo to mono post process as the resampler always outputs stereo. 4802 if (mChannelCount == 1 && mReqChannelCount == 2) { 4803 channelCount = 1; 4804 } else { 4805 channelCount = 2; 4806 } 4807 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4808 mResampler->setSampleRate(mSampleRate); 4809 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4810 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4811 4812 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4813 if (mChannelCount == 1 && mReqChannelCount == 1) { 4814 mFrameCount >>= 1; 4815 } 4816 4817 } 4818 mRsmpInIndex = mFrameCount; 4819} 4820 4821unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4822{ 4823 Mutex::Autolock _l(mLock); 4824 if (initCheck() != NO_ERROR) { 4825 return 0; 4826 } 4827 4828 return mInput->stream->get_input_frames_lost(mInput->stream); 4829} 4830 4831uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4832{ 4833 Mutex::Autolock _l(mLock); 4834 uint32_t result = 0; 4835 if (getEffectChain_l(sessionId) != 0) { 4836 result = EFFECT_SESSION; 4837 } 4838 4839 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4840 result |= TRACK_SESSION; 4841 } 4842 4843 return result; 4844} 4845 4846AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4847{ 4848 Mutex::Autolock _l(mLock); 4849 return mTrack; 4850} 4851 4852AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4853{ 4854 Mutex::Autolock _l(mLock); 4855 return mInput; 4856} 4857 4858AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4859{ 4860 Mutex::Autolock _l(mLock); 4861 AudioStreamIn *input = mInput; 4862 mInput = NULL; 4863 return input; 4864} 4865 4866// this method must always be called either with ThreadBase mLock held or inside the thread loop 4867audio_stream_t* AudioFlinger::RecordThread::stream() 4868{ 4869 if (mInput == NULL) { 4870 return NULL; 4871 } 4872 return &mInput->stream->common; 4873} 4874 4875 4876// ---------------------------------------------------------------------------- 4877 4878audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 4879 uint32_t *pSamplingRate, 4880 audio_format_t *pFormat, 4881 uint32_t *pChannels, 4882 uint32_t *pLatencyMs, 4883 uint32_t flags) 4884{ 4885 status_t status; 4886 PlaybackThread *thread = NULL; 4887 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4888 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4889 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4890 uint32_t channels = pChannels ? *pChannels : 0; 4891 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4892 audio_stream_out_t *outStream; 4893 audio_hw_device_t *outHwDev; 4894 4895 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4896 pDevices ? *pDevices : 0, 4897 samplingRate, 4898 format, 4899 channels, 4900 flags); 4901 4902 if (pDevices == NULL || *pDevices == 0) { 4903 return 0; 4904 } 4905 4906 Mutex::Autolock _l(mLock); 4907 4908 outHwDev = findSuitableHwDev_l(*pDevices); 4909 if (outHwDev == NULL) 4910 return 0; 4911 4912 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4913 &channels, &samplingRate, &outStream); 4914 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4915 outStream, 4916 samplingRate, 4917 format, 4918 channels, 4919 status); 4920 4921 mHardwareStatus = AUDIO_HW_IDLE; 4922 if (outStream != NULL) { 4923 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4924 audio_io_handle_t id = nextUniqueId(); 4925 4926 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4927 (format != AUDIO_FORMAT_PCM_16_BIT) || 4928 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4929 thread = new DirectOutputThread(this, output, id, *pDevices); 4930 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4931 } else { 4932 thread = new MixerThread(this, output, id, *pDevices); 4933 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4934 } 4935 mPlaybackThreads.add(id, thread); 4936 4937 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 4938 if (pFormat != NULL) *pFormat = format; 4939 if (pChannels != NULL) *pChannels = channels; 4940 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 4941 4942 // notify client processes of the new output creation 4943 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4944 return id; 4945 } 4946 4947 return 0; 4948} 4949 4950audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 4951 audio_io_handle_t output2) 4952{ 4953 Mutex::Autolock _l(mLock); 4954 MixerThread *thread1 = checkMixerThread_l(output1); 4955 MixerThread *thread2 = checkMixerThread_l(output2); 4956 4957 if (thread1 == NULL || thread2 == NULL) { 4958 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4959 return 0; 4960 } 4961 4962 audio_io_handle_t id = nextUniqueId(); 4963 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4964 thread->addOutputTrack(thread2); 4965 mPlaybackThreads.add(id, thread); 4966 // notify client processes of the new output creation 4967 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4968 return id; 4969} 4970 4971status_t AudioFlinger::closeOutput(audio_io_handle_t output) 4972{ 4973 // keep strong reference on the playback thread so that 4974 // it is not destroyed while exit() is executed 4975 sp <PlaybackThread> thread; 4976 { 4977 Mutex::Autolock _l(mLock); 4978 thread = checkPlaybackThread_l(output); 4979 if (thread == NULL) { 4980 return BAD_VALUE; 4981 } 4982 4983 ALOGV("closeOutput() %d", output); 4984 4985 if (thread->type() == ThreadBase::MIXER) { 4986 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4987 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4988 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4989 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4990 } 4991 } 4992 } 4993 void *param2 = NULL; 4994 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4995 mPlaybackThreads.removeItem(output); 4996 } 4997 thread->exit(); 4998 // The thread entity (active unit of execution) is no longer running here, 4999 // but the ThreadBase container still exists. 5000 5001 if (thread->type() != ThreadBase::DUPLICATING) { 5002 AudioStreamOut *out = thread->clearOutput(); 5003 assert(out != NULL); 5004 // from now on thread->mOutput is NULL 5005 out->hwDev->close_output_stream(out->hwDev, out->stream); 5006 delete out; 5007 } 5008 return NO_ERROR; 5009} 5010 5011status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5012{ 5013 Mutex::Autolock _l(mLock); 5014 PlaybackThread *thread = checkPlaybackThread_l(output); 5015 5016 if (thread == NULL) { 5017 return BAD_VALUE; 5018 } 5019 5020 ALOGV("suspendOutput() %d", output); 5021 thread->suspend(); 5022 5023 return NO_ERROR; 5024} 5025 5026status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5027{ 5028 Mutex::Autolock _l(mLock); 5029 PlaybackThread *thread = checkPlaybackThread_l(output); 5030 5031 if (thread == NULL) { 5032 return BAD_VALUE; 5033 } 5034 5035 ALOGV("restoreOutput() %d", output); 5036 5037 thread->restore(); 5038 5039 return NO_ERROR; 5040} 5041 5042audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5043 uint32_t *pSamplingRate, 5044 audio_format_t *pFormat, 5045 uint32_t *pChannels, 5046 audio_in_acoustics_t acoustics) 5047{ 5048 status_t status; 5049 RecordThread *thread = NULL; 5050 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5051 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5052 uint32_t channels = pChannels ? *pChannels : 0; 5053 uint32_t reqSamplingRate = samplingRate; 5054 audio_format_t reqFormat = format; 5055 uint32_t reqChannels = channels; 5056 audio_stream_in_t *inStream; 5057 audio_hw_device_t *inHwDev; 5058 5059 if (pDevices == NULL || *pDevices == 0) { 5060 return 0; 5061 } 5062 5063 Mutex::Autolock _l(mLock); 5064 5065 inHwDev = findSuitableHwDev_l(*pDevices); 5066 if (inHwDev == NULL) 5067 return 0; 5068 5069 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5070 &channels, &samplingRate, 5071 acoustics, 5072 &inStream); 5073 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5074 inStream, 5075 samplingRate, 5076 format, 5077 channels, 5078 acoustics, 5079 status); 5080 5081 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5082 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5083 // or stereo to mono conversions on 16 bit PCM inputs. 5084 if (inStream == NULL && status == BAD_VALUE && 5085 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5086 (samplingRate <= 2 * reqSamplingRate) && 5087 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5088 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5089 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5090 &channels, &samplingRate, 5091 acoustics, 5092 &inStream); 5093 } 5094 5095 if (inStream != NULL) { 5096 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5097 5098 audio_io_handle_t id = nextUniqueId(); 5099 // Start record thread 5100 // RecorThread require both input and output device indication to forward to audio 5101 // pre processing modules 5102 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5103 thread = new RecordThread(this, 5104 input, 5105 reqSamplingRate, 5106 reqChannels, 5107 id, 5108 device); 5109 mRecordThreads.add(id, thread); 5110 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5111 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5112 if (pFormat != NULL) *pFormat = format; 5113 if (pChannels != NULL) *pChannels = reqChannels; 5114 5115 input->stream->common.standby(&input->stream->common); 5116 5117 // notify client processes of the new input creation 5118 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5119 return id; 5120 } 5121 5122 return 0; 5123} 5124 5125status_t AudioFlinger::closeInput(audio_io_handle_t input) 5126{ 5127 // keep strong reference on the record thread so that 5128 // it is not destroyed while exit() is executed 5129 sp <RecordThread> thread; 5130 { 5131 Mutex::Autolock _l(mLock); 5132 thread = checkRecordThread_l(input); 5133 if (thread == NULL) { 5134 return BAD_VALUE; 5135 } 5136 5137 ALOGV("closeInput() %d", input); 5138 void *param2 = NULL; 5139 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5140 mRecordThreads.removeItem(input); 5141 } 5142 thread->exit(); 5143 // The thread entity (active unit of execution) is no longer running here, 5144 // but the ThreadBase container still exists. 5145 5146 AudioStreamIn *in = thread->clearInput(); 5147 assert(in != NULL); 5148 // from now on thread->mInput is NULL 5149 in->hwDev->close_input_stream(in->hwDev, in->stream); 5150 delete in; 5151 5152 return NO_ERROR; 5153} 5154 5155status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5156{ 5157 Mutex::Autolock _l(mLock); 5158 MixerThread *dstThread = checkMixerThread_l(output); 5159 if (dstThread == NULL) { 5160 ALOGW("setStreamOutput() bad output id %d", output); 5161 return BAD_VALUE; 5162 } 5163 5164 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5165 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5166 5167 dstThread->setStreamValid(stream, true); 5168 5169 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5170 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5171 if (thread != dstThread && 5172 thread->type() != ThreadBase::DIRECT) { 5173 MixerThread *srcThread = (MixerThread *)thread; 5174 srcThread->setStreamValid(stream, false); 5175 srcThread->invalidateTracks(stream); 5176 } 5177 } 5178 5179 return NO_ERROR; 5180} 5181 5182 5183int AudioFlinger::newAudioSessionId() 5184{ 5185 return nextUniqueId(); 5186} 5187 5188void AudioFlinger::acquireAudioSessionId(int audioSession) 5189{ 5190 Mutex::Autolock _l(mLock); 5191 pid_t caller = IPCThreadState::self()->getCallingPid(); 5192 ALOGV("acquiring %d from %d", audioSession, caller); 5193 int num = mAudioSessionRefs.size(); 5194 for (int i = 0; i< num; i++) { 5195 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5196 if (ref->sessionid == audioSession && ref->pid == caller) { 5197 ref->cnt++; 5198 ALOGV(" incremented refcount to %d", ref->cnt); 5199 return; 5200 } 5201 } 5202 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5203 ALOGV(" added new entry for %d", audioSession); 5204} 5205 5206void AudioFlinger::releaseAudioSessionId(int audioSession) 5207{ 5208 Mutex::Autolock _l(mLock); 5209 pid_t caller = IPCThreadState::self()->getCallingPid(); 5210 ALOGV("releasing %d from %d", audioSession, caller); 5211 int num = mAudioSessionRefs.size(); 5212 for (int i = 0; i< num; i++) { 5213 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5214 if (ref->sessionid == audioSession && ref->pid == caller) { 5215 ref->cnt--; 5216 ALOGV(" decremented refcount to %d", ref->cnt); 5217 if (ref->cnt == 0) { 5218 mAudioSessionRefs.removeAt(i); 5219 delete ref; 5220 purgeStaleEffects_l(); 5221 } 5222 return; 5223 } 5224 } 5225 ALOGW("session id %d not found for pid %d", audioSession, caller); 5226} 5227 5228void AudioFlinger::purgeStaleEffects_l() { 5229 5230 ALOGV("purging stale effects"); 5231 5232 Vector< sp<EffectChain> > chains; 5233 5234 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5235 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5236 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5237 sp<EffectChain> ec = t->mEffectChains[j]; 5238 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5239 chains.push(ec); 5240 } 5241 } 5242 } 5243 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5244 sp<RecordThread> t = mRecordThreads.valueAt(i); 5245 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5246 sp<EffectChain> ec = t->mEffectChains[j]; 5247 chains.push(ec); 5248 } 5249 } 5250 5251 for (size_t i = 0; i < chains.size(); i++) { 5252 sp<EffectChain> ec = chains[i]; 5253 int sessionid = ec->sessionId(); 5254 sp<ThreadBase> t = ec->mThread.promote(); 5255 if (t == 0) { 5256 continue; 5257 } 5258 size_t numsessionrefs = mAudioSessionRefs.size(); 5259 bool found = false; 5260 for (size_t k = 0; k < numsessionrefs; k++) { 5261 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5262 if (ref->sessionid == sessionid) { 5263 ALOGV(" session %d still exists for %d with %d refs", 5264 sessionid, ref->pid, ref->cnt); 5265 found = true; 5266 break; 5267 } 5268 } 5269 if (!found) { 5270 // remove all effects from the chain 5271 while (ec->mEffects.size()) { 5272 sp<EffectModule> effect = ec->mEffects[0]; 5273 effect->unPin(); 5274 Mutex::Autolock _l (t->mLock); 5275 t->removeEffect_l(effect); 5276 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5277 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5278 if (handle != 0) { 5279 handle->mEffect.clear(); 5280 if (handle->mHasControl && handle->mEnabled) { 5281 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5282 } 5283 } 5284 } 5285 AudioSystem::unregisterEffect(effect->id()); 5286 } 5287 } 5288 } 5289 return; 5290} 5291 5292// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5293AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5294{ 5295 PlaybackThread *thread = NULL; 5296 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5297 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5298 } 5299 return thread; 5300} 5301 5302// checkMixerThread_l() must be called with AudioFlinger::mLock held 5303AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5304{ 5305 PlaybackThread *thread = checkPlaybackThread_l(output); 5306 if (thread != NULL) { 5307 if (thread->type() == ThreadBase::DIRECT) { 5308 thread = NULL; 5309 } 5310 } 5311 return (MixerThread *)thread; 5312} 5313 5314// checkRecordThread_l() must be called with AudioFlinger::mLock held 5315AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5316{ 5317 RecordThread *thread = NULL; 5318 if (mRecordThreads.indexOfKey(input) >= 0) { 5319 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5320 } 5321 return thread; 5322} 5323 5324uint32_t AudioFlinger::nextUniqueId() 5325{ 5326 return android_atomic_inc(&mNextUniqueId); 5327} 5328 5329AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5330{ 5331 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5332 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5333 AudioStreamOut *output = thread->getOutput(); 5334 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5335 return thread; 5336 } 5337 } 5338 return NULL; 5339} 5340 5341uint32_t AudioFlinger::primaryOutputDevice_l() 5342{ 5343 PlaybackThread *thread = primaryPlaybackThread_l(); 5344 5345 if (thread == NULL) { 5346 return 0; 5347 } 5348 5349 return thread->device(); 5350} 5351 5352 5353// ---------------------------------------------------------------------------- 5354// Effect management 5355// ---------------------------------------------------------------------------- 5356 5357 5358status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5359{ 5360 Mutex::Autolock _l(mLock); 5361 return EffectQueryNumberEffects(numEffects); 5362} 5363 5364status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5365{ 5366 Mutex::Autolock _l(mLock); 5367 return EffectQueryEffect(index, descriptor); 5368} 5369 5370status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5371 effect_descriptor_t *descriptor) const 5372{ 5373 Mutex::Autolock _l(mLock); 5374 return EffectGetDescriptor(pUuid, descriptor); 5375} 5376 5377 5378sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5379 effect_descriptor_t *pDesc, 5380 const sp<IEffectClient>& effectClient, 5381 int32_t priority, 5382 audio_io_handle_t io, 5383 int sessionId, 5384 status_t *status, 5385 int *id, 5386 int *enabled) 5387{ 5388 status_t lStatus = NO_ERROR; 5389 sp<EffectHandle> handle; 5390 effect_descriptor_t desc; 5391 5392 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5393 pid, effectClient.get(), priority, sessionId, io); 5394 5395 if (pDesc == NULL) { 5396 lStatus = BAD_VALUE; 5397 goto Exit; 5398 } 5399 5400 // check audio settings permission for global effects 5401 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5402 lStatus = PERMISSION_DENIED; 5403 goto Exit; 5404 } 5405 5406 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5407 // that can only be created by audio policy manager (running in same process) 5408 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5409 lStatus = PERMISSION_DENIED; 5410 goto Exit; 5411 } 5412 5413 if (io == 0) { 5414 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5415 // output must be specified by AudioPolicyManager when using session 5416 // AUDIO_SESSION_OUTPUT_STAGE 5417 lStatus = BAD_VALUE; 5418 goto Exit; 5419 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5420 // if the output returned by getOutputForEffect() is removed before we lock the 5421 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5422 // and we will exit safely 5423 io = AudioSystem::getOutputForEffect(&desc); 5424 } 5425 } 5426 5427 { 5428 Mutex::Autolock _l(mLock); 5429 5430 5431 if (!EffectIsNullUuid(&pDesc->uuid)) { 5432 // if uuid is specified, request effect descriptor 5433 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5434 if (lStatus < 0) { 5435 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5436 goto Exit; 5437 } 5438 } else { 5439 // if uuid is not specified, look for an available implementation 5440 // of the required type in effect factory 5441 if (EffectIsNullUuid(&pDesc->type)) { 5442 ALOGW("createEffect() no effect type"); 5443 lStatus = BAD_VALUE; 5444 goto Exit; 5445 } 5446 uint32_t numEffects = 0; 5447 effect_descriptor_t d; 5448 d.flags = 0; // prevent compiler warning 5449 bool found = false; 5450 5451 lStatus = EffectQueryNumberEffects(&numEffects); 5452 if (lStatus < 0) { 5453 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5454 goto Exit; 5455 } 5456 for (uint32_t i = 0; i < numEffects; i++) { 5457 lStatus = EffectQueryEffect(i, &desc); 5458 if (lStatus < 0) { 5459 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5460 continue; 5461 } 5462 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5463 // If matching type found save effect descriptor. If the session is 5464 // 0 and the effect is not auxiliary, continue enumeration in case 5465 // an auxiliary version of this effect type is available 5466 found = true; 5467 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5468 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5469 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5470 break; 5471 } 5472 } 5473 } 5474 if (!found) { 5475 lStatus = BAD_VALUE; 5476 ALOGW("createEffect() effect not found"); 5477 goto Exit; 5478 } 5479 // For same effect type, chose auxiliary version over insert version if 5480 // connect to output mix (Compliance to OpenSL ES) 5481 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5482 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5483 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5484 } 5485 } 5486 5487 // Do not allow auxiliary effects on a session different from 0 (output mix) 5488 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5489 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5490 lStatus = INVALID_OPERATION; 5491 goto Exit; 5492 } 5493 5494 // check recording permission for visualizer 5495 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5496 !recordingAllowed()) { 5497 lStatus = PERMISSION_DENIED; 5498 goto Exit; 5499 } 5500 5501 // return effect descriptor 5502 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5503 5504 // If output is not specified try to find a matching audio session ID in one of the 5505 // output threads. 5506 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5507 // because of code checking output when entering the function. 5508 // Note: io is never 0 when creating an effect on an input 5509 if (io == 0) { 5510 // look for the thread where the specified audio session is present 5511 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5512 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5513 io = mPlaybackThreads.keyAt(i); 5514 break; 5515 } 5516 } 5517 if (io == 0) { 5518 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5519 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5520 io = mRecordThreads.keyAt(i); 5521 break; 5522 } 5523 } 5524 } 5525 // If no output thread contains the requested session ID, default to 5526 // first output. The effect chain will be moved to the correct output 5527 // thread when a track with the same session ID is created 5528 if (io == 0 && mPlaybackThreads.size()) { 5529 io = mPlaybackThreads.keyAt(0); 5530 } 5531 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5532 } 5533 ThreadBase *thread = checkRecordThread_l(io); 5534 if (thread == NULL) { 5535 thread = checkPlaybackThread_l(io); 5536 if (thread == NULL) { 5537 ALOGE("createEffect() unknown output thread"); 5538 lStatus = BAD_VALUE; 5539 goto Exit; 5540 } 5541 } 5542 5543 sp<Client> client = registerPid_l(pid); 5544 5545 // create effect on selected output thread 5546 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5547 &desc, enabled, &lStatus); 5548 if (handle != 0 && id != NULL) { 5549 *id = handle->id(); 5550 } 5551 } 5552 5553Exit: 5554 if(status) { 5555 *status = lStatus; 5556 } 5557 return handle; 5558} 5559 5560status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 5561 audio_io_handle_t dstOutput) 5562{ 5563 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5564 sessionId, srcOutput, dstOutput); 5565 Mutex::Autolock _l(mLock); 5566 if (srcOutput == dstOutput) { 5567 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5568 return NO_ERROR; 5569 } 5570 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5571 if (srcThread == NULL) { 5572 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5573 return BAD_VALUE; 5574 } 5575 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5576 if (dstThread == NULL) { 5577 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5578 return BAD_VALUE; 5579 } 5580 5581 Mutex::Autolock _dl(dstThread->mLock); 5582 Mutex::Autolock _sl(srcThread->mLock); 5583 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5584 5585 return NO_ERROR; 5586} 5587 5588// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5589status_t AudioFlinger::moveEffectChain_l(int sessionId, 5590 AudioFlinger::PlaybackThread *srcThread, 5591 AudioFlinger::PlaybackThread *dstThread, 5592 bool reRegister) 5593{ 5594 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5595 sessionId, srcThread, dstThread); 5596 5597 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5598 if (chain == 0) { 5599 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5600 sessionId, srcThread); 5601 return INVALID_OPERATION; 5602 } 5603 5604 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5605 // so that a new chain is created with correct parameters when first effect is added. This is 5606 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5607 // removed. 5608 srcThread->removeEffectChain_l(chain); 5609 5610 // transfer all effects one by one so that new effect chain is created on new thread with 5611 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5612 audio_io_handle_t dstOutput = dstThread->id(); 5613 sp<EffectChain> dstChain; 5614 uint32_t strategy = 0; // prevent compiler warning 5615 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5616 while (effect != 0) { 5617 srcThread->removeEffect_l(effect); 5618 dstThread->addEffect_l(effect); 5619 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5620 if (effect->state() == EffectModule::ACTIVE || 5621 effect->state() == EffectModule::STOPPING) { 5622 effect->start(); 5623 } 5624 // if the move request is not received from audio policy manager, the effect must be 5625 // re-registered with the new strategy and output 5626 if (dstChain == 0) { 5627 dstChain = effect->chain().promote(); 5628 if (dstChain == 0) { 5629 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5630 srcThread->addEffect_l(effect); 5631 return NO_INIT; 5632 } 5633 strategy = dstChain->strategy(); 5634 } 5635 if (reRegister) { 5636 AudioSystem::unregisterEffect(effect->id()); 5637 AudioSystem::registerEffect(&effect->desc(), 5638 dstOutput, 5639 strategy, 5640 sessionId, 5641 effect->id()); 5642 } 5643 effect = chain->getEffectFromId_l(0); 5644 } 5645 5646 return NO_ERROR; 5647} 5648 5649 5650// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5651sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5652 const sp<AudioFlinger::Client>& client, 5653 const sp<IEffectClient>& effectClient, 5654 int32_t priority, 5655 int sessionId, 5656 effect_descriptor_t *desc, 5657 int *enabled, 5658 status_t *status 5659 ) 5660{ 5661 sp<EffectModule> effect; 5662 sp<EffectHandle> handle; 5663 status_t lStatus; 5664 sp<EffectChain> chain; 5665 bool chainCreated = false; 5666 bool effectCreated = false; 5667 bool effectRegistered = false; 5668 5669 lStatus = initCheck(); 5670 if (lStatus != NO_ERROR) { 5671 ALOGW("createEffect_l() Audio driver not initialized."); 5672 goto Exit; 5673 } 5674 5675 // Do not allow effects with session ID 0 on direct output or duplicating threads 5676 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5677 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5678 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5679 desc->name, sessionId); 5680 lStatus = BAD_VALUE; 5681 goto Exit; 5682 } 5683 // Only Pre processor effects are allowed on input threads and only on input threads 5684 if ((mType == RECORD && 5685 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5686 (mType != RECORD && 5687 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5688 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5689 desc->name, desc->flags, mType); 5690 lStatus = BAD_VALUE; 5691 goto Exit; 5692 } 5693 5694 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5695 5696 { // scope for mLock 5697 Mutex::Autolock _l(mLock); 5698 5699 // check for existing effect chain with the requested audio session 5700 chain = getEffectChain_l(sessionId); 5701 if (chain == 0) { 5702 // create a new chain for this session 5703 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5704 chain = new EffectChain(this, sessionId); 5705 addEffectChain_l(chain); 5706 chain->setStrategy(getStrategyForSession_l(sessionId)); 5707 chainCreated = true; 5708 } else { 5709 effect = chain->getEffectFromDesc_l(desc); 5710 } 5711 5712 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5713 5714 if (effect == 0) { 5715 int id = mAudioFlinger->nextUniqueId(); 5716 // Check CPU and memory usage 5717 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5718 if (lStatus != NO_ERROR) { 5719 goto Exit; 5720 } 5721 effectRegistered = true; 5722 // create a new effect module if none present in the chain 5723 effect = new EffectModule(this, chain, desc, id, sessionId); 5724 lStatus = effect->status(); 5725 if (lStatus != NO_ERROR) { 5726 goto Exit; 5727 } 5728 lStatus = chain->addEffect_l(effect); 5729 if (lStatus != NO_ERROR) { 5730 goto Exit; 5731 } 5732 effectCreated = true; 5733 5734 effect->setDevice(mDevice); 5735 effect->setMode(mAudioFlinger->getMode()); 5736 } 5737 // create effect handle and connect it to effect module 5738 handle = new EffectHandle(effect, client, effectClient, priority); 5739 lStatus = effect->addHandle(handle); 5740 if (enabled != NULL) { 5741 *enabled = (int)effect->isEnabled(); 5742 } 5743 } 5744 5745Exit: 5746 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5747 Mutex::Autolock _l(mLock); 5748 if (effectCreated) { 5749 chain->removeEffect_l(effect); 5750 } 5751 if (effectRegistered) { 5752 AudioSystem::unregisterEffect(effect->id()); 5753 } 5754 if (chainCreated) { 5755 removeEffectChain_l(chain); 5756 } 5757 handle.clear(); 5758 } 5759 5760 if(status) { 5761 *status = lStatus; 5762 } 5763 return handle; 5764} 5765 5766sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5767{ 5768 sp<EffectChain> chain = getEffectChain_l(sessionId); 5769 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 5770} 5771 5772// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5773// PlaybackThread::mLock held 5774status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5775{ 5776 // check for existing effect chain with the requested audio session 5777 int sessionId = effect->sessionId(); 5778 sp<EffectChain> chain = getEffectChain_l(sessionId); 5779 bool chainCreated = false; 5780 5781 if (chain == 0) { 5782 // create a new chain for this session 5783 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5784 chain = new EffectChain(this, sessionId); 5785 addEffectChain_l(chain); 5786 chain->setStrategy(getStrategyForSession_l(sessionId)); 5787 chainCreated = true; 5788 } 5789 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5790 5791 if (chain->getEffectFromId_l(effect->id()) != 0) { 5792 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5793 this, effect->desc().name, chain.get()); 5794 return BAD_VALUE; 5795 } 5796 5797 status_t status = chain->addEffect_l(effect); 5798 if (status != NO_ERROR) { 5799 if (chainCreated) { 5800 removeEffectChain_l(chain); 5801 } 5802 return status; 5803 } 5804 5805 effect->setDevice(mDevice); 5806 effect->setMode(mAudioFlinger->getMode()); 5807 return NO_ERROR; 5808} 5809 5810void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5811 5812 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5813 effect_descriptor_t desc = effect->desc(); 5814 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5815 detachAuxEffect_l(effect->id()); 5816 } 5817 5818 sp<EffectChain> chain = effect->chain().promote(); 5819 if (chain != 0) { 5820 // remove effect chain if removing last effect 5821 if (chain->removeEffect_l(effect) == 0) { 5822 removeEffectChain_l(chain); 5823 } 5824 } else { 5825 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5826 } 5827} 5828 5829void AudioFlinger::ThreadBase::lockEffectChains_l( 5830 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5831{ 5832 effectChains = mEffectChains; 5833 for (size_t i = 0; i < mEffectChains.size(); i++) { 5834 mEffectChains[i]->lock(); 5835 } 5836} 5837 5838void AudioFlinger::ThreadBase::unlockEffectChains( 5839 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5840{ 5841 for (size_t i = 0; i < effectChains.size(); i++) { 5842 effectChains[i]->unlock(); 5843 } 5844} 5845 5846sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5847{ 5848 Mutex::Autolock _l(mLock); 5849 return getEffectChain_l(sessionId); 5850} 5851 5852sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5853{ 5854 size_t size = mEffectChains.size(); 5855 for (size_t i = 0; i < size; i++) { 5856 if (mEffectChains[i]->sessionId() == sessionId) { 5857 return mEffectChains[i]; 5858 } 5859 } 5860 return 0; 5861} 5862 5863void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5864{ 5865 Mutex::Autolock _l(mLock); 5866 size_t size = mEffectChains.size(); 5867 for (size_t i = 0; i < size; i++) { 5868 mEffectChains[i]->setMode_l(mode); 5869 } 5870} 5871 5872void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5873 const wp<EffectHandle>& handle, 5874 bool unpinIfLast) { 5875 5876 Mutex::Autolock _l(mLock); 5877 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5878 // delete the effect module if removing last handle on it 5879 if (effect->removeHandle(handle) == 0) { 5880 if (!effect->isPinned() || unpinIfLast) { 5881 removeEffect_l(effect); 5882 AudioSystem::unregisterEffect(effect->id()); 5883 } 5884 } 5885} 5886 5887status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5888{ 5889 int session = chain->sessionId(); 5890 int16_t *buffer = mMixBuffer; 5891 bool ownsBuffer = false; 5892 5893 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5894 if (session > 0) { 5895 // Only one effect chain can be present in direct output thread and it uses 5896 // the mix buffer as input 5897 if (mType != DIRECT) { 5898 size_t numSamples = mFrameCount * mChannelCount; 5899 buffer = new int16_t[numSamples]; 5900 memset(buffer, 0, numSamples * sizeof(int16_t)); 5901 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5902 ownsBuffer = true; 5903 } 5904 5905 // Attach all tracks with same session ID to this chain. 5906 for (size_t i = 0; i < mTracks.size(); ++i) { 5907 sp<Track> track = mTracks[i]; 5908 if (session == track->sessionId()) { 5909 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5910 track->setMainBuffer(buffer); 5911 chain->incTrackCnt(); 5912 } 5913 } 5914 5915 // indicate all active tracks in the chain 5916 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5917 sp<Track> track = mActiveTracks[i].promote(); 5918 if (track == 0) continue; 5919 if (session == track->sessionId()) { 5920 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5921 chain->incActiveTrackCnt(); 5922 } 5923 } 5924 } 5925 5926 chain->setInBuffer(buffer, ownsBuffer); 5927 chain->setOutBuffer(mMixBuffer); 5928 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5929 // chains list in order to be processed last as it contains output stage effects 5930 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5931 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5932 // after track specific effects and before output stage 5933 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5934 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5935 // Effect chain for other sessions are inserted at beginning of effect 5936 // chains list to be processed before output mix effects. Relative order between other 5937 // sessions is not important 5938 size_t size = mEffectChains.size(); 5939 size_t i = 0; 5940 for (i = 0; i < size; i++) { 5941 if (mEffectChains[i]->sessionId() < session) break; 5942 } 5943 mEffectChains.insertAt(chain, i); 5944 checkSuspendOnAddEffectChain_l(chain); 5945 5946 return NO_ERROR; 5947} 5948 5949size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5950{ 5951 int session = chain->sessionId(); 5952 5953 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5954 5955 for (size_t i = 0; i < mEffectChains.size(); i++) { 5956 if (chain == mEffectChains[i]) { 5957 mEffectChains.removeAt(i); 5958 // detach all active tracks from the chain 5959 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5960 sp<Track> track = mActiveTracks[i].promote(); 5961 if (track == 0) continue; 5962 if (session == track->sessionId()) { 5963 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5964 chain.get(), session); 5965 chain->decActiveTrackCnt(); 5966 } 5967 } 5968 5969 // detach all tracks with same session ID from this chain 5970 for (size_t i = 0; i < mTracks.size(); ++i) { 5971 sp<Track> track = mTracks[i]; 5972 if (session == track->sessionId()) { 5973 track->setMainBuffer(mMixBuffer); 5974 chain->decTrackCnt(); 5975 } 5976 } 5977 break; 5978 } 5979 } 5980 return mEffectChains.size(); 5981} 5982 5983status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5984 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5985{ 5986 Mutex::Autolock _l(mLock); 5987 return attachAuxEffect_l(track, EffectId); 5988} 5989 5990status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5991 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5992{ 5993 status_t status = NO_ERROR; 5994 5995 if (EffectId == 0) { 5996 track->setAuxBuffer(0, NULL); 5997 } else { 5998 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 5999 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6000 if (effect != 0) { 6001 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6002 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6003 } else { 6004 status = INVALID_OPERATION; 6005 } 6006 } else { 6007 status = BAD_VALUE; 6008 } 6009 } 6010 return status; 6011} 6012 6013void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6014{ 6015 for (size_t i = 0; i < mTracks.size(); ++i) { 6016 sp<Track> track = mTracks[i]; 6017 if (track->auxEffectId() == effectId) { 6018 attachAuxEffect_l(track, 0); 6019 } 6020 } 6021} 6022 6023status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6024{ 6025 // only one chain per input thread 6026 if (mEffectChains.size() != 0) { 6027 return INVALID_OPERATION; 6028 } 6029 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6030 6031 chain->setInBuffer(NULL); 6032 chain->setOutBuffer(NULL); 6033 6034 checkSuspendOnAddEffectChain_l(chain); 6035 6036 mEffectChains.add(chain); 6037 6038 return NO_ERROR; 6039} 6040 6041size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6042{ 6043 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6044 ALOGW_IF(mEffectChains.size() != 1, 6045 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6046 chain.get(), mEffectChains.size(), this); 6047 if (mEffectChains.size() == 1) { 6048 mEffectChains.removeAt(0); 6049 } 6050 return 0; 6051} 6052 6053// ---------------------------------------------------------------------------- 6054// EffectModule implementation 6055// ---------------------------------------------------------------------------- 6056 6057#undef LOG_TAG 6058#define LOG_TAG "AudioFlinger::EffectModule" 6059 6060AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6061 const wp<AudioFlinger::EffectChain>& chain, 6062 effect_descriptor_t *desc, 6063 int id, 6064 int sessionId) 6065 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6066 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6067{ 6068 ALOGV("Constructor %p", this); 6069 int lStatus; 6070 sp<ThreadBase> thread = mThread.promote(); 6071 if (thread == 0) { 6072 return; 6073 } 6074 6075 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6076 6077 // create effect engine from effect factory 6078 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6079 6080 if (mStatus != NO_ERROR) { 6081 return; 6082 } 6083 lStatus = init(); 6084 if (lStatus < 0) { 6085 mStatus = lStatus; 6086 goto Error; 6087 } 6088 6089 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6090 mPinned = true; 6091 } 6092 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6093 return; 6094Error: 6095 EffectRelease(mEffectInterface); 6096 mEffectInterface = NULL; 6097 ALOGV("Constructor Error %d", mStatus); 6098} 6099 6100AudioFlinger::EffectModule::~EffectModule() 6101{ 6102 ALOGV("Destructor %p", this); 6103 if (mEffectInterface != NULL) { 6104 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6105 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6106 sp<ThreadBase> thread = mThread.promote(); 6107 if (thread != 0) { 6108 audio_stream_t *stream = thread->stream(); 6109 if (stream != NULL) { 6110 stream->remove_audio_effect(stream, mEffectInterface); 6111 } 6112 } 6113 } 6114 // release effect engine 6115 EffectRelease(mEffectInterface); 6116 } 6117} 6118 6119status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6120{ 6121 status_t status; 6122 6123 Mutex::Autolock _l(mLock); 6124 // First handle in mHandles has highest priority and controls the effect module 6125 int priority = handle->priority(); 6126 size_t size = mHandles.size(); 6127 sp<EffectHandle> h; 6128 size_t i; 6129 for (i = 0; i < size; i++) { 6130 h = mHandles[i].promote(); 6131 if (h == 0) continue; 6132 if (h->priority() <= priority) break; 6133 } 6134 // if inserted in first place, move effect control from previous owner to this handle 6135 if (i == 0) { 6136 bool enabled = false; 6137 if (h != 0) { 6138 enabled = h->enabled(); 6139 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6140 } 6141 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6142 status = NO_ERROR; 6143 } else { 6144 status = ALREADY_EXISTS; 6145 } 6146 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6147 mHandles.insertAt(handle, i); 6148 return status; 6149} 6150 6151size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6152{ 6153 Mutex::Autolock _l(mLock); 6154 size_t size = mHandles.size(); 6155 size_t i; 6156 for (i = 0; i < size; i++) { 6157 if (mHandles[i] == handle) break; 6158 } 6159 if (i == size) { 6160 return size; 6161 } 6162 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6163 6164 bool enabled = false; 6165 EffectHandle *hdl = handle.unsafe_get(); 6166 if (hdl != NULL) { 6167 ALOGV("removeHandle() unsafe_get OK"); 6168 enabled = hdl->enabled(); 6169 } 6170 mHandles.removeAt(i); 6171 size = mHandles.size(); 6172 // if removed from first place, move effect control from this handle to next in line 6173 if (i == 0 && size != 0) { 6174 sp<EffectHandle> h = mHandles[0].promote(); 6175 if (h != 0) { 6176 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6177 } 6178 } 6179 6180 // Prevent calls to process() and other functions on effect interface from now on. 6181 // The effect engine will be released by the destructor when the last strong reference on 6182 // this object is released which can happen after next process is called. 6183 if (size == 0 && !mPinned) { 6184 mState = DESTROYED; 6185 } 6186 6187 return size; 6188} 6189 6190sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6191{ 6192 Mutex::Autolock _l(mLock); 6193 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6194} 6195 6196void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6197{ 6198 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6199 // keep a strong reference on this EffectModule to avoid calling the 6200 // destructor before we exit 6201 sp<EffectModule> keep(this); 6202 { 6203 sp<ThreadBase> thread = mThread.promote(); 6204 if (thread != 0) { 6205 thread->disconnectEffect(keep, handle, unpinIfLast); 6206 } 6207 } 6208} 6209 6210void AudioFlinger::EffectModule::updateState() { 6211 Mutex::Autolock _l(mLock); 6212 6213 switch (mState) { 6214 case RESTART: 6215 reset_l(); 6216 // FALL THROUGH 6217 6218 case STARTING: 6219 // clear auxiliary effect input buffer for next accumulation 6220 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6221 memset(mConfig.inputCfg.buffer.raw, 6222 0, 6223 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6224 } 6225 start_l(); 6226 mState = ACTIVE; 6227 break; 6228 case STOPPING: 6229 stop_l(); 6230 mDisableWaitCnt = mMaxDisableWaitCnt; 6231 mState = STOPPED; 6232 break; 6233 case STOPPED: 6234 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6235 // turn off sequence. 6236 if (--mDisableWaitCnt == 0) { 6237 reset_l(); 6238 mState = IDLE; 6239 } 6240 break; 6241 default: //IDLE , ACTIVE, DESTROYED 6242 break; 6243 } 6244} 6245 6246void AudioFlinger::EffectModule::process() 6247{ 6248 Mutex::Autolock _l(mLock); 6249 6250 if (mState == DESTROYED || mEffectInterface == NULL || 6251 mConfig.inputCfg.buffer.raw == NULL || 6252 mConfig.outputCfg.buffer.raw == NULL) { 6253 return; 6254 } 6255 6256 if (isProcessEnabled()) { 6257 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6258 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6259 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6260 mConfig.inputCfg.buffer.s32, 6261 mConfig.inputCfg.buffer.frameCount/2); 6262 } 6263 6264 // do the actual processing in the effect engine 6265 int ret = (*mEffectInterface)->process(mEffectInterface, 6266 &mConfig.inputCfg.buffer, 6267 &mConfig.outputCfg.buffer); 6268 6269 // force transition to IDLE state when engine is ready 6270 if (mState == STOPPED && ret == -ENODATA) { 6271 mDisableWaitCnt = 1; 6272 } 6273 6274 // clear auxiliary effect input buffer for next accumulation 6275 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6276 memset(mConfig.inputCfg.buffer.raw, 0, 6277 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6278 } 6279 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6280 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6281 // If an insert effect is idle and input buffer is different from output buffer, 6282 // accumulate input onto output 6283 sp<EffectChain> chain = mChain.promote(); 6284 if (chain != 0 && chain->activeTrackCnt() != 0) { 6285 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6286 int16_t *in = mConfig.inputCfg.buffer.s16; 6287 int16_t *out = mConfig.outputCfg.buffer.s16; 6288 for (size_t i = 0; i < frameCnt; i++) { 6289 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6290 } 6291 } 6292 } 6293} 6294 6295void AudioFlinger::EffectModule::reset_l() 6296{ 6297 if (mEffectInterface == NULL) { 6298 return; 6299 } 6300 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6301} 6302 6303status_t AudioFlinger::EffectModule::configure() 6304{ 6305 uint32_t channels; 6306 if (mEffectInterface == NULL) { 6307 return NO_INIT; 6308 } 6309 6310 sp<ThreadBase> thread = mThread.promote(); 6311 if (thread == 0) { 6312 return DEAD_OBJECT; 6313 } 6314 6315 // TODO: handle configuration of effects replacing track process 6316 if (thread->channelCount() == 1) { 6317 channels = AUDIO_CHANNEL_OUT_MONO; 6318 } else { 6319 channels = AUDIO_CHANNEL_OUT_STEREO; 6320 } 6321 6322 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6323 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6324 } else { 6325 mConfig.inputCfg.channels = channels; 6326 } 6327 mConfig.outputCfg.channels = channels; 6328 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6329 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6330 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6331 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6332 mConfig.inputCfg.bufferProvider.cookie = NULL; 6333 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6334 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6335 mConfig.outputCfg.bufferProvider.cookie = NULL; 6336 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6337 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6338 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6339 // Insert effect: 6340 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6341 // always overwrites output buffer: input buffer == output buffer 6342 // - in other sessions: 6343 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6344 // other effect: overwrites output buffer: input buffer == output buffer 6345 // Auxiliary effect: 6346 // accumulates in output buffer: input buffer != output buffer 6347 // Therefore: accumulate <=> input buffer != output buffer 6348 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6349 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6350 } else { 6351 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6352 } 6353 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6354 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6355 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6356 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6357 6358 ALOGV("configure() %p thread %p buffer %p framecount %d", 6359 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6360 6361 status_t cmdStatus; 6362 uint32_t size = sizeof(int); 6363 status_t status = (*mEffectInterface)->command(mEffectInterface, 6364 EFFECT_CMD_SET_CONFIG, 6365 sizeof(effect_config_t), 6366 &mConfig, 6367 &size, 6368 &cmdStatus); 6369 if (status == 0) { 6370 status = cmdStatus; 6371 } 6372 6373 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6374 (1000 * mConfig.outputCfg.buffer.frameCount); 6375 6376 return status; 6377} 6378 6379status_t AudioFlinger::EffectModule::init() 6380{ 6381 Mutex::Autolock _l(mLock); 6382 if (mEffectInterface == NULL) { 6383 return NO_INIT; 6384 } 6385 status_t cmdStatus; 6386 uint32_t size = sizeof(status_t); 6387 status_t status = (*mEffectInterface)->command(mEffectInterface, 6388 EFFECT_CMD_INIT, 6389 0, 6390 NULL, 6391 &size, 6392 &cmdStatus); 6393 if (status == 0) { 6394 status = cmdStatus; 6395 } 6396 return status; 6397} 6398 6399status_t AudioFlinger::EffectModule::start() 6400{ 6401 Mutex::Autolock _l(mLock); 6402 return start_l(); 6403} 6404 6405status_t AudioFlinger::EffectModule::start_l() 6406{ 6407 if (mEffectInterface == NULL) { 6408 return NO_INIT; 6409 } 6410 status_t cmdStatus; 6411 uint32_t size = sizeof(status_t); 6412 status_t status = (*mEffectInterface)->command(mEffectInterface, 6413 EFFECT_CMD_ENABLE, 6414 0, 6415 NULL, 6416 &size, 6417 &cmdStatus); 6418 if (status == 0) { 6419 status = cmdStatus; 6420 } 6421 if (status == 0 && 6422 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6423 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6424 sp<ThreadBase> thread = mThread.promote(); 6425 if (thread != 0) { 6426 audio_stream_t *stream = thread->stream(); 6427 if (stream != NULL) { 6428 stream->add_audio_effect(stream, mEffectInterface); 6429 } 6430 } 6431 } 6432 return status; 6433} 6434 6435status_t AudioFlinger::EffectModule::stop() 6436{ 6437 Mutex::Autolock _l(mLock); 6438 return stop_l(); 6439} 6440 6441status_t AudioFlinger::EffectModule::stop_l() 6442{ 6443 if (mEffectInterface == NULL) { 6444 return NO_INIT; 6445 } 6446 status_t cmdStatus; 6447 uint32_t size = sizeof(status_t); 6448 status_t status = (*mEffectInterface)->command(mEffectInterface, 6449 EFFECT_CMD_DISABLE, 6450 0, 6451 NULL, 6452 &size, 6453 &cmdStatus); 6454 if (status == 0) { 6455 status = cmdStatus; 6456 } 6457 if (status == 0 && 6458 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6459 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6460 sp<ThreadBase> thread = mThread.promote(); 6461 if (thread != 0) { 6462 audio_stream_t *stream = thread->stream(); 6463 if (stream != NULL) { 6464 stream->remove_audio_effect(stream, mEffectInterface); 6465 } 6466 } 6467 } 6468 return status; 6469} 6470 6471status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6472 uint32_t cmdSize, 6473 void *pCmdData, 6474 uint32_t *replySize, 6475 void *pReplyData) 6476{ 6477 Mutex::Autolock _l(mLock); 6478// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6479 6480 if (mState == DESTROYED || mEffectInterface == NULL) { 6481 return NO_INIT; 6482 } 6483 status_t status = (*mEffectInterface)->command(mEffectInterface, 6484 cmdCode, 6485 cmdSize, 6486 pCmdData, 6487 replySize, 6488 pReplyData); 6489 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6490 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6491 for (size_t i = 1; i < mHandles.size(); i++) { 6492 sp<EffectHandle> h = mHandles[i].promote(); 6493 if (h != 0) { 6494 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6495 } 6496 } 6497 } 6498 return status; 6499} 6500 6501status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6502{ 6503 6504 Mutex::Autolock _l(mLock); 6505 ALOGV("setEnabled %p enabled %d", this, enabled); 6506 6507 if (enabled != isEnabled()) { 6508 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6509 if (enabled && status != NO_ERROR) { 6510 return status; 6511 } 6512 6513 switch (mState) { 6514 // going from disabled to enabled 6515 case IDLE: 6516 mState = STARTING; 6517 break; 6518 case STOPPED: 6519 mState = RESTART; 6520 break; 6521 case STOPPING: 6522 mState = ACTIVE; 6523 break; 6524 6525 // going from enabled to disabled 6526 case RESTART: 6527 mState = STOPPED; 6528 break; 6529 case STARTING: 6530 mState = IDLE; 6531 break; 6532 case ACTIVE: 6533 mState = STOPPING; 6534 break; 6535 case DESTROYED: 6536 return NO_ERROR; // simply ignore as we are being destroyed 6537 } 6538 for (size_t i = 1; i < mHandles.size(); i++) { 6539 sp<EffectHandle> h = mHandles[i].promote(); 6540 if (h != 0) { 6541 h->setEnabled(enabled); 6542 } 6543 } 6544 } 6545 return NO_ERROR; 6546} 6547 6548bool AudioFlinger::EffectModule::isEnabled() const 6549{ 6550 switch (mState) { 6551 case RESTART: 6552 case STARTING: 6553 case ACTIVE: 6554 return true; 6555 case IDLE: 6556 case STOPPING: 6557 case STOPPED: 6558 case DESTROYED: 6559 default: 6560 return false; 6561 } 6562} 6563 6564bool AudioFlinger::EffectModule::isProcessEnabled() const 6565{ 6566 switch (mState) { 6567 case RESTART: 6568 case ACTIVE: 6569 case STOPPING: 6570 case STOPPED: 6571 return true; 6572 case IDLE: 6573 case STARTING: 6574 case DESTROYED: 6575 default: 6576 return false; 6577 } 6578} 6579 6580status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6581{ 6582 Mutex::Autolock _l(mLock); 6583 status_t status = NO_ERROR; 6584 6585 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6586 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6587 if (isProcessEnabled() && 6588 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6589 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6590 status_t cmdStatus; 6591 uint32_t volume[2]; 6592 uint32_t *pVolume = NULL; 6593 uint32_t size = sizeof(volume); 6594 volume[0] = *left; 6595 volume[1] = *right; 6596 if (controller) { 6597 pVolume = volume; 6598 } 6599 status = (*mEffectInterface)->command(mEffectInterface, 6600 EFFECT_CMD_SET_VOLUME, 6601 size, 6602 volume, 6603 &size, 6604 pVolume); 6605 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6606 *left = volume[0]; 6607 *right = volume[1]; 6608 } 6609 } 6610 return status; 6611} 6612 6613status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6614{ 6615 Mutex::Autolock _l(mLock); 6616 status_t status = NO_ERROR; 6617 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6618 // audio pre processing modules on RecordThread can receive both output and 6619 // input device indication in the same call 6620 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6621 if (dev) { 6622 status_t cmdStatus; 6623 uint32_t size = sizeof(status_t); 6624 6625 status = (*mEffectInterface)->command(mEffectInterface, 6626 EFFECT_CMD_SET_DEVICE, 6627 sizeof(uint32_t), 6628 &dev, 6629 &size, 6630 &cmdStatus); 6631 if (status == NO_ERROR) { 6632 status = cmdStatus; 6633 } 6634 } 6635 dev = device & AUDIO_DEVICE_IN_ALL; 6636 if (dev) { 6637 status_t cmdStatus; 6638 uint32_t size = sizeof(status_t); 6639 6640 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6641 EFFECT_CMD_SET_INPUT_DEVICE, 6642 sizeof(uint32_t), 6643 &dev, 6644 &size, 6645 &cmdStatus); 6646 if (status2 == NO_ERROR) { 6647 status2 = cmdStatus; 6648 } 6649 if (status == NO_ERROR) { 6650 status = status2; 6651 } 6652 } 6653 } 6654 return status; 6655} 6656 6657status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6658{ 6659 Mutex::Autolock _l(mLock); 6660 status_t status = NO_ERROR; 6661 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6662 status_t cmdStatus; 6663 uint32_t size = sizeof(status_t); 6664 status = (*mEffectInterface)->command(mEffectInterface, 6665 EFFECT_CMD_SET_AUDIO_MODE, 6666 sizeof(audio_mode_t), 6667 &mode, 6668 &size, 6669 &cmdStatus); 6670 if (status == NO_ERROR) { 6671 status = cmdStatus; 6672 } 6673 } 6674 return status; 6675} 6676 6677void AudioFlinger::EffectModule::setSuspended(bool suspended) 6678{ 6679 Mutex::Autolock _l(mLock); 6680 mSuspended = suspended; 6681} 6682 6683bool AudioFlinger::EffectModule::suspended() const 6684{ 6685 Mutex::Autolock _l(mLock); 6686 return mSuspended; 6687} 6688 6689status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6690{ 6691 const size_t SIZE = 256; 6692 char buffer[SIZE]; 6693 String8 result; 6694 6695 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6696 result.append(buffer); 6697 6698 bool locked = tryLock(mLock); 6699 // failed to lock - AudioFlinger is probably deadlocked 6700 if (!locked) { 6701 result.append("\t\tCould not lock Fx mutex:\n"); 6702 } 6703 6704 result.append("\t\tSession Status State Engine:\n"); 6705 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6706 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6707 result.append(buffer); 6708 6709 result.append("\t\tDescriptor:\n"); 6710 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6711 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6712 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6713 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6714 result.append(buffer); 6715 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6716 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6717 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6718 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6719 result.append(buffer); 6720 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6721 mDescriptor.apiVersion, 6722 mDescriptor.flags); 6723 result.append(buffer); 6724 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6725 mDescriptor.name); 6726 result.append(buffer); 6727 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6728 mDescriptor.implementor); 6729 result.append(buffer); 6730 6731 result.append("\t\t- Input configuration:\n"); 6732 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6733 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6734 (uint32_t)mConfig.inputCfg.buffer.raw, 6735 mConfig.inputCfg.buffer.frameCount, 6736 mConfig.inputCfg.samplingRate, 6737 mConfig.inputCfg.channels, 6738 mConfig.inputCfg.format); 6739 result.append(buffer); 6740 6741 result.append("\t\t- Output configuration:\n"); 6742 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6743 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6744 (uint32_t)mConfig.outputCfg.buffer.raw, 6745 mConfig.outputCfg.buffer.frameCount, 6746 mConfig.outputCfg.samplingRate, 6747 mConfig.outputCfg.channels, 6748 mConfig.outputCfg.format); 6749 result.append(buffer); 6750 6751 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6752 result.append(buffer); 6753 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6754 for (size_t i = 0; i < mHandles.size(); ++i) { 6755 sp<EffectHandle> handle = mHandles[i].promote(); 6756 if (handle != 0) { 6757 handle->dump(buffer, SIZE); 6758 result.append(buffer); 6759 } 6760 } 6761 6762 result.append("\n"); 6763 6764 write(fd, result.string(), result.length()); 6765 6766 if (locked) { 6767 mLock.unlock(); 6768 } 6769 6770 return NO_ERROR; 6771} 6772 6773// ---------------------------------------------------------------------------- 6774// EffectHandle implementation 6775// ---------------------------------------------------------------------------- 6776 6777#undef LOG_TAG 6778#define LOG_TAG "AudioFlinger::EffectHandle" 6779 6780AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6781 const sp<AudioFlinger::Client>& client, 6782 const sp<IEffectClient>& effectClient, 6783 int32_t priority) 6784 : BnEffect(), 6785 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6786 mPriority(priority), mHasControl(false), mEnabled(false) 6787{ 6788 ALOGV("constructor %p", this); 6789 6790 if (client == 0) { 6791 return; 6792 } 6793 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6794 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6795 if (mCblkMemory != 0) { 6796 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6797 6798 if (mCblk != NULL) { 6799 new(mCblk) effect_param_cblk_t(); 6800 mBuffer = (uint8_t *)mCblk + bufOffset; 6801 } 6802 } else { 6803 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6804 return; 6805 } 6806} 6807 6808AudioFlinger::EffectHandle::~EffectHandle() 6809{ 6810 ALOGV("Destructor %p", this); 6811 disconnect(false); 6812 ALOGV("Destructor DONE %p", this); 6813} 6814 6815status_t AudioFlinger::EffectHandle::enable() 6816{ 6817 ALOGV("enable %p", this); 6818 if (!mHasControl) return INVALID_OPERATION; 6819 if (mEffect == 0) return DEAD_OBJECT; 6820 6821 if (mEnabled) { 6822 return NO_ERROR; 6823 } 6824 6825 mEnabled = true; 6826 6827 sp<ThreadBase> thread = mEffect->thread().promote(); 6828 if (thread != 0) { 6829 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6830 } 6831 6832 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6833 if (mEffect->suspended()) { 6834 return NO_ERROR; 6835 } 6836 6837 status_t status = mEffect->setEnabled(true); 6838 if (status != NO_ERROR) { 6839 if (thread != 0) { 6840 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6841 } 6842 mEnabled = false; 6843 } 6844 return status; 6845} 6846 6847status_t AudioFlinger::EffectHandle::disable() 6848{ 6849 ALOGV("disable %p", this); 6850 if (!mHasControl) return INVALID_OPERATION; 6851 if (mEffect == 0) return DEAD_OBJECT; 6852 6853 if (!mEnabled) { 6854 return NO_ERROR; 6855 } 6856 mEnabled = false; 6857 6858 if (mEffect->suspended()) { 6859 return NO_ERROR; 6860 } 6861 6862 status_t status = mEffect->setEnabled(false); 6863 6864 sp<ThreadBase> thread = mEffect->thread().promote(); 6865 if (thread != 0) { 6866 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6867 } 6868 6869 return status; 6870} 6871 6872void AudioFlinger::EffectHandle::disconnect() 6873{ 6874 disconnect(true); 6875} 6876 6877void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 6878{ 6879 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 6880 if (mEffect == 0) { 6881 return; 6882 } 6883 mEffect->disconnect(this, unpinIfLast); 6884 6885 if (mHasControl && mEnabled) { 6886 sp<ThreadBase> thread = mEffect->thread().promote(); 6887 if (thread != 0) { 6888 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6889 } 6890 } 6891 6892 // release sp on module => module destructor can be called now 6893 mEffect.clear(); 6894 if (mClient != 0) { 6895 if (mCblk != NULL) { 6896 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 6897 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6898 } 6899 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 6900 // Client destructor must run with AudioFlinger mutex locked 6901 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6902 mClient.clear(); 6903 } 6904} 6905 6906status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6907 uint32_t cmdSize, 6908 void *pCmdData, 6909 uint32_t *replySize, 6910 void *pReplyData) 6911{ 6912// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6913// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6914 6915 // only get parameter command is permitted for applications not controlling the effect 6916 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6917 return INVALID_OPERATION; 6918 } 6919 if (mEffect == 0) return DEAD_OBJECT; 6920 if (mClient == 0) return INVALID_OPERATION; 6921 6922 // handle commands that are not forwarded transparently to effect engine 6923 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6924 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6925 // no risk to block the whole media server process or mixer threads is we are stuck here 6926 Mutex::Autolock _l(mCblk->lock); 6927 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6928 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6929 mCblk->serverIndex = 0; 6930 mCblk->clientIndex = 0; 6931 return BAD_VALUE; 6932 } 6933 status_t status = NO_ERROR; 6934 while (mCblk->serverIndex < mCblk->clientIndex) { 6935 int reply; 6936 uint32_t rsize = sizeof(int); 6937 int *p = (int *)(mBuffer + mCblk->serverIndex); 6938 int size = *p++; 6939 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6940 ALOGW("command(): invalid parameter block size"); 6941 break; 6942 } 6943 effect_param_t *param = (effect_param_t *)p; 6944 if (param->psize == 0 || param->vsize == 0) { 6945 ALOGW("command(): null parameter or value size"); 6946 mCblk->serverIndex += size; 6947 continue; 6948 } 6949 uint32_t psize = sizeof(effect_param_t) + 6950 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6951 param->vsize; 6952 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6953 psize, 6954 p, 6955 &rsize, 6956 &reply); 6957 // stop at first error encountered 6958 if (ret != NO_ERROR) { 6959 status = ret; 6960 *(int *)pReplyData = reply; 6961 break; 6962 } else if (reply != NO_ERROR) { 6963 *(int *)pReplyData = reply; 6964 break; 6965 } 6966 mCblk->serverIndex += size; 6967 } 6968 mCblk->serverIndex = 0; 6969 mCblk->clientIndex = 0; 6970 return status; 6971 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6972 *(int *)pReplyData = NO_ERROR; 6973 return enable(); 6974 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6975 *(int *)pReplyData = NO_ERROR; 6976 return disable(); 6977 } 6978 6979 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6980} 6981 6982void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 6983{ 6984 ALOGV("setControl %p control %d", this, hasControl); 6985 6986 mHasControl = hasControl; 6987 mEnabled = enabled; 6988 6989 if (signal && mEffectClient != 0) { 6990 mEffectClient->controlStatusChanged(hasControl); 6991 } 6992} 6993 6994void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6995 uint32_t cmdSize, 6996 void *pCmdData, 6997 uint32_t replySize, 6998 void *pReplyData) 6999{ 7000 if (mEffectClient != 0) { 7001 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7002 } 7003} 7004 7005 7006 7007void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7008{ 7009 if (mEffectClient != 0) { 7010 mEffectClient->enableStatusChanged(enabled); 7011 } 7012} 7013 7014status_t AudioFlinger::EffectHandle::onTransact( 7015 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7016{ 7017 return BnEffect::onTransact(code, data, reply, flags); 7018} 7019 7020 7021void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7022{ 7023 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7024 7025 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7026 (mClient == 0) ? getpid() : mClient->pid(), 7027 mPriority, 7028 mHasControl, 7029 !locked, 7030 mCblk ? mCblk->clientIndex : 0, 7031 mCblk ? mCblk->serverIndex : 0 7032 ); 7033 7034 if (locked) { 7035 mCblk->lock.unlock(); 7036 } 7037} 7038 7039#undef LOG_TAG 7040#define LOG_TAG "AudioFlinger::EffectChain" 7041 7042AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7043 int sessionId) 7044 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7045 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7046 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7047{ 7048 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7049 sp<ThreadBase> thread = mThread.promote(); 7050 if (thread == 0) { 7051 return; 7052 } 7053 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7054 thread->frameCount(); 7055} 7056 7057AudioFlinger::EffectChain::~EffectChain() 7058{ 7059 if (mOwnInBuffer) { 7060 delete mInBuffer; 7061 } 7062 7063} 7064 7065// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7066sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7067{ 7068 size_t size = mEffects.size(); 7069 7070 for (size_t i = 0; i < size; i++) { 7071 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7072 return mEffects[i]; 7073 } 7074 } 7075 return 0; 7076} 7077 7078// getEffectFromId_l() must be called with ThreadBase::mLock held 7079sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7080{ 7081 size_t size = mEffects.size(); 7082 7083 for (size_t i = 0; i < size; i++) { 7084 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7085 if (id == 0 || mEffects[i]->id() == id) { 7086 return mEffects[i]; 7087 } 7088 } 7089 return 0; 7090} 7091 7092// getEffectFromType_l() must be called with ThreadBase::mLock held 7093sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7094 const effect_uuid_t *type) 7095{ 7096 size_t size = mEffects.size(); 7097 7098 for (size_t i = 0; i < size; i++) { 7099 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7100 return mEffects[i]; 7101 } 7102 } 7103 return 0; 7104} 7105 7106// Must be called with EffectChain::mLock locked 7107void AudioFlinger::EffectChain::process_l() 7108{ 7109 sp<ThreadBase> thread = mThread.promote(); 7110 if (thread == 0) { 7111 ALOGW("process_l(): cannot promote mixer thread"); 7112 return; 7113 } 7114 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7115 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7116 // always process effects unless no more tracks are on the session and the effect tail 7117 // has been rendered 7118 bool doProcess = true; 7119 if (!isGlobalSession) { 7120 bool tracksOnSession = (trackCnt() != 0); 7121 7122 if (!tracksOnSession && mTailBufferCount == 0) { 7123 doProcess = false; 7124 } 7125 7126 if (activeTrackCnt() == 0) { 7127 // if no track is active and the effect tail has not been rendered, 7128 // the input buffer must be cleared here as the mixer process will not do it 7129 if (tracksOnSession || mTailBufferCount > 0) { 7130 size_t numSamples = thread->frameCount() * thread->channelCount(); 7131 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7132 if (mTailBufferCount > 0) { 7133 mTailBufferCount--; 7134 } 7135 } 7136 } 7137 } 7138 7139 size_t size = mEffects.size(); 7140 if (doProcess) { 7141 for (size_t i = 0; i < size; i++) { 7142 mEffects[i]->process(); 7143 } 7144 } 7145 for (size_t i = 0; i < size; i++) { 7146 mEffects[i]->updateState(); 7147 } 7148} 7149 7150// addEffect_l() must be called with PlaybackThread::mLock held 7151status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7152{ 7153 effect_descriptor_t desc = effect->desc(); 7154 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7155 7156 Mutex::Autolock _l(mLock); 7157 effect->setChain(this); 7158 sp<ThreadBase> thread = mThread.promote(); 7159 if (thread == 0) { 7160 return NO_INIT; 7161 } 7162 effect->setThread(thread); 7163 7164 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7165 // Auxiliary effects are inserted at the beginning of mEffects vector as 7166 // they are processed first and accumulated in chain input buffer 7167 mEffects.insertAt(effect, 0); 7168 7169 // the input buffer for auxiliary effect contains mono samples in 7170 // 32 bit format. This is to avoid saturation in AudoMixer 7171 // accumulation stage. Saturation is done in EffectModule::process() before 7172 // calling the process in effect engine 7173 size_t numSamples = thread->frameCount(); 7174 int32_t *buffer = new int32_t[numSamples]; 7175 memset(buffer, 0, numSamples * sizeof(int32_t)); 7176 effect->setInBuffer((int16_t *)buffer); 7177 // auxiliary effects output samples to chain input buffer for further processing 7178 // by insert effects 7179 effect->setOutBuffer(mInBuffer); 7180 } else { 7181 // Insert effects are inserted at the end of mEffects vector as they are processed 7182 // after track and auxiliary effects. 7183 // Insert effect order as a function of indicated preference: 7184 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7185 // another effect is present 7186 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7187 // last effect claiming first position 7188 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7189 // first effect claiming last position 7190 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7191 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7192 // already present 7193 7194 int size = (int)mEffects.size(); 7195 int idx_insert = size; 7196 int idx_insert_first = -1; 7197 int idx_insert_last = -1; 7198 7199 for (int i = 0; i < size; i++) { 7200 effect_descriptor_t d = mEffects[i]->desc(); 7201 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7202 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7203 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7204 // check invalid effect chaining combinations 7205 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7206 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7207 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7208 return INVALID_OPERATION; 7209 } 7210 // remember position of first insert effect and by default 7211 // select this as insert position for new effect 7212 if (idx_insert == size) { 7213 idx_insert = i; 7214 } 7215 // remember position of last insert effect claiming 7216 // first position 7217 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7218 idx_insert_first = i; 7219 } 7220 // remember position of first insert effect claiming 7221 // last position 7222 if (iPref == EFFECT_FLAG_INSERT_LAST && 7223 idx_insert_last == -1) { 7224 idx_insert_last = i; 7225 } 7226 } 7227 } 7228 7229 // modify idx_insert from first position if needed 7230 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7231 if (idx_insert_last != -1) { 7232 idx_insert = idx_insert_last; 7233 } else { 7234 idx_insert = size; 7235 } 7236 } else { 7237 if (idx_insert_first != -1) { 7238 idx_insert = idx_insert_first + 1; 7239 } 7240 } 7241 7242 // always read samples from chain input buffer 7243 effect->setInBuffer(mInBuffer); 7244 7245 // if last effect in the chain, output samples to chain 7246 // output buffer, otherwise to chain input buffer 7247 if (idx_insert == size) { 7248 if (idx_insert != 0) { 7249 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7250 mEffects[idx_insert-1]->configure(); 7251 } 7252 effect->setOutBuffer(mOutBuffer); 7253 } else { 7254 effect->setOutBuffer(mInBuffer); 7255 } 7256 mEffects.insertAt(effect, idx_insert); 7257 7258 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7259 } 7260 effect->configure(); 7261 return NO_ERROR; 7262} 7263 7264// removeEffect_l() must be called with PlaybackThread::mLock held 7265size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7266{ 7267 Mutex::Autolock _l(mLock); 7268 int size = (int)mEffects.size(); 7269 int i; 7270 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7271 7272 for (i = 0; i < size; i++) { 7273 if (effect == mEffects[i]) { 7274 // calling stop here will remove pre-processing effect from the audio HAL. 7275 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7276 // the middle of a read from audio HAL 7277 if (mEffects[i]->state() == EffectModule::ACTIVE || 7278 mEffects[i]->state() == EffectModule::STOPPING) { 7279 mEffects[i]->stop(); 7280 } 7281 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7282 delete[] effect->inBuffer(); 7283 } else { 7284 if (i == size - 1 && i != 0) { 7285 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7286 mEffects[i - 1]->configure(); 7287 } 7288 } 7289 mEffects.removeAt(i); 7290 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7291 break; 7292 } 7293 } 7294 7295 return mEffects.size(); 7296} 7297 7298// setDevice_l() must be called with PlaybackThread::mLock held 7299void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7300{ 7301 size_t size = mEffects.size(); 7302 for (size_t i = 0; i < size; i++) { 7303 mEffects[i]->setDevice(device); 7304 } 7305} 7306 7307// setMode_l() must be called with PlaybackThread::mLock held 7308void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7309{ 7310 size_t size = mEffects.size(); 7311 for (size_t i = 0; i < size; i++) { 7312 mEffects[i]->setMode(mode); 7313 } 7314} 7315 7316// setVolume_l() must be called with PlaybackThread::mLock held 7317bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7318{ 7319 uint32_t newLeft = *left; 7320 uint32_t newRight = *right; 7321 bool hasControl = false; 7322 int ctrlIdx = -1; 7323 size_t size = mEffects.size(); 7324 7325 // first update volume controller 7326 for (size_t i = size; i > 0; i--) { 7327 if (mEffects[i - 1]->isProcessEnabled() && 7328 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7329 ctrlIdx = i - 1; 7330 hasControl = true; 7331 break; 7332 } 7333 } 7334 7335 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7336 if (hasControl) { 7337 *left = mNewLeftVolume; 7338 *right = mNewRightVolume; 7339 } 7340 return hasControl; 7341 } 7342 7343 mVolumeCtrlIdx = ctrlIdx; 7344 mLeftVolume = newLeft; 7345 mRightVolume = newRight; 7346 7347 // second get volume update from volume controller 7348 if (ctrlIdx >= 0) { 7349 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7350 mNewLeftVolume = newLeft; 7351 mNewRightVolume = newRight; 7352 } 7353 // then indicate volume to all other effects in chain. 7354 // Pass altered volume to effects before volume controller 7355 // and requested volume to effects after controller 7356 uint32_t lVol = newLeft; 7357 uint32_t rVol = newRight; 7358 7359 for (size_t i = 0; i < size; i++) { 7360 if ((int)i == ctrlIdx) continue; 7361 // this also works for ctrlIdx == -1 when there is no volume controller 7362 if ((int)i > ctrlIdx) { 7363 lVol = *left; 7364 rVol = *right; 7365 } 7366 mEffects[i]->setVolume(&lVol, &rVol, false); 7367 } 7368 *left = newLeft; 7369 *right = newRight; 7370 7371 return hasControl; 7372} 7373 7374status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7375{ 7376 const size_t SIZE = 256; 7377 char buffer[SIZE]; 7378 String8 result; 7379 7380 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7381 result.append(buffer); 7382 7383 bool locked = tryLock(mLock); 7384 // failed to lock - AudioFlinger is probably deadlocked 7385 if (!locked) { 7386 result.append("\tCould not lock mutex:\n"); 7387 } 7388 7389 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7390 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7391 mEffects.size(), 7392 (uint32_t)mInBuffer, 7393 (uint32_t)mOutBuffer, 7394 mActiveTrackCnt); 7395 result.append(buffer); 7396 write(fd, result.string(), result.size()); 7397 7398 for (size_t i = 0; i < mEffects.size(); ++i) { 7399 sp<EffectModule> effect = mEffects[i]; 7400 if (effect != 0) { 7401 effect->dump(fd, args); 7402 } 7403 } 7404 7405 if (locked) { 7406 mLock.unlock(); 7407 } 7408 7409 return NO_ERROR; 7410} 7411 7412// must be called with ThreadBase::mLock held 7413void AudioFlinger::EffectChain::setEffectSuspended_l( 7414 const effect_uuid_t *type, bool suspend) 7415{ 7416 sp<SuspendedEffectDesc> desc; 7417 // use effect type UUID timelow as key as there is no real risk of identical 7418 // timeLow fields among effect type UUIDs. 7419 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7420 if (suspend) { 7421 if (index >= 0) { 7422 desc = mSuspendedEffects.valueAt(index); 7423 } else { 7424 desc = new SuspendedEffectDesc(); 7425 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7426 mSuspendedEffects.add(type->timeLow, desc); 7427 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7428 } 7429 if (desc->mRefCount++ == 0) { 7430 sp<EffectModule> effect = getEffectIfEnabled(type); 7431 if (effect != 0) { 7432 desc->mEffect = effect; 7433 effect->setSuspended(true); 7434 effect->setEnabled(false); 7435 } 7436 } 7437 } else { 7438 if (index < 0) { 7439 return; 7440 } 7441 desc = mSuspendedEffects.valueAt(index); 7442 if (desc->mRefCount <= 0) { 7443 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7444 desc->mRefCount = 1; 7445 } 7446 if (--desc->mRefCount == 0) { 7447 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7448 if (desc->mEffect != 0) { 7449 sp<EffectModule> effect = desc->mEffect.promote(); 7450 if (effect != 0) { 7451 effect->setSuspended(false); 7452 sp<EffectHandle> handle = effect->controlHandle(); 7453 if (handle != 0) { 7454 effect->setEnabled(handle->enabled()); 7455 } 7456 } 7457 desc->mEffect.clear(); 7458 } 7459 mSuspendedEffects.removeItemsAt(index); 7460 } 7461 } 7462} 7463 7464// must be called with ThreadBase::mLock held 7465void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7466{ 7467 sp<SuspendedEffectDesc> desc; 7468 7469 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7470 if (suspend) { 7471 if (index >= 0) { 7472 desc = mSuspendedEffects.valueAt(index); 7473 } else { 7474 desc = new SuspendedEffectDesc(); 7475 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7476 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7477 } 7478 if (desc->mRefCount++ == 0) { 7479 Vector< sp<EffectModule> > effects; 7480 getSuspendEligibleEffects(effects); 7481 for (size_t i = 0; i < effects.size(); i++) { 7482 setEffectSuspended_l(&effects[i]->desc().type, true); 7483 } 7484 } 7485 } else { 7486 if (index < 0) { 7487 return; 7488 } 7489 desc = mSuspendedEffects.valueAt(index); 7490 if (desc->mRefCount <= 0) { 7491 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7492 desc->mRefCount = 1; 7493 } 7494 if (--desc->mRefCount == 0) { 7495 Vector<const effect_uuid_t *> types; 7496 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7497 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7498 continue; 7499 } 7500 types.add(&mSuspendedEffects.valueAt(i)->mType); 7501 } 7502 for (size_t i = 0; i < types.size(); i++) { 7503 setEffectSuspended_l(types[i], false); 7504 } 7505 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7506 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7507 } 7508 } 7509} 7510 7511 7512// The volume effect is used for automated tests only 7513#ifndef OPENSL_ES_H_ 7514static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7515 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7516const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7517#endif //OPENSL_ES_H_ 7518 7519bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7520{ 7521 // auxiliary effects and visualizer are never suspended on output mix 7522 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7523 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7524 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7525 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7526 return false; 7527 } 7528 return true; 7529} 7530 7531void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 7532{ 7533 effects.clear(); 7534 for (size_t i = 0; i < mEffects.size(); i++) { 7535 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 7536 effects.add(mEffects[i]); 7537 } 7538 } 7539} 7540 7541sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7542 const effect_uuid_t *type) 7543{ 7544 sp<EffectModule> effect = getEffectFromType_l(type); 7545 return effect != 0 && effect->isEnabled() ? effect : 0; 7546} 7547 7548void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7549 bool enabled) 7550{ 7551 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7552 if (enabled) { 7553 if (index < 0) { 7554 // if the effect is not suspend check if all effects are suspended 7555 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7556 if (index < 0) { 7557 return; 7558 } 7559 if (!isEffectEligibleForSuspend(effect->desc())) { 7560 return; 7561 } 7562 setEffectSuspended_l(&effect->desc().type, enabled); 7563 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7564 if (index < 0) { 7565 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7566 return; 7567 } 7568 } 7569 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7570 effect->desc().type.timeLow); 7571 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7572 // if effect is requested to suspended but was not yet enabled, supend it now. 7573 if (desc->mEffect == 0) { 7574 desc->mEffect = effect; 7575 effect->setEnabled(false); 7576 effect->setSuspended(true); 7577 } 7578 } else { 7579 if (index < 0) { 7580 return; 7581 } 7582 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7583 effect->desc().type.timeLow); 7584 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7585 desc->mEffect.clear(); 7586 effect->setSuspended(false); 7587 } 7588} 7589 7590#undef LOG_TAG 7591#define LOG_TAG "AudioFlinger" 7592 7593// ---------------------------------------------------------------------------- 7594 7595status_t AudioFlinger::onTransact( 7596 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7597{ 7598 return BnAudioFlinger::onTransact(code, data, reply, flags); 7599} 7600 7601}; // namespace android 7602