AudioFlinger.cpp revision 0ba18ec1b343a8de70924f87630dd1f329b00fe6
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), mExiting(false), 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 // keep a strong ref on ourself so that we won't get 1021 // destroyed in the middle of requestExitAndWait() 1022 sp <ThreadBase> strongMe = this; 1023 1024 ALOGV("ThreadBase::exit"); 1025 { 1026 AutoMutex lock(mLock); 1027 mExiting = true; 1028 requestExit(); 1029 mWaitWorkCV.signal(); 1030 } 1031 requestExitAndWait(); 1032} 1033 1034status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1035{ 1036 status_t status; 1037 1038 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1039 Mutex::Autolock _l(mLock); 1040 1041 mNewParameters.add(keyValuePairs); 1042 mWaitWorkCV.signal(); 1043 // wait condition with timeout in case the thread loop has exited 1044 // before the request could be processed 1045 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1046 status = mParamStatus; 1047 mWaitWorkCV.signal(); 1048 } else { 1049 status = TIMED_OUT; 1050 } 1051 return status; 1052} 1053 1054void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1055{ 1056 Mutex::Autolock _l(mLock); 1057 sendConfigEvent_l(event, param); 1058} 1059 1060// sendConfigEvent_l() must be called with ThreadBase::mLock held 1061void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1062{ 1063 ConfigEvent configEvent; 1064 configEvent.mEvent = event; 1065 configEvent.mParam = param; 1066 mConfigEvents.add(configEvent); 1067 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1068 mWaitWorkCV.signal(); 1069} 1070 1071void AudioFlinger::ThreadBase::processConfigEvents() 1072{ 1073 mLock.lock(); 1074 while(!mConfigEvents.isEmpty()) { 1075 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1076 ConfigEvent configEvent = mConfigEvents[0]; 1077 mConfigEvents.removeAt(0); 1078 // release mLock before locking AudioFlinger mLock: lock order is always 1079 // AudioFlinger then ThreadBase to avoid cross deadlock 1080 mLock.unlock(); 1081 mAudioFlinger->mLock.lock(); 1082 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1083 mAudioFlinger->mLock.unlock(); 1084 mLock.lock(); 1085 } 1086 mLock.unlock(); 1087} 1088 1089status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1090{ 1091 const size_t SIZE = 256; 1092 char buffer[SIZE]; 1093 String8 result; 1094 1095 bool locked = tryLock(mLock); 1096 if (!locked) { 1097 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1098 write(fd, buffer, strlen(buffer)); 1099 } 1100 1101 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1102 result.append(buffer); 1103 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1104 result.append(buffer); 1105 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1106 result.append(buffer); 1107 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1108 result.append(buffer); 1109 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1110 result.append(buffer); 1111 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1112 result.append(buffer); 1113 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1114 result.append(buffer); 1115 1116 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1117 result.append(buffer); 1118 result.append(" Index Command"); 1119 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1120 snprintf(buffer, SIZE, "\n %02d ", i); 1121 result.append(buffer); 1122 result.append(mNewParameters[i]); 1123 } 1124 1125 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1126 result.append(buffer); 1127 snprintf(buffer, SIZE, " Index event param\n"); 1128 result.append(buffer); 1129 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1130 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1131 result.append(buffer); 1132 } 1133 result.append("\n"); 1134 1135 write(fd, result.string(), result.size()); 1136 1137 if (locked) { 1138 mLock.unlock(); 1139 } 1140 return NO_ERROR; 1141} 1142 1143status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1144{ 1145 const size_t SIZE = 256; 1146 char buffer[SIZE]; 1147 String8 result; 1148 1149 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1150 write(fd, buffer, strlen(buffer)); 1151 1152 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1153 sp<EffectChain> chain = mEffectChains[i]; 1154 if (chain != 0) { 1155 chain->dump(fd, args); 1156 } 1157 } 1158 return NO_ERROR; 1159} 1160 1161void AudioFlinger::ThreadBase::acquireWakeLock() 1162{ 1163 Mutex::Autolock _l(mLock); 1164 acquireWakeLock_l(); 1165} 1166 1167void AudioFlinger::ThreadBase::acquireWakeLock_l() 1168{ 1169 if (mPowerManager == 0) { 1170 // use checkService() to avoid blocking if power service is not up yet 1171 sp<IBinder> binder = 1172 defaultServiceManager()->checkService(String16("power")); 1173 if (binder == 0) { 1174 ALOGW("Thread %s cannot connect to the power manager service", mName); 1175 } else { 1176 mPowerManager = interface_cast<IPowerManager>(binder); 1177 binder->linkToDeath(mDeathRecipient); 1178 } 1179 } 1180 if (mPowerManager != 0) { 1181 sp<IBinder> binder = new BBinder(); 1182 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1183 binder, 1184 String16(mName)); 1185 if (status == NO_ERROR) { 1186 mWakeLockToken = binder; 1187 } 1188 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1189 } 1190} 1191 1192void AudioFlinger::ThreadBase::releaseWakeLock() 1193{ 1194 Mutex::Autolock _l(mLock); 1195 releaseWakeLock_l(); 1196} 1197 1198void AudioFlinger::ThreadBase::releaseWakeLock_l() 1199{ 1200 if (mWakeLockToken != 0) { 1201 ALOGV("releaseWakeLock_l() %s", mName); 1202 if (mPowerManager != 0) { 1203 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1204 } 1205 mWakeLockToken.clear(); 1206 } 1207} 1208 1209void AudioFlinger::ThreadBase::clearPowerManager() 1210{ 1211 Mutex::Autolock _l(mLock); 1212 releaseWakeLock_l(); 1213 mPowerManager.clear(); 1214} 1215 1216void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1217{ 1218 sp<ThreadBase> thread = mThread.promote(); 1219 if (thread != 0) { 1220 thread->clearPowerManager(); 1221 } 1222 ALOGW("power manager service died !!!"); 1223} 1224 1225void AudioFlinger::ThreadBase::setEffectSuspended( 1226 const effect_uuid_t *type, bool suspend, int sessionId) 1227{ 1228 Mutex::Autolock _l(mLock); 1229 setEffectSuspended_l(type, suspend, sessionId); 1230} 1231 1232void AudioFlinger::ThreadBase::setEffectSuspended_l( 1233 const effect_uuid_t *type, bool suspend, int sessionId) 1234{ 1235 sp<EffectChain> chain = getEffectChain_l(sessionId); 1236 if (chain != 0) { 1237 if (type != NULL) { 1238 chain->setEffectSuspended_l(type, suspend); 1239 } else { 1240 chain->setEffectSuspendedAll_l(suspend); 1241 } 1242 } 1243 1244 updateSuspendedSessions_l(type, suspend, sessionId); 1245} 1246 1247void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1248{ 1249 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1250 if (index < 0) { 1251 return; 1252 } 1253 1254 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1255 mSuspendedSessions.editValueAt(index); 1256 1257 for (size_t i = 0; i < sessionEffects.size(); i++) { 1258 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1259 for (int j = 0; j < desc->mRefCount; j++) { 1260 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1261 chain->setEffectSuspendedAll_l(true); 1262 } else { 1263 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1264 desc->mType.timeLow); 1265 chain->setEffectSuspended_l(&desc->mType, true); 1266 } 1267 } 1268 } 1269} 1270 1271void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1272 bool suspend, 1273 int sessionId) 1274{ 1275 int index = mSuspendedSessions.indexOfKey(sessionId); 1276 1277 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1278 1279 if (suspend) { 1280 if (index >= 0) { 1281 sessionEffects = mSuspendedSessions.editValueAt(index); 1282 } else { 1283 mSuspendedSessions.add(sessionId, sessionEffects); 1284 } 1285 } else { 1286 if (index < 0) { 1287 return; 1288 } 1289 sessionEffects = mSuspendedSessions.editValueAt(index); 1290 } 1291 1292 1293 int key = EffectChain::kKeyForSuspendAll; 1294 if (type != NULL) { 1295 key = type->timeLow; 1296 } 1297 index = sessionEffects.indexOfKey(key); 1298 1299 sp <SuspendedSessionDesc> desc; 1300 if (suspend) { 1301 if (index >= 0) { 1302 desc = sessionEffects.valueAt(index); 1303 } else { 1304 desc = new SuspendedSessionDesc(); 1305 if (type != NULL) { 1306 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1307 } 1308 sessionEffects.add(key, desc); 1309 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1310 } 1311 desc->mRefCount++; 1312 } else { 1313 if (index < 0) { 1314 return; 1315 } 1316 desc = sessionEffects.valueAt(index); 1317 if (--desc->mRefCount == 0) { 1318 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1319 sessionEffects.removeItemsAt(index); 1320 if (sessionEffects.isEmpty()) { 1321 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1322 sessionId); 1323 mSuspendedSessions.removeItem(sessionId); 1324 } 1325 } 1326 } 1327 if (!sessionEffects.isEmpty()) { 1328 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1329 } 1330} 1331 1332void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1333 bool enabled, 1334 int sessionId) 1335{ 1336 Mutex::Autolock _l(mLock); 1337 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1338} 1339 1340void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1341 bool enabled, 1342 int sessionId) 1343{ 1344 if (mType != RECORD) { 1345 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1346 // another session. This gives the priority to well behaved effect control panels 1347 // and applications not using global effects. 1348 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1349 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1350 } 1351 } 1352 1353 sp<EffectChain> chain = getEffectChain_l(sessionId); 1354 if (chain != 0) { 1355 chain->checkSuspendOnEffectEnabled(effect, enabled); 1356 } 1357} 1358 1359// ---------------------------------------------------------------------------- 1360 1361AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1362 AudioStreamOut* output, 1363 audio_io_handle_t id, 1364 uint32_t device, 1365 type_t type) 1366 : ThreadBase(audioFlinger, id, device, type), 1367 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1368 // Assumes constructor is called by AudioFlinger with it's mLock held, 1369 // but it would be safer to explicitly pass initial masterMute as parameter 1370 mMasterMute(audioFlinger->masterMute_l()), 1371 // mStreamTypes[] initialized in constructor body 1372 mOutput(output), 1373 // Assumes constructor is called by AudioFlinger with it's mLock held, 1374 // but it would be safer to explicitly pass initial masterVolume as parameter 1375 mMasterVolume(audioFlinger->masterVolume_l()), 1376 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1377{ 1378 snprintf(mName, kNameLength, "AudioOut_%d", id); 1379 1380 readOutputParameters(); 1381 1382 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1383 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1384 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1385 stream = (audio_stream_type_t) (stream + 1)) { 1386 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1387 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1388 // initialized by stream_type_t default constructor 1389 // mStreamTypes[stream].valid = true; 1390 } 1391} 1392 1393AudioFlinger::PlaybackThread::~PlaybackThread() 1394{ 1395 delete [] mMixBuffer; 1396} 1397 1398status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1399{ 1400 dumpInternals(fd, args); 1401 dumpTracks(fd, args); 1402 dumpEffectChains(fd, args); 1403 return NO_ERROR; 1404} 1405 1406status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1407{ 1408 const size_t SIZE = 256; 1409 char buffer[SIZE]; 1410 String8 result; 1411 1412 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1413 result.append(buffer); 1414 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1415 for (size_t i = 0; i < mTracks.size(); ++i) { 1416 sp<Track> track = mTracks[i]; 1417 if (track != 0) { 1418 track->dump(buffer, SIZE); 1419 result.append(buffer); 1420 } 1421 } 1422 1423 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1424 result.append(buffer); 1425 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1426 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1427 sp<Track> track = mActiveTracks[i].promote(); 1428 if (track != 0) { 1429 track->dump(buffer, SIZE); 1430 result.append(buffer); 1431 } 1432 } 1433 write(fd, result.string(), result.size()); 1434 return NO_ERROR; 1435} 1436 1437status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1438{ 1439 const size_t SIZE = 256; 1440 char buffer[SIZE]; 1441 String8 result; 1442 1443 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1444 result.append(buffer); 1445 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1446 result.append(buffer); 1447 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1448 result.append(buffer); 1449 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1450 result.append(buffer); 1451 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1452 result.append(buffer); 1453 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1454 result.append(buffer); 1455 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1456 result.append(buffer); 1457 write(fd, result.string(), result.size()); 1458 1459 dumpBase(fd, args); 1460 1461 return NO_ERROR; 1462} 1463 1464// Thread virtuals 1465status_t AudioFlinger::PlaybackThread::readyToRun() 1466{ 1467 status_t status = initCheck(); 1468 if (status == NO_ERROR) { 1469 ALOGI("AudioFlinger's thread %p ready to run", this); 1470 } else { 1471 ALOGE("No working audio driver found."); 1472 } 1473 return status; 1474} 1475 1476void AudioFlinger::PlaybackThread::onFirstRef() 1477{ 1478 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1479} 1480 1481// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1482sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1483 const sp<AudioFlinger::Client>& client, 1484 audio_stream_type_t streamType, 1485 uint32_t sampleRate, 1486 audio_format_t format, 1487 uint32_t channelMask, 1488 int frameCount, 1489 const sp<IMemory>& sharedBuffer, 1490 int sessionId, 1491 status_t *status) 1492{ 1493 sp<Track> track; 1494 status_t lStatus; 1495 1496 if (mType == DIRECT) { 1497 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1498 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1499 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1500 "for output %p with format %d", 1501 sampleRate, format, channelMask, mOutput, mFormat); 1502 lStatus = BAD_VALUE; 1503 goto Exit; 1504 } 1505 } 1506 } else { 1507 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1508 if (sampleRate > mSampleRate*2) { 1509 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1510 lStatus = BAD_VALUE; 1511 goto Exit; 1512 } 1513 } 1514 1515 lStatus = initCheck(); 1516 if (lStatus != NO_ERROR) { 1517 ALOGE("Audio driver not initialized."); 1518 goto Exit; 1519 } 1520 1521 { // scope for mLock 1522 Mutex::Autolock _l(mLock); 1523 1524 // all tracks in same audio session must share the same routing strategy otherwise 1525 // conflicts will happen when tracks are moved from one output to another by audio policy 1526 // manager 1527 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1528 for (size_t i = 0; i < mTracks.size(); ++i) { 1529 sp<Track> t = mTracks[i]; 1530 if (t != 0) { 1531 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1532 if (sessionId == t->sessionId() && strategy != actual) { 1533 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1534 strategy, actual); 1535 lStatus = BAD_VALUE; 1536 goto Exit; 1537 } 1538 } 1539 } 1540 1541 track = new Track(this, client, streamType, sampleRate, format, 1542 channelMask, frameCount, sharedBuffer, sessionId); 1543 if (track->getCblk() == NULL || track->name() < 0) { 1544 lStatus = NO_MEMORY; 1545 goto Exit; 1546 } 1547 mTracks.add(track); 1548 1549 sp<EffectChain> chain = getEffectChain_l(sessionId); 1550 if (chain != 0) { 1551 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1552 track->setMainBuffer(chain->inBuffer()); 1553 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1554 chain->incTrackCnt(); 1555 } 1556 1557 // invalidate track immediately if the stream type was moved to another thread since 1558 // createTrack() was called by the client process. 1559 if (!mStreamTypes[streamType].valid) { 1560 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1561 this, streamType); 1562 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1563 } 1564 } 1565 lStatus = NO_ERROR; 1566 1567Exit: 1568 if(status) { 1569 *status = lStatus; 1570 } 1571 return track; 1572} 1573 1574uint32_t AudioFlinger::PlaybackThread::latency() const 1575{ 1576 Mutex::Autolock _l(mLock); 1577 if (initCheck() == NO_ERROR) { 1578 return mOutput->stream->get_latency(mOutput->stream); 1579 } else { 1580 return 0; 1581 } 1582} 1583 1584status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1585{ 1586 mMasterVolume = value; 1587 return NO_ERROR; 1588} 1589 1590status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1591{ 1592 mMasterMute = muted; 1593 return NO_ERROR; 1594} 1595 1596status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1597{ 1598 mStreamTypes[stream].volume = value; 1599 return NO_ERROR; 1600} 1601 1602status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1603{ 1604 mStreamTypes[stream].mute = muted; 1605 return NO_ERROR; 1606} 1607 1608float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1609{ 1610 return mStreamTypes[stream].volume; 1611} 1612 1613bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1614{ 1615 return mStreamTypes[stream].mute; 1616} 1617 1618// addTrack_l() must be called with ThreadBase::mLock held 1619status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1620{ 1621 status_t status = ALREADY_EXISTS; 1622 1623 // set retry count for buffer fill 1624 track->mRetryCount = kMaxTrackStartupRetries; 1625 if (mActiveTracks.indexOf(track) < 0) { 1626 // the track is newly added, make sure it fills up all its 1627 // buffers before playing. This is to ensure the client will 1628 // effectively get the latency it requested. 1629 track->mFillingUpStatus = Track::FS_FILLING; 1630 track->mResetDone = false; 1631 mActiveTracks.add(track); 1632 if (track->mainBuffer() != mMixBuffer) { 1633 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1634 if (chain != 0) { 1635 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1636 chain->incActiveTrackCnt(); 1637 } 1638 } 1639 1640 status = NO_ERROR; 1641 } 1642 1643 ALOGV("mWaitWorkCV.broadcast"); 1644 mWaitWorkCV.broadcast(); 1645 1646 return status; 1647} 1648 1649// destroyTrack_l() must be called with ThreadBase::mLock held 1650void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1651{ 1652 track->mState = TrackBase::TERMINATED; 1653 if (mActiveTracks.indexOf(track) < 0) { 1654 removeTrack_l(track); 1655 } 1656} 1657 1658void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1659{ 1660 mTracks.remove(track); 1661 deleteTrackName_l(track->name()); 1662 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1663 if (chain != 0) { 1664 chain->decTrackCnt(); 1665 } 1666} 1667 1668String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1669{ 1670 String8 out_s8 = String8(""); 1671 char *s; 1672 1673 Mutex::Autolock _l(mLock); 1674 if (initCheck() != NO_ERROR) { 1675 return out_s8; 1676 } 1677 1678 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1679 out_s8 = String8(s); 1680 free(s); 1681 return out_s8; 1682} 1683 1684// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1685void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1686 AudioSystem::OutputDescriptor desc; 1687 void *param2 = NULL; 1688 1689 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1690 1691 switch (event) { 1692 case AudioSystem::OUTPUT_OPENED: 1693 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1694 desc.channels = mChannelMask; 1695 desc.samplingRate = mSampleRate; 1696 desc.format = mFormat; 1697 desc.frameCount = mFrameCount; 1698 desc.latency = latency(); 1699 param2 = &desc; 1700 break; 1701 1702 case AudioSystem::STREAM_CONFIG_CHANGED: 1703 param2 = ¶m; 1704 case AudioSystem::OUTPUT_CLOSED: 1705 default: 1706 break; 1707 } 1708 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1709} 1710 1711void AudioFlinger::PlaybackThread::readOutputParameters() 1712{ 1713 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1714 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1715 mChannelCount = (uint16_t)popcount(mChannelMask); 1716 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1717 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1718 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1719 1720 // FIXME - Current mixer implementation only supports stereo output: Always 1721 // Allocate a stereo buffer even if HW output is mono. 1722 delete[] mMixBuffer; 1723 mMixBuffer = new int16_t[mFrameCount * 2]; 1724 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1725 1726 // force reconfiguration of effect chains and engines to take new buffer size and audio 1727 // parameters into account 1728 // Note that mLock is not held when readOutputParameters() is called from the constructor 1729 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1730 // matter. 1731 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1732 Vector< sp<EffectChain> > effectChains = mEffectChains; 1733 for (size_t i = 0; i < effectChains.size(); i ++) { 1734 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1735 } 1736} 1737 1738status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1739{ 1740 if (halFrames == NULL || dspFrames == NULL) { 1741 return BAD_VALUE; 1742 } 1743 Mutex::Autolock _l(mLock); 1744 if (initCheck() != NO_ERROR) { 1745 return INVALID_OPERATION; 1746 } 1747 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1748 1749 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1750} 1751 1752uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1753{ 1754 Mutex::Autolock _l(mLock); 1755 uint32_t result = 0; 1756 if (getEffectChain_l(sessionId) != 0) { 1757 result = EFFECT_SESSION; 1758 } 1759 1760 for (size_t i = 0; i < mTracks.size(); ++i) { 1761 sp<Track> track = mTracks[i]; 1762 if (sessionId == track->sessionId() && 1763 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1764 result |= TRACK_SESSION; 1765 break; 1766 } 1767 } 1768 1769 return result; 1770} 1771 1772uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1773{ 1774 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1775 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1776 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1777 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1778 } 1779 for (size_t i = 0; i < mTracks.size(); i++) { 1780 sp<Track> track = mTracks[i]; 1781 if (sessionId == track->sessionId() && 1782 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1783 return AudioSystem::getStrategyForStream(track->streamType()); 1784 } 1785 } 1786 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1787} 1788 1789 1790AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1791{ 1792 Mutex::Autolock _l(mLock); 1793 return mOutput; 1794} 1795 1796AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1797{ 1798 Mutex::Autolock _l(mLock); 1799 AudioStreamOut *output = mOutput; 1800 mOutput = NULL; 1801 return output; 1802} 1803 1804// this method must always be called either with ThreadBase mLock held or inside the thread loop 1805audio_stream_t* AudioFlinger::PlaybackThread::stream() 1806{ 1807 if (mOutput == NULL) { 1808 return NULL; 1809 } 1810 return &mOutput->stream->common; 1811} 1812 1813uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1814{ 1815 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1816 // decoding and transfer time. So sleeping for half of the latency would likely cause 1817 // underruns 1818 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1819 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1820 } else { 1821 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1822 } 1823} 1824 1825// ---------------------------------------------------------------------------- 1826 1827AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1828 audio_io_handle_t id, uint32_t device, type_t type) 1829 : PlaybackThread(audioFlinger, output, id, device, type), 1830 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1831 mPrevMixerStatus(MIXER_IDLE) 1832{ 1833 // FIXME - Current mixer implementation only supports stereo output 1834 if (mChannelCount == 1) { 1835 ALOGE("Invalid audio hardware channel count"); 1836 } 1837} 1838 1839AudioFlinger::MixerThread::~MixerThread() 1840{ 1841 delete mAudioMixer; 1842} 1843 1844bool AudioFlinger::MixerThread::threadLoop() 1845{ 1846 Vector< sp<Track> > tracksToRemove; 1847 mixer_state mixerStatus = MIXER_IDLE; 1848 nsecs_t standbyTime = systemTime(); 1849 size_t mixBufferSize = mFrameCount * mFrameSize; 1850 // FIXME: Relaxed timing because of a certain device that can't meet latency 1851 // Should be reduced to 2x after the vendor fixes the driver issue 1852 // increase threshold again due to low power audio mode. The way this warning threshold is 1853 // calculated and its usefulness should be reconsidered anyway. 1854 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1855 nsecs_t lastWarning = 0; 1856 bool longStandbyExit = false; 1857 uint32_t activeSleepTime = activeSleepTimeUs(); 1858 uint32_t idleSleepTime = idleSleepTimeUs(); 1859 uint32_t sleepTime = idleSleepTime; 1860 uint32_t sleepTimeShift = 0; 1861 Vector< sp<EffectChain> > effectChains; 1862#ifdef DEBUG_CPU_USAGE 1863 ThreadCpuUsage cpu; 1864 const CentralTendencyStatistics& stats = cpu.statistics(); 1865#endif 1866 1867 acquireWakeLock(); 1868 1869 while (!exitPending()) 1870 { 1871#ifdef DEBUG_CPU_USAGE 1872 cpu.sampleAndEnable(); 1873 unsigned n = stats.n(); 1874 // cpu.elapsed() is expensive, so don't call it every loop 1875 if ((n & 127) == 1) { 1876 long long elapsed = cpu.elapsed(); 1877 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1878 double perLoop = elapsed / (double) n; 1879 double perLoop100 = perLoop * 0.01; 1880 double mean = stats.mean(); 1881 double stddev = stats.stddev(); 1882 double minimum = stats.minimum(); 1883 double maximum = stats.maximum(); 1884 cpu.resetStatistics(); 1885 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", 1886 elapsed * .000000001, n, perLoop * .000001, 1887 mean * .001, 1888 stddev * .001, 1889 minimum * .001, 1890 maximum * .001, 1891 mean / perLoop100, 1892 stddev / perLoop100, 1893 minimum / perLoop100, 1894 maximum / perLoop100); 1895 } 1896 } 1897#endif 1898 processConfigEvents(); 1899 1900 mixerStatus = MIXER_IDLE; 1901 { // scope for mLock 1902 1903 Mutex::Autolock _l(mLock); 1904 1905 if (checkForNewParameters_l()) { 1906 mixBufferSize = mFrameCount * mFrameSize; 1907 // FIXME: Relaxed timing because of a certain device that can't meet latency 1908 // Should be reduced to 2x after the vendor fixes the driver issue 1909 // increase threshold again due to low power audio mode. The way this warning 1910 // threshold is calculated and its usefulness should be reconsidered anyway. 1911 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1912 activeSleepTime = activeSleepTimeUs(); 1913 idleSleepTime = idleSleepTimeUs(); 1914 } 1915 1916 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1917 1918 // put audio hardware into standby after short delay 1919 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1920 mSuspended)) { 1921 if (!mStandby) { 1922 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 1923 mOutput->stream->common.standby(&mOutput->stream->common); 1924 mStandby = true; 1925 mBytesWritten = 0; 1926 } 1927 1928 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1929 // we're about to wait, flush the binder command buffer 1930 IPCThreadState::self()->flushCommands(); 1931 1932 if (exitPending()) break; 1933 1934 releaseWakeLock_l(); 1935 // wait until we have something to do... 1936 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 1937 mWaitWorkCV.wait(mLock); 1938 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 1939 acquireWakeLock_l(); 1940 1941 mPrevMixerStatus = MIXER_IDLE; 1942 if (!mMasterMute) { 1943 char value[PROPERTY_VALUE_MAX]; 1944 property_get("ro.audio.silent", value, "0"); 1945 if (atoi(value)) { 1946 ALOGD("Silence is golden"); 1947 setMasterMute(true); 1948 } 1949 } 1950 1951 standbyTime = systemTime() + kStandbyTimeInNsecs; 1952 sleepTime = idleSleepTime; 1953 sleepTimeShift = 0; 1954 continue; 1955 } 1956 } 1957 1958 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1959 1960 // prevent any changes in effect chain list and in each effect chain 1961 // during mixing and effect process as the audio buffers could be deleted 1962 // or modified if an effect is created or deleted 1963 lockEffectChains_l(effectChains); 1964 } 1965 1966 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1967 // mix buffers... 1968 mAudioMixer->process(); 1969 // increase sleep time progressively when application underrun condition clears. 1970 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1971 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1972 // such that we would underrun the audio HAL. 1973 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 1974 sleepTimeShift--; 1975 } 1976 sleepTime = 0; 1977 standbyTime = systemTime() + kStandbyTimeInNsecs; 1978 //TODO: delay standby when effects have a tail 1979 } else { 1980 // If no tracks are ready, sleep once for the duration of an output 1981 // buffer size, then write 0s to the output 1982 if (sleepTime == 0) { 1983 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1984 sleepTime = activeSleepTime >> sleepTimeShift; 1985 if (sleepTime < kMinThreadSleepTimeUs) { 1986 sleepTime = kMinThreadSleepTimeUs; 1987 } 1988 // reduce sleep time in case of consecutive application underruns to avoid 1989 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 1990 // duration we would end up writing less data than needed by the audio HAL if 1991 // the condition persists. 1992 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 1993 sleepTimeShift++; 1994 } 1995 } else { 1996 sleepTime = idleSleepTime; 1997 } 1998 } else if (mBytesWritten != 0 || 1999 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2000 memset (mMixBuffer, 0, mixBufferSize); 2001 sleepTime = 0; 2002 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2003 } 2004 // TODO add standby time extension fct of effect tail 2005 } 2006 2007 if (mSuspended) { 2008 sleepTime = suspendSleepTimeUs(); 2009 } 2010 // sleepTime == 0 means we must write to audio hardware 2011 if (sleepTime == 0) { 2012 for (size_t i = 0; i < effectChains.size(); i ++) { 2013 effectChains[i]->process_l(); 2014 } 2015 // enable changes in effect chain 2016 unlockEffectChains(effectChains); 2017 mLastWriteTime = systemTime(); 2018 mInWrite = true; 2019 mBytesWritten += mixBufferSize; 2020 2021 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2022 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2023 mNumWrites++; 2024 mInWrite = false; 2025 nsecs_t now = systemTime(); 2026 nsecs_t delta = now - mLastWriteTime; 2027 if (!mStandby && delta > maxPeriod) { 2028 mNumDelayedWrites++; 2029 if ((now - lastWarning) > kWarningThrottleNs) { 2030 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2031 ns2ms(delta), mNumDelayedWrites, this); 2032 lastWarning = now; 2033 } 2034 if (mStandby) { 2035 longStandbyExit = true; 2036 } 2037 } 2038 mStandby = false; 2039 } else { 2040 // enable changes in effect chain 2041 unlockEffectChains(effectChains); 2042 usleep(sleepTime); 2043 } 2044 2045 // finally let go of all our tracks, without the lock held 2046 // since we can't guarantee the destructors won't acquire that 2047 // same lock. 2048 tracksToRemove.clear(); 2049 2050 // Effect chains will be actually deleted here if they were removed from 2051 // mEffectChains list during mixing or effects processing 2052 effectChains.clear(); 2053 } 2054 2055 if (!mStandby) { 2056 mOutput->stream->common.standby(&mOutput->stream->common); 2057 } 2058 2059 releaseWakeLock(); 2060 2061 ALOGV("MixerThread %p exiting", this); 2062 return false; 2063} 2064 2065// prepareTracks_l() must be called with ThreadBase::mLock held 2066AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2067 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2068{ 2069 2070 mixer_state mixerStatus = MIXER_IDLE; 2071 // find out which tracks need to be processed 2072 size_t count = activeTracks.size(); 2073 size_t mixedTracks = 0; 2074 size_t tracksWithEffect = 0; 2075 2076 float masterVolume = mMasterVolume; 2077 bool masterMute = mMasterMute; 2078 2079 if (masterMute) { 2080 masterVolume = 0; 2081 } 2082 // Delegate master volume control to effect in output mix effect chain if needed 2083 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2084 if (chain != 0) { 2085 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2086 chain->setVolume_l(&v, &v); 2087 masterVolume = (float)((v + (1 << 23)) >> 24); 2088 chain.clear(); 2089 } 2090 2091 for (size_t i=0 ; i<count ; i++) { 2092 sp<Track> t = activeTracks[i].promote(); 2093 if (t == 0) continue; 2094 2095 // this const just means the local variable doesn't change 2096 Track* const track = t.get(); 2097 audio_track_cblk_t* cblk = track->cblk(); 2098 2099 // The first time a track is added we wait 2100 // for all its buffers to be filled before processing it 2101 int name = track->name(); 2102 // make sure that we have enough frames to mix one full buffer. 2103 // enforce this condition only once to enable draining the buffer in case the client 2104 // app does not call stop() and relies on underrun to stop: 2105 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2106 // during last round 2107 uint32_t minFrames = 1; 2108 if (!track->isStopped() && !track->isPausing() && 2109 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2110 if (t->sampleRate() == (int)mSampleRate) { 2111 minFrames = mFrameCount; 2112 } else { 2113 // +1 for rounding and +1 for additional sample needed for interpolation 2114 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2115 // add frames already consumed but not yet released by the resampler 2116 // because cblk->framesReady() will include these frames 2117 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2118 // the minimum track buffer size is normally twice the number of frames necessary 2119 // to fill one buffer and the resampler should not leave more than one buffer worth 2120 // of unreleased frames after each pass, but just in case... 2121 ALOG_ASSERT(minFrames <= cblk->frameCount); 2122 } 2123 } 2124 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2125 !track->isPaused() && !track->isTerminated()) 2126 { 2127 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2128 2129 mixedTracks++; 2130 2131 // track->mainBuffer() != mMixBuffer means there is an effect chain 2132 // connected to the track 2133 chain.clear(); 2134 if (track->mainBuffer() != mMixBuffer) { 2135 chain = getEffectChain_l(track->sessionId()); 2136 // Delegate volume control to effect in track effect chain if needed 2137 if (chain != 0) { 2138 tracksWithEffect++; 2139 } else { 2140 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2141 name, track->sessionId()); 2142 } 2143 } 2144 2145 2146 int param = AudioMixer::VOLUME; 2147 if (track->mFillingUpStatus == Track::FS_FILLED) { 2148 // no ramp for the first volume setting 2149 track->mFillingUpStatus = Track::FS_ACTIVE; 2150 if (track->mState == TrackBase::RESUMING) { 2151 track->mState = TrackBase::ACTIVE; 2152 param = AudioMixer::RAMP_VOLUME; 2153 } 2154 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2155 } else if (cblk->server != 0) { 2156 // If the track is stopped before the first frame was mixed, 2157 // do not apply ramp 2158 param = AudioMixer::RAMP_VOLUME; 2159 } 2160 2161 // compute volume for this track 2162 uint32_t vl, vr, va; 2163 if (track->isMuted() || track->isPausing() || 2164 mStreamTypes[track->streamType()].mute) { 2165 vl = vr = va = 0; 2166 if (track->isPausing()) { 2167 track->setPaused(); 2168 } 2169 } else { 2170 2171 // read original volumes with volume control 2172 float typeVolume = mStreamTypes[track->streamType()].volume; 2173 float v = masterVolume * typeVolume; 2174 uint32_t vlr = cblk->getVolumeLR(); 2175 vl = vlr & 0xFFFF; 2176 vr = vlr >> 16; 2177 // track volumes come from shared memory, so can't be trusted and must be clamped 2178 if (vl > MAX_GAIN_INT) { 2179 ALOGV("Track left volume out of range: %04X", vl); 2180 vl = MAX_GAIN_INT; 2181 } 2182 if (vr > MAX_GAIN_INT) { 2183 ALOGV("Track right volume out of range: %04X", vr); 2184 vr = MAX_GAIN_INT; 2185 } 2186 // now apply the master volume and stream type volume 2187 vl = (uint32_t)(v * vl) << 12; 2188 vr = (uint32_t)(v * vr) << 12; 2189 // assuming master volume and stream type volume each go up to 1.0, 2190 // vl and vr are now in 8.24 format 2191 2192 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2193 // send level comes from shared memory and so may be corrupt 2194 if (sendLevel >= MAX_GAIN_INT) { 2195 ALOGV("Track send level out of range: %04X", sendLevel); 2196 sendLevel = MAX_GAIN_INT; 2197 } 2198 va = (uint32_t)(v * sendLevel); 2199 } 2200 // Delegate volume control to effect in track effect chain if needed 2201 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2202 // Do not ramp volume if volume is controlled by effect 2203 param = AudioMixer::VOLUME; 2204 track->mHasVolumeController = true; 2205 } else { 2206 // force no volume ramp when volume controller was just disabled or removed 2207 // from effect chain to avoid volume spike 2208 if (track->mHasVolumeController) { 2209 param = AudioMixer::VOLUME; 2210 } 2211 track->mHasVolumeController = false; 2212 } 2213 2214 // Convert volumes from 8.24 to 4.12 format 2215 int16_t left, right, aux; 2216 // This additional clamping is needed in case chain->setVolume_l() overshot 2217 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2218 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2219 left = int16_t(v_clamped); 2220 v_clamped = (vr + (1 << 11)) >> 12; 2221 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2222 right = int16_t(v_clamped); 2223 2224 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2225 aux = int16_t(va); 2226 2227 // XXX: these things DON'T need to be done each time 2228 mAudioMixer->setBufferProvider(name, track); 2229 mAudioMixer->enable(name); 2230 2231 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2232 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2233 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2234 mAudioMixer->setParameter( 2235 name, 2236 AudioMixer::TRACK, 2237 AudioMixer::FORMAT, (void *)track->format()); 2238 mAudioMixer->setParameter( 2239 name, 2240 AudioMixer::TRACK, 2241 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2242 mAudioMixer->setParameter( 2243 name, 2244 AudioMixer::RESAMPLE, 2245 AudioMixer::SAMPLE_RATE, 2246 (void *)(cblk->sampleRate)); 2247 mAudioMixer->setParameter( 2248 name, 2249 AudioMixer::TRACK, 2250 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2251 mAudioMixer->setParameter( 2252 name, 2253 AudioMixer::TRACK, 2254 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2255 2256 // reset retry count 2257 track->mRetryCount = kMaxTrackRetries; 2258 // If one track is ready, set the mixer ready if: 2259 // - the mixer was not ready during previous round OR 2260 // - no other track is not ready 2261 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2262 mixerStatus != MIXER_TRACKS_ENABLED) { 2263 mixerStatus = MIXER_TRACKS_READY; 2264 } 2265 } else { 2266 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2267 if (track->isStopped()) { 2268 track->reset(); 2269 } 2270 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2271 // We have consumed all the buffers of this track. 2272 // Remove it from the list of active tracks. 2273 tracksToRemove->add(track); 2274 } else { 2275 // No buffers for this track. Give it a few chances to 2276 // fill a buffer, then remove it from active list. 2277 if (--(track->mRetryCount) <= 0) { 2278 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2279 tracksToRemove->add(track); 2280 // indicate to client process that the track was disabled because of underrun 2281 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2282 // If one track is not ready, mark the mixer also not ready if: 2283 // - the mixer was ready during previous round OR 2284 // - no other track is ready 2285 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2286 mixerStatus != MIXER_TRACKS_READY) { 2287 mixerStatus = MIXER_TRACKS_ENABLED; 2288 } 2289 } 2290 mAudioMixer->disable(name); 2291 } 2292 } 2293 2294 // remove all the tracks that need to be... 2295 count = tracksToRemove->size(); 2296 if (CC_UNLIKELY(count)) { 2297 for (size_t i=0 ; i<count ; i++) { 2298 const sp<Track>& track = tracksToRemove->itemAt(i); 2299 mActiveTracks.remove(track); 2300 if (track->mainBuffer() != mMixBuffer) { 2301 chain = getEffectChain_l(track->sessionId()); 2302 if (chain != 0) { 2303 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2304 chain->decActiveTrackCnt(); 2305 } 2306 } 2307 if (track->isTerminated()) { 2308 removeTrack_l(track); 2309 } 2310 } 2311 } 2312 2313 // mix buffer must be cleared if all tracks are connected to an 2314 // effect chain as in this case the mixer will not write to 2315 // mix buffer and track effects will accumulate into it 2316 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2317 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2318 } 2319 2320 mPrevMixerStatus = mixerStatus; 2321 return mixerStatus; 2322} 2323 2324void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2325{ 2326 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2327 this, streamType, mTracks.size()); 2328 Mutex::Autolock _l(mLock); 2329 2330 size_t size = mTracks.size(); 2331 for (size_t i = 0; i < size; i++) { 2332 sp<Track> t = mTracks[i]; 2333 if (t->streamType() == streamType) { 2334 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2335 t->mCblk->cv.signal(); 2336 } 2337 } 2338} 2339 2340void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2341{ 2342 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2343 this, streamType, valid); 2344 Mutex::Autolock _l(mLock); 2345 2346 mStreamTypes[streamType].valid = valid; 2347} 2348 2349// getTrackName_l() must be called with ThreadBase::mLock held 2350int AudioFlinger::MixerThread::getTrackName_l() 2351{ 2352 return mAudioMixer->getTrackName(); 2353} 2354 2355// deleteTrackName_l() must be called with ThreadBase::mLock held 2356void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2357{ 2358 ALOGV("remove track (%d) and delete from mixer", name); 2359 mAudioMixer->deleteTrackName(name); 2360} 2361 2362// checkForNewParameters_l() must be called with ThreadBase::mLock held 2363bool AudioFlinger::MixerThread::checkForNewParameters_l() 2364{ 2365 bool reconfig = false; 2366 2367 while (!mNewParameters.isEmpty()) { 2368 status_t status = NO_ERROR; 2369 String8 keyValuePair = mNewParameters[0]; 2370 AudioParameter param = AudioParameter(keyValuePair); 2371 int value; 2372 2373 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2374 reconfig = true; 2375 } 2376 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2377 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2378 status = BAD_VALUE; 2379 } else { 2380 reconfig = true; 2381 } 2382 } 2383 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2384 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2385 status = BAD_VALUE; 2386 } else { 2387 reconfig = true; 2388 } 2389 } 2390 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2391 // do not accept frame count changes if tracks are open as the track buffer 2392 // size depends on frame count and correct behavior would not be guaranteed 2393 // if frame count is changed after track creation 2394 if (!mTracks.isEmpty()) { 2395 status = INVALID_OPERATION; 2396 } else { 2397 reconfig = true; 2398 } 2399 } 2400 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2401 // when changing the audio output device, call addBatteryData to notify 2402 // the change 2403 if ((int)mDevice != value) { 2404 uint32_t params = 0; 2405 // check whether speaker is on 2406 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2407 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2408 } 2409 2410 int deviceWithoutSpeaker 2411 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2412 // check if any other device (except speaker) is on 2413 if (value & deviceWithoutSpeaker ) { 2414 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2415 } 2416 2417 if (params != 0) { 2418 addBatteryData(params); 2419 } 2420 } 2421 2422 // forward device change to effects that have requested to be 2423 // aware of attached audio device. 2424 mDevice = (uint32_t)value; 2425 for (size_t i = 0; i < mEffectChains.size(); i++) { 2426 mEffectChains[i]->setDevice_l(mDevice); 2427 } 2428 } 2429 2430 if (status == NO_ERROR) { 2431 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2432 keyValuePair.string()); 2433 if (!mStandby && status == INVALID_OPERATION) { 2434 mOutput->stream->common.standby(&mOutput->stream->common); 2435 mStandby = true; 2436 mBytesWritten = 0; 2437 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2438 keyValuePair.string()); 2439 } 2440 if (status == NO_ERROR && reconfig) { 2441 delete mAudioMixer; 2442 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2443 mAudioMixer = NULL; 2444 readOutputParameters(); 2445 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2446 for (size_t i = 0; i < mTracks.size() ; i++) { 2447 int name = getTrackName_l(); 2448 if (name < 0) break; 2449 mTracks[i]->mName = name; 2450 // limit track sample rate to 2 x new output sample rate 2451 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2452 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2453 } 2454 } 2455 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2456 } 2457 } 2458 2459 mNewParameters.removeAt(0); 2460 2461 mParamStatus = status; 2462 mParamCond.signal(); 2463 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2464 // already timed out waiting for the status and will never signal the condition. 2465 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2466 } 2467 return reconfig; 2468} 2469 2470status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2471{ 2472 const size_t SIZE = 256; 2473 char buffer[SIZE]; 2474 String8 result; 2475 2476 PlaybackThread::dumpInternals(fd, args); 2477 2478 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2479 result.append(buffer); 2480 write(fd, result.string(), result.size()); 2481 return NO_ERROR; 2482} 2483 2484uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2485{ 2486 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2487} 2488 2489uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2490{ 2491 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2492} 2493 2494// ---------------------------------------------------------------------------- 2495AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2496 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2497 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2498 // mLeftVolFloat, mRightVolFloat 2499 // mLeftVolShort, mRightVolShort 2500{ 2501} 2502 2503AudioFlinger::DirectOutputThread::~DirectOutputThread() 2504{ 2505} 2506 2507static inline 2508int32_t mul(int16_t in, int16_t v) 2509{ 2510#if defined(__arm__) && !defined(__thumb__) 2511 int32_t out; 2512 asm( "smulbb %[out], %[in], %[v] \n" 2513 : [out]"=r"(out) 2514 : [in]"%r"(in), [v]"r"(v) 2515 : ); 2516 return out; 2517#else 2518 return in * int32_t(v); 2519#endif 2520} 2521 2522void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2523{ 2524 // Do not apply volume on compressed audio 2525 if (!audio_is_linear_pcm(mFormat)) { 2526 return; 2527 } 2528 2529 // convert to signed 16 bit before volume calculation 2530 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2531 size_t count = mFrameCount * mChannelCount; 2532 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2533 int16_t *dst = mMixBuffer + count-1; 2534 while(count--) { 2535 *dst-- = (int16_t)(*src--^0x80) << 8; 2536 } 2537 } 2538 2539 size_t frameCount = mFrameCount; 2540 int16_t *out = mMixBuffer; 2541 if (ramp) { 2542 if (mChannelCount == 1) { 2543 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2544 int32_t vlInc = d / (int32_t)frameCount; 2545 int32_t vl = ((int32_t)mLeftVolShort << 16); 2546 do { 2547 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2548 out++; 2549 vl += vlInc; 2550 } while (--frameCount); 2551 2552 } else { 2553 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2554 int32_t vlInc = d / (int32_t)frameCount; 2555 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2556 int32_t vrInc = d / (int32_t)frameCount; 2557 int32_t vl = ((int32_t)mLeftVolShort << 16); 2558 int32_t vr = ((int32_t)mRightVolShort << 16); 2559 do { 2560 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2561 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2562 out += 2; 2563 vl += vlInc; 2564 vr += vrInc; 2565 } while (--frameCount); 2566 } 2567 } else { 2568 if (mChannelCount == 1) { 2569 do { 2570 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2571 out++; 2572 } while (--frameCount); 2573 } else { 2574 do { 2575 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2576 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2577 out += 2; 2578 } while (--frameCount); 2579 } 2580 } 2581 2582 // convert back to unsigned 8 bit after volume calculation 2583 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2584 size_t count = mFrameCount * mChannelCount; 2585 int16_t *src = mMixBuffer; 2586 uint8_t *dst = (uint8_t *)mMixBuffer; 2587 while(count--) { 2588 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2589 } 2590 } 2591 2592 mLeftVolShort = leftVol; 2593 mRightVolShort = rightVol; 2594} 2595 2596bool AudioFlinger::DirectOutputThread::threadLoop() 2597{ 2598 mixer_state mixerStatus = MIXER_IDLE; 2599 sp<Track> trackToRemove; 2600 sp<Track> activeTrack; 2601 nsecs_t standbyTime = systemTime(); 2602 int8_t *curBuf; 2603 size_t mixBufferSize = mFrameCount*mFrameSize; 2604 uint32_t activeSleepTime = activeSleepTimeUs(); 2605 uint32_t idleSleepTime = idleSleepTimeUs(); 2606 uint32_t sleepTime = idleSleepTime; 2607 // use shorter standby delay as on normal output to release 2608 // hardware resources as soon as possible 2609 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2610 2611 acquireWakeLock(); 2612 2613 while (!exitPending()) 2614 { 2615 bool rampVolume; 2616 uint16_t leftVol; 2617 uint16_t rightVol; 2618 Vector< sp<EffectChain> > effectChains; 2619 2620 processConfigEvents(); 2621 2622 mixerStatus = MIXER_IDLE; 2623 2624 { // scope for the mLock 2625 2626 Mutex::Autolock _l(mLock); 2627 2628 if (checkForNewParameters_l()) { 2629 mixBufferSize = mFrameCount*mFrameSize; 2630 activeSleepTime = activeSleepTimeUs(); 2631 idleSleepTime = idleSleepTimeUs(); 2632 standbyDelay = microseconds(activeSleepTime*2); 2633 } 2634 2635 // put audio hardware into standby after short delay 2636 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2637 mSuspended)) { 2638 // wait until we have something to do... 2639 if (!mStandby) { 2640 ALOGV("Audio hardware entering standby, mixer %p", this); 2641 mOutput->stream->common.standby(&mOutput->stream->common); 2642 mStandby = true; 2643 mBytesWritten = 0; 2644 } 2645 2646 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2647 // we're about to wait, flush the binder command buffer 2648 IPCThreadState::self()->flushCommands(); 2649 2650 if (exitPending()) break; 2651 2652 releaseWakeLock_l(); 2653 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2654 mWaitWorkCV.wait(mLock); 2655 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2656 acquireWakeLock_l(); 2657 2658 if (!mMasterMute) { 2659 char value[PROPERTY_VALUE_MAX]; 2660 property_get("ro.audio.silent", value, "0"); 2661 if (atoi(value)) { 2662 ALOGD("Silence is golden"); 2663 setMasterMute(true); 2664 } 2665 } 2666 2667 standbyTime = systemTime() + standbyDelay; 2668 sleepTime = idleSleepTime; 2669 continue; 2670 } 2671 } 2672 2673 effectChains = mEffectChains; 2674 2675 // find out which tracks need to be processed 2676 if (mActiveTracks.size() != 0) { 2677 sp<Track> t = mActiveTracks[0].promote(); 2678 if (t == 0) continue; 2679 2680 Track* const track = t.get(); 2681 audio_track_cblk_t* cblk = track->cblk(); 2682 2683 // The first time a track is added we wait 2684 // for all its buffers to be filled before processing it 2685 if (cblk->framesReady() && track->isReady() && 2686 !track->isPaused() && !track->isTerminated()) 2687 { 2688 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2689 2690 if (track->mFillingUpStatus == Track::FS_FILLED) { 2691 track->mFillingUpStatus = Track::FS_ACTIVE; 2692 mLeftVolFloat = mRightVolFloat = 0; 2693 mLeftVolShort = mRightVolShort = 0; 2694 if (track->mState == TrackBase::RESUMING) { 2695 track->mState = TrackBase::ACTIVE; 2696 rampVolume = true; 2697 } 2698 } else if (cblk->server != 0) { 2699 // If the track is stopped before the first frame was mixed, 2700 // do not apply ramp 2701 rampVolume = true; 2702 } 2703 // compute volume for this track 2704 float left, right; 2705 if (track->isMuted() || mMasterMute || track->isPausing() || 2706 mStreamTypes[track->streamType()].mute) { 2707 left = right = 0; 2708 if (track->isPausing()) { 2709 track->setPaused(); 2710 } 2711 } else { 2712 float typeVolume = mStreamTypes[track->streamType()].volume; 2713 float v = mMasterVolume * typeVolume; 2714 uint32_t vlr = cblk->getVolumeLR(); 2715 float v_clamped = v * (vlr & 0xFFFF); 2716 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2717 left = v_clamped/MAX_GAIN; 2718 v_clamped = v * (vlr >> 16); 2719 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2720 right = v_clamped/MAX_GAIN; 2721 } 2722 2723 if (left != mLeftVolFloat || right != mRightVolFloat) { 2724 mLeftVolFloat = left; 2725 mRightVolFloat = right; 2726 2727 // If audio HAL implements volume control, 2728 // force software volume to nominal value 2729 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2730 left = 1.0f; 2731 right = 1.0f; 2732 } 2733 2734 // Convert volumes from float to 8.24 2735 uint32_t vl = (uint32_t)(left * (1 << 24)); 2736 uint32_t vr = (uint32_t)(right * (1 << 24)); 2737 2738 // Delegate volume control to effect in track effect chain if needed 2739 // only one effect chain can be present on DirectOutputThread, so if 2740 // there is one, the track is connected to it 2741 if (!effectChains.isEmpty()) { 2742 // Do not ramp volume if volume is controlled by effect 2743 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2744 rampVolume = false; 2745 } 2746 } 2747 2748 // Convert volumes from 8.24 to 4.12 format 2749 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2750 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2751 leftVol = (uint16_t)v_clamped; 2752 v_clamped = (vr + (1 << 11)) >> 12; 2753 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2754 rightVol = (uint16_t)v_clamped; 2755 } else { 2756 leftVol = mLeftVolShort; 2757 rightVol = mRightVolShort; 2758 rampVolume = false; 2759 } 2760 2761 // reset retry count 2762 track->mRetryCount = kMaxTrackRetriesDirect; 2763 activeTrack = t; 2764 mixerStatus = MIXER_TRACKS_READY; 2765 } else { 2766 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2767 if (track->isStopped()) { 2768 track->reset(); 2769 } 2770 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2771 // We have consumed all the buffers of this track. 2772 // Remove it from the list of active tracks. 2773 trackToRemove = track; 2774 } else { 2775 // No buffers for this track. Give it a few chances to 2776 // fill a buffer, then remove it from active list. 2777 if (--(track->mRetryCount) <= 0) { 2778 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2779 trackToRemove = track; 2780 } else { 2781 mixerStatus = MIXER_TRACKS_ENABLED; 2782 } 2783 } 2784 } 2785 } 2786 2787 // remove all the tracks that need to be... 2788 if (CC_UNLIKELY(trackToRemove != 0)) { 2789 mActiveTracks.remove(trackToRemove); 2790 if (!effectChains.isEmpty()) { 2791 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2792 trackToRemove->sessionId()); 2793 effectChains[0]->decActiveTrackCnt(); 2794 } 2795 if (trackToRemove->isTerminated()) { 2796 removeTrack_l(trackToRemove); 2797 } 2798 } 2799 2800 lockEffectChains_l(effectChains); 2801 } 2802 2803 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2804 AudioBufferProvider::Buffer buffer; 2805 size_t frameCount = mFrameCount; 2806 curBuf = (int8_t *)mMixBuffer; 2807 // output audio to hardware 2808 while (frameCount) { 2809 buffer.frameCount = frameCount; 2810 activeTrack->getNextBuffer(&buffer); 2811 if (CC_UNLIKELY(buffer.raw == NULL)) { 2812 memset(curBuf, 0, frameCount * mFrameSize); 2813 break; 2814 } 2815 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2816 frameCount -= buffer.frameCount; 2817 curBuf += buffer.frameCount * mFrameSize; 2818 activeTrack->releaseBuffer(&buffer); 2819 } 2820 sleepTime = 0; 2821 standbyTime = systemTime() + standbyDelay; 2822 } else { 2823 if (sleepTime == 0) { 2824 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2825 sleepTime = activeSleepTime; 2826 } else { 2827 sleepTime = idleSleepTime; 2828 } 2829 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2830 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2831 sleepTime = 0; 2832 } 2833 } 2834 2835 if (mSuspended) { 2836 sleepTime = suspendSleepTimeUs(); 2837 } 2838 // sleepTime == 0 means we must write to audio hardware 2839 if (sleepTime == 0) { 2840 if (mixerStatus == MIXER_TRACKS_READY) { 2841 applyVolume(leftVol, rightVol, rampVolume); 2842 } 2843 for (size_t i = 0; i < effectChains.size(); i ++) { 2844 effectChains[i]->process_l(); 2845 } 2846 unlockEffectChains(effectChains); 2847 2848 mLastWriteTime = systemTime(); 2849 mInWrite = true; 2850 mBytesWritten += mixBufferSize; 2851 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2852 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2853 mNumWrites++; 2854 mInWrite = false; 2855 mStandby = false; 2856 } else { 2857 unlockEffectChains(effectChains); 2858 usleep(sleepTime); 2859 } 2860 2861 // finally let go of removed track, without the lock held 2862 // since we can't guarantee the destructors won't acquire that 2863 // same lock. 2864 trackToRemove.clear(); 2865 activeTrack.clear(); 2866 2867 // Effect chains will be actually deleted here if they were removed from 2868 // mEffectChains list during mixing or effects processing 2869 effectChains.clear(); 2870 } 2871 2872 if (!mStandby) { 2873 mOutput->stream->common.standby(&mOutput->stream->common); 2874 } 2875 2876 releaseWakeLock(); 2877 2878 ALOGV("DirectOutputThread %p exiting", this); 2879 return false; 2880} 2881 2882// getTrackName_l() must be called with ThreadBase::mLock held 2883int AudioFlinger::DirectOutputThread::getTrackName_l() 2884{ 2885 return 0; 2886} 2887 2888// deleteTrackName_l() must be called with ThreadBase::mLock held 2889void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2890{ 2891} 2892 2893// checkForNewParameters_l() must be called with ThreadBase::mLock held 2894bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2895{ 2896 bool reconfig = false; 2897 2898 while (!mNewParameters.isEmpty()) { 2899 status_t status = NO_ERROR; 2900 String8 keyValuePair = mNewParameters[0]; 2901 AudioParameter param = AudioParameter(keyValuePair); 2902 int value; 2903 2904 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2905 // do not accept frame count changes if tracks are open as the track buffer 2906 // size depends on frame count and correct behavior would not be garantied 2907 // if frame count is changed after track creation 2908 if (!mTracks.isEmpty()) { 2909 status = INVALID_OPERATION; 2910 } else { 2911 reconfig = true; 2912 } 2913 } 2914 if (status == NO_ERROR) { 2915 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2916 keyValuePair.string()); 2917 if (!mStandby && status == INVALID_OPERATION) { 2918 mOutput->stream->common.standby(&mOutput->stream->common); 2919 mStandby = true; 2920 mBytesWritten = 0; 2921 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2922 keyValuePair.string()); 2923 } 2924 if (status == NO_ERROR && reconfig) { 2925 readOutputParameters(); 2926 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2927 } 2928 } 2929 2930 mNewParameters.removeAt(0); 2931 2932 mParamStatus = status; 2933 mParamCond.signal(); 2934 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2935 // already timed out waiting for the status and will never signal the condition. 2936 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2937 } 2938 return reconfig; 2939} 2940 2941uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2942{ 2943 uint32_t time; 2944 if (audio_is_linear_pcm(mFormat)) { 2945 time = PlaybackThread::activeSleepTimeUs(); 2946 } else { 2947 time = 10000; 2948 } 2949 return time; 2950} 2951 2952uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2953{ 2954 uint32_t time; 2955 if (audio_is_linear_pcm(mFormat)) { 2956 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2957 } else { 2958 time = 10000; 2959 } 2960 return time; 2961} 2962 2963uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2964{ 2965 uint32_t time; 2966 if (audio_is_linear_pcm(mFormat)) { 2967 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2968 } else { 2969 time = 10000; 2970 } 2971 return time; 2972} 2973 2974 2975// ---------------------------------------------------------------------------- 2976 2977AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 2978 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 2979 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 2980 mWaitTimeMs(UINT_MAX) 2981{ 2982 addOutputTrack(mainThread); 2983} 2984 2985AudioFlinger::DuplicatingThread::~DuplicatingThread() 2986{ 2987 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2988 mOutputTracks[i]->destroy(); 2989 } 2990} 2991 2992bool AudioFlinger::DuplicatingThread::threadLoop() 2993{ 2994 Vector< sp<Track> > tracksToRemove; 2995 mixer_state mixerStatus = MIXER_IDLE; 2996 nsecs_t standbyTime = systemTime(); 2997 size_t mixBufferSize = mFrameCount*mFrameSize; 2998 SortedVector< sp<OutputTrack> > outputTracks; 2999 uint32_t writeFrames = 0; 3000 uint32_t activeSleepTime = activeSleepTimeUs(); 3001 uint32_t idleSleepTime = idleSleepTimeUs(); 3002 uint32_t sleepTime = idleSleepTime; 3003 Vector< sp<EffectChain> > effectChains; 3004 3005 acquireWakeLock(); 3006 3007 while (!exitPending()) 3008 { 3009 processConfigEvents(); 3010 3011 mixerStatus = MIXER_IDLE; 3012 { // scope for the mLock 3013 3014 Mutex::Autolock _l(mLock); 3015 3016 if (checkForNewParameters_l()) { 3017 mixBufferSize = mFrameCount*mFrameSize; 3018 updateWaitTime(); 3019 activeSleepTime = activeSleepTimeUs(); 3020 idleSleepTime = idleSleepTimeUs(); 3021 } 3022 3023 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3024 3025 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3026 outputTracks.add(mOutputTracks[i]); 3027 } 3028 3029 // put audio hardware into standby after short delay 3030 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3031 mSuspended)) { 3032 if (!mStandby) { 3033 for (size_t i = 0; i < outputTracks.size(); i++) { 3034 outputTracks[i]->stop(); 3035 } 3036 mStandby = true; 3037 mBytesWritten = 0; 3038 } 3039 3040 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3041 // we're about to wait, flush the binder command buffer 3042 IPCThreadState::self()->flushCommands(); 3043 outputTracks.clear(); 3044 3045 if (exitPending()) break; 3046 3047 releaseWakeLock_l(); 3048 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3049 mWaitWorkCV.wait(mLock); 3050 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3051 acquireWakeLock_l(); 3052 3053 mPrevMixerStatus = MIXER_IDLE; 3054 if (!mMasterMute) { 3055 char value[PROPERTY_VALUE_MAX]; 3056 property_get("ro.audio.silent", value, "0"); 3057 if (atoi(value)) { 3058 ALOGD("Silence is golden"); 3059 setMasterMute(true); 3060 } 3061 } 3062 3063 standbyTime = systemTime() + kStandbyTimeInNsecs; 3064 sleepTime = idleSleepTime; 3065 continue; 3066 } 3067 } 3068 3069 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3070 3071 // prevent any changes in effect chain list and in each effect chain 3072 // during mixing and effect process as the audio buffers could be deleted 3073 // or modified if an effect is created or deleted 3074 lockEffectChains_l(effectChains); 3075 } 3076 3077 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3078 // mix buffers... 3079 if (outputsReady(outputTracks)) { 3080 mAudioMixer->process(); 3081 } else { 3082 memset(mMixBuffer, 0, mixBufferSize); 3083 } 3084 sleepTime = 0; 3085 writeFrames = mFrameCount; 3086 } else { 3087 if (sleepTime == 0) { 3088 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3089 sleepTime = activeSleepTime; 3090 } else { 3091 sleepTime = idleSleepTime; 3092 } 3093 } else if (mBytesWritten != 0) { 3094 // flush remaining overflow buffers in output tracks 3095 for (size_t i = 0; i < outputTracks.size(); i++) { 3096 if (outputTracks[i]->isActive()) { 3097 sleepTime = 0; 3098 writeFrames = 0; 3099 memset(mMixBuffer, 0, mixBufferSize); 3100 break; 3101 } 3102 } 3103 } 3104 } 3105 3106 if (mSuspended) { 3107 sleepTime = suspendSleepTimeUs(); 3108 } 3109 // sleepTime == 0 means we must write to audio hardware 3110 if (sleepTime == 0) { 3111 for (size_t i = 0; i < effectChains.size(); i ++) { 3112 effectChains[i]->process_l(); 3113 } 3114 // enable changes in effect chain 3115 unlockEffectChains(effectChains); 3116 3117 standbyTime = systemTime() + kStandbyTimeInNsecs; 3118 for (size_t i = 0; i < outputTracks.size(); i++) { 3119 outputTracks[i]->write(mMixBuffer, writeFrames); 3120 } 3121 mStandby = false; 3122 mBytesWritten += mixBufferSize; 3123 } else { 3124 // enable changes in effect chain 3125 unlockEffectChains(effectChains); 3126 usleep(sleepTime); 3127 } 3128 3129 // finally let go of all our tracks, without the lock held 3130 // since we can't guarantee the destructors won't acquire that 3131 // same lock. 3132 tracksToRemove.clear(); 3133 outputTracks.clear(); 3134 3135 // Effect chains will be actually deleted here if they were removed from 3136 // mEffectChains list during mixing or effects processing 3137 effectChains.clear(); 3138 } 3139 3140 releaseWakeLock(); 3141 3142 return false; 3143} 3144 3145void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3146{ 3147 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3148 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3149 this, 3150 mSampleRate, 3151 mFormat, 3152 mChannelMask, 3153 frameCount); 3154 if (outputTrack->cblk() != NULL) { 3155 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3156 mOutputTracks.add(outputTrack); 3157 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3158 updateWaitTime(); 3159 } 3160} 3161 3162void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3163{ 3164 Mutex::Autolock _l(mLock); 3165 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3166 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3167 mOutputTracks[i]->destroy(); 3168 mOutputTracks.removeAt(i); 3169 updateWaitTime(); 3170 return; 3171 } 3172 } 3173 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3174} 3175 3176void AudioFlinger::DuplicatingThread::updateWaitTime() 3177{ 3178 mWaitTimeMs = UINT_MAX; 3179 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3180 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3181 if (strong != 0) { 3182 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3183 if (waitTimeMs < mWaitTimeMs) { 3184 mWaitTimeMs = waitTimeMs; 3185 } 3186 } 3187 } 3188} 3189 3190 3191bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3192{ 3193 for (size_t i = 0; i < outputTracks.size(); i++) { 3194 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3195 if (thread == 0) { 3196 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3197 return false; 3198 } 3199 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3200 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3201 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3202 return false; 3203 } 3204 } 3205 return true; 3206} 3207 3208uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3209{ 3210 return (mWaitTimeMs * 1000) / 2; 3211} 3212 3213// ---------------------------------------------------------------------------- 3214 3215// TrackBase constructor must be called with AudioFlinger::mLock held 3216AudioFlinger::ThreadBase::TrackBase::TrackBase( 3217 const wp<ThreadBase>& thread, 3218 const sp<Client>& client, 3219 uint32_t sampleRate, 3220 audio_format_t format, 3221 uint32_t channelMask, 3222 int frameCount, 3223 uint32_t flags, 3224 const sp<IMemory>& sharedBuffer, 3225 int sessionId) 3226 : RefBase(), 3227 mThread(thread), 3228 mClient(client), 3229 mCblk(NULL), 3230 // mBuffer 3231 // mBufferEnd 3232 mFrameCount(0), 3233 mState(IDLE), 3234 mFormat(format), 3235 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3236 mSessionId(sessionId) 3237 // mChannelCount 3238 // mChannelMask 3239{ 3240 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3241 3242 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3243 size_t size = sizeof(audio_track_cblk_t); 3244 uint8_t channelCount = popcount(channelMask); 3245 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3246 if (sharedBuffer == 0) { 3247 size += bufferSize; 3248 } 3249 3250 if (client != NULL) { 3251 mCblkMemory = client->heap()->allocate(size); 3252 if (mCblkMemory != 0) { 3253 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3254 if (mCblk != NULL) { // construct the shared structure in-place. 3255 new(mCblk) audio_track_cblk_t(); 3256 // clear all buffers 3257 mCblk->frameCount = frameCount; 3258 mCblk->sampleRate = sampleRate; 3259 mChannelCount = channelCount; 3260 mChannelMask = channelMask; 3261 if (sharedBuffer == 0) { 3262 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3263 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3264 // Force underrun condition to avoid false underrun callback until first data is 3265 // written to buffer (other flags are cleared) 3266 mCblk->flags = CBLK_UNDERRUN_ON; 3267 } else { 3268 mBuffer = sharedBuffer->pointer(); 3269 } 3270 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3271 } 3272 } else { 3273 ALOGE("not enough memory for AudioTrack size=%u", size); 3274 client->heap()->dump("AudioTrack"); 3275 return; 3276 } 3277 } else { 3278 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3279 // construct the shared structure in-place. 3280 new(mCblk) audio_track_cblk_t(); 3281 // clear all buffers 3282 mCblk->frameCount = frameCount; 3283 mCblk->sampleRate = sampleRate; 3284 mChannelCount = channelCount; 3285 mChannelMask = channelMask; 3286 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3287 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3288 // Force underrun condition to avoid false underrun callback until first data is 3289 // written to buffer (other flags are cleared) 3290 mCblk->flags = CBLK_UNDERRUN_ON; 3291 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3292 } 3293} 3294 3295AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3296{ 3297 if (mCblk != NULL) { 3298 if (mClient == 0) { 3299 delete mCblk; 3300 } else { 3301 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3302 } 3303 } 3304 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3305 if (mClient != 0) { 3306 // Client destructor must run with AudioFlinger mutex locked 3307 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3308 // If the client's reference count drops to zero, the associated destructor 3309 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3310 // relying on the automatic clear() at end of scope. 3311 mClient.clear(); 3312 } 3313} 3314 3315void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3316{ 3317 buffer->raw = NULL; 3318 mFrameCount = buffer->frameCount; 3319 step(); 3320 buffer->frameCount = 0; 3321} 3322 3323bool AudioFlinger::ThreadBase::TrackBase::step() { 3324 bool result; 3325 audio_track_cblk_t* cblk = this->cblk(); 3326 3327 result = cblk->stepServer(mFrameCount); 3328 if (!result) { 3329 ALOGV("stepServer failed acquiring cblk mutex"); 3330 mFlags |= STEPSERVER_FAILED; 3331 } 3332 return result; 3333} 3334 3335void AudioFlinger::ThreadBase::TrackBase::reset() { 3336 audio_track_cblk_t* cblk = this->cblk(); 3337 3338 cblk->user = 0; 3339 cblk->server = 0; 3340 cblk->userBase = 0; 3341 cblk->serverBase = 0; 3342 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3343 ALOGV("TrackBase::reset"); 3344} 3345 3346int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3347 return (int)mCblk->sampleRate; 3348} 3349 3350void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3351 audio_track_cblk_t* cblk = this->cblk(); 3352 size_t frameSize = cblk->frameSize; 3353 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3354 int8_t *bufferEnd = bufferStart + frames * frameSize; 3355 3356 // Check validity of returned pointer in case the track control block would have been corrupted. 3357 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3358 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3359 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3360 server %d, serverBase %d, user %d, userBase %d", 3361 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3362 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3363 return NULL; 3364 } 3365 3366 return bufferStart; 3367} 3368 3369// ---------------------------------------------------------------------------- 3370 3371// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3372AudioFlinger::PlaybackThread::Track::Track( 3373 const wp<ThreadBase>& thread, 3374 const sp<Client>& client, 3375 audio_stream_type_t streamType, 3376 uint32_t sampleRate, 3377 audio_format_t format, 3378 uint32_t channelMask, 3379 int frameCount, 3380 const sp<IMemory>& sharedBuffer, 3381 int sessionId) 3382 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3383 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3384 mAuxEffectId(0), mHasVolumeController(false) 3385{ 3386 if (mCblk != NULL) { 3387 sp<ThreadBase> baseThread = thread.promote(); 3388 if (baseThread != 0) { 3389 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3390 mName = playbackThread->getTrackName_l(); 3391 mMainBuffer = playbackThread->mixBuffer(); 3392 } 3393 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3394 if (mName < 0) { 3395 ALOGE("no more track names available"); 3396 } 3397 mStreamType = streamType; 3398 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3399 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3400 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3401 } 3402} 3403 3404AudioFlinger::PlaybackThread::Track::~Track() 3405{ 3406 ALOGV("PlaybackThread::Track destructor"); 3407 sp<ThreadBase> thread = mThread.promote(); 3408 if (thread != 0) { 3409 Mutex::Autolock _l(thread->mLock); 3410 mState = TERMINATED; 3411 } 3412} 3413 3414void AudioFlinger::PlaybackThread::Track::destroy() 3415{ 3416 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3417 // by removing it from mTracks vector, so there is a risk that this Tracks's 3418 // desctructor is called. As the destructor needs to lock mLock, 3419 // we must acquire a strong reference on this Track before locking mLock 3420 // here so that the destructor is called only when exiting this function. 3421 // On the other hand, as long as Track::destroy() is only called by 3422 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3423 // this Track with its member mTrack. 3424 sp<Track> keep(this); 3425 { // scope for mLock 3426 sp<ThreadBase> thread = mThread.promote(); 3427 if (thread != 0) { 3428 if (!isOutputTrack()) { 3429 if (mState == ACTIVE || mState == RESUMING) { 3430 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3431 3432 // to track the speaker usage 3433 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3434 } 3435 AudioSystem::releaseOutput(thread->id()); 3436 } 3437 Mutex::Autolock _l(thread->mLock); 3438 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3439 playbackThread->destroyTrack_l(this); 3440 } 3441 } 3442} 3443 3444void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3445{ 3446 uint32_t vlr = mCblk->getVolumeLR(); 3447 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", 3448 mName - AudioMixer::TRACK0, 3449 (mClient == 0) ? getpid() : mClient->pid(), 3450 mStreamType, 3451 mFormat, 3452 mChannelMask, 3453 mSessionId, 3454 mFrameCount, 3455 mState, 3456 mMute, 3457 mFillingUpStatus, 3458 mCblk->sampleRate, 3459 vlr & 0xFFFF, 3460 vlr >> 16, 3461 mCblk->server, 3462 mCblk->user, 3463 (int)mMainBuffer, 3464 (int)mAuxBuffer); 3465} 3466 3467status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3468{ 3469 audio_track_cblk_t* cblk = this->cblk(); 3470 uint32_t framesReady; 3471 uint32_t framesReq = buffer->frameCount; 3472 3473 // Check if last stepServer failed, try to step now 3474 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3475 if (!step()) goto getNextBuffer_exit; 3476 ALOGV("stepServer recovered"); 3477 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3478 } 3479 3480 framesReady = cblk->framesReady(); 3481 3482 if (CC_LIKELY(framesReady)) { 3483 uint32_t s = cblk->server; 3484 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3485 3486 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3487 if (framesReq > framesReady) { 3488 framesReq = framesReady; 3489 } 3490 if (s + framesReq > bufferEnd) { 3491 framesReq = bufferEnd - s; 3492 } 3493 3494 buffer->raw = getBuffer(s, framesReq); 3495 if (buffer->raw == NULL) goto getNextBuffer_exit; 3496 3497 buffer->frameCount = framesReq; 3498 return NO_ERROR; 3499 } 3500 3501getNextBuffer_exit: 3502 buffer->raw = NULL; 3503 buffer->frameCount = 0; 3504 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3505 return NOT_ENOUGH_DATA; 3506} 3507 3508bool AudioFlinger::PlaybackThread::Track::isReady() const { 3509 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3510 3511 if (mCblk->framesReady() >= mCblk->frameCount || 3512 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3513 mFillingUpStatus = FS_FILLED; 3514 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3515 return true; 3516 } 3517 return false; 3518} 3519 3520status_t AudioFlinger::PlaybackThread::Track::start() 3521{ 3522 status_t status = NO_ERROR; 3523 ALOGV("start(%d), calling pid %d session %d", 3524 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3525 sp<ThreadBase> thread = mThread.promote(); 3526 if (thread != 0) { 3527 Mutex::Autolock _l(thread->mLock); 3528 track_state state = mState; 3529 // here the track could be either new, or restarted 3530 // in both cases "unstop" the track 3531 if (mState == PAUSED) { 3532 mState = TrackBase::RESUMING; 3533 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3534 } else { 3535 mState = TrackBase::ACTIVE; 3536 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3537 } 3538 3539 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3540 thread->mLock.unlock(); 3541 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3542 thread->mLock.lock(); 3543 3544 // to track the speaker usage 3545 if (status == NO_ERROR) { 3546 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3547 } 3548 } 3549 if (status == NO_ERROR) { 3550 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3551 playbackThread->addTrack_l(this); 3552 } else { 3553 mState = state; 3554 } 3555 } else { 3556 status = BAD_VALUE; 3557 } 3558 return status; 3559} 3560 3561void AudioFlinger::PlaybackThread::Track::stop() 3562{ 3563 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3564 sp<ThreadBase> thread = mThread.promote(); 3565 if (thread != 0) { 3566 Mutex::Autolock _l(thread->mLock); 3567 track_state state = mState; 3568 if (mState > STOPPED) { 3569 mState = STOPPED; 3570 // If the track is not active (PAUSED and buffers full), flush buffers 3571 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3572 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3573 reset(); 3574 } 3575 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3576 } 3577 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3578 thread->mLock.unlock(); 3579 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3580 thread->mLock.lock(); 3581 3582 // to track the speaker usage 3583 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3584 } 3585 } 3586} 3587 3588void AudioFlinger::PlaybackThread::Track::pause() 3589{ 3590 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3591 sp<ThreadBase> thread = mThread.promote(); 3592 if (thread != 0) { 3593 Mutex::Autolock _l(thread->mLock); 3594 if (mState == ACTIVE || mState == RESUMING) { 3595 mState = PAUSING; 3596 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3597 if (!isOutputTrack()) { 3598 thread->mLock.unlock(); 3599 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3600 thread->mLock.lock(); 3601 3602 // to track the speaker usage 3603 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3604 } 3605 } 3606 } 3607} 3608 3609void AudioFlinger::PlaybackThread::Track::flush() 3610{ 3611 ALOGV("flush(%d)", mName); 3612 sp<ThreadBase> thread = mThread.promote(); 3613 if (thread != 0) { 3614 Mutex::Autolock _l(thread->mLock); 3615 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3616 return; 3617 } 3618 // No point remaining in PAUSED state after a flush => go to 3619 // STOPPED state 3620 mState = STOPPED; 3621 3622 // do not reset the track if it is still in the process of being stopped or paused. 3623 // this will be done by prepareTracks_l() when the track is stopped. 3624 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3625 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3626 reset(); 3627 } 3628 } 3629} 3630 3631void AudioFlinger::PlaybackThread::Track::reset() 3632{ 3633 // Do not reset twice to avoid discarding data written just after a flush and before 3634 // the audioflinger thread detects the track is stopped. 3635 if (!mResetDone) { 3636 TrackBase::reset(); 3637 // Force underrun condition to avoid false underrun callback until first data is 3638 // written to buffer 3639 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3640 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3641 mFillingUpStatus = FS_FILLING; 3642 mResetDone = true; 3643 } 3644} 3645 3646void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3647{ 3648 mMute = muted; 3649} 3650 3651status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3652{ 3653 status_t status = DEAD_OBJECT; 3654 sp<ThreadBase> thread = mThread.promote(); 3655 if (thread != 0) { 3656 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3657 status = playbackThread->attachAuxEffect(this, EffectId); 3658 } 3659 return status; 3660} 3661 3662void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3663{ 3664 mAuxEffectId = EffectId; 3665 mAuxBuffer = buffer; 3666} 3667 3668// ---------------------------------------------------------------------------- 3669 3670// RecordTrack constructor must be called with AudioFlinger::mLock held 3671AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3672 const wp<ThreadBase>& thread, 3673 const sp<Client>& client, 3674 uint32_t sampleRate, 3675 audio_format_t format, 3676 uint32_t channelMask, 3677 int frameCount, 3678 uint32_t flags, 3679 int sessionId) 3680 : TrackBase(thread, client, sampleRate, format, 3681 channelMask, frameCount, flags, 0, sessionId), 3682 mOverflow(false) 3683{ 3684 if (mCblk != NULL) { 3685 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3686 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3687 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3688 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3689 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3690 } else { 3691 mCblk->frameSize = sizeof(int8_t); 3692 } 3693 } 3694} 3695 3696AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3697{ 3698 sp<ThreadBase> thread = mThread.promote(); 3699 if (thread != 0) { 3700 AudioSystem::releaseInput(thread->id()); 3701 } 3702} 3703 3704status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3705{ 3706 audio_track_cblk_t* cblk = this->cblk(); 3707 uint32_t framesAvail; 3708 uint32_t framesReq = buffer->frameCount; 3709 3710 // Check if last stepServer failed, try to step now 3711 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3712 if (!step()) goto getNextBuffer_exit; 3713 ALOGV("stepServer recovered"); 3714 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3715 } 3716 3717 framesAvail = cblk->framesAvailable_l(); 3718 3719 if (CC_LIKELY(framesAvail)) { 3720 uint32_t s = cblk->server; 3721 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3722 3723 if (framesReq > framesAvail) { 3724 framesReq = framesAvail; 3725 } 3726 if (s + framesReq > bufferEnd) { 3727 framesReq = bufferEnd - s; 3728 } 3729 3730 buffer->raw = getBuffer(s, framesReq); 3731 if (buffer->raw == NULL) goto getNextBuffer_exit; 3732 3733 buffer->frameCount = framesReq; 3734 return NO_ERROR; 3735 } 3736 3737getNextBuffer_exit: 3738 buffer->raw = NULL; 3739 buffer->frameCount = 0; 3740 return NOT_ENOUGH_DATA; 3741} 3742 3743status_t AudioFlinger::RecordThread::RecordTrack::start() 3744{ 3745 sp<ThreadBase> thread = mThread.promote(); 3746 if (thread != 0) { 3747 RecordThread *recordThread = (RecordThread *)thread.get(); 3748 return recordThread->start(this); 3749 } else { 3750 return BAD_VALUE; 3751 } 3752} 3753 3754void AudioFlinger::RecordThread::RecordTrack::stop() 3755{ 3756 sp<ThreadBase> thread = mThread.promote(); 3757 if (thread != 0) { 3758 RecordThread *recordThread = (RecordThread *)thread.get(); 3759 recordThread->stop(this); 3760 TrackBase::reset(); 3761 // Force overerrun condition to avoid false overrun callback until first data is 3762 // read from buffer 3763 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3764 } 3765} 3766 3767void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3768{ 3769 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3770 (mClient == 0) ? getpid() : mClient->pid(), 3771 mFormat, 3772 mChannelMask, 3773 mSessionId, 3774 mFrameCount, 3775 mState, 3776 mCblk->sampleRate, 3777 mCblk->server, 3778 mCblk->user); 3779} 3780 3781 3782// ---------------------------------------------------------------------------- 3783 3784AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3785 const wp<ThreadBase>& thread, 3786 DuplicatingThread *sourceThread, 3787 uint32_t sampleRate, 3788 audio_format_t format, 3789 uint32_t channelMask, 3790 int frameCount) 3791 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3792 mActive(false), mSourceThread(sourceThread) 3793{ 3794 3795 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3796 if (mCblk != NULL) { 3797 mCblk->flags |= CBLK_DIRECTION_OUT; 3798 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3799 mOutBuffer.frameCount = 0; 3800 playbackThread->mTracks.add(this); 3801 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3802 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3803 mCblk, mBuffer, mCblk->buffers, 3804 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3805 } else { 3806 ALOGW("Error creating output track on thread %p", playbackThread); 3807 } 3808} 3809 3810AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3811{ 3812 clearBufferQueue(); 3813} 3814 3815status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3816{ 3817 status_t status = Track::start(); 3818 if (status != NO_ERROR) { 3819 return status; 3820 } 3821 3822 mActive = true; 3823 mRetryCount = 127; 3824 return status; 3825} 3826 3827void AudioFlinger::PlaybackThread::OutputTrack::stop() 3828{ 3829 Track::stop(); 3830 clearBufferQueue(); 3831 mOutBuffer.frameCount = 0; 3832 mActive = false; 3833} 3834 3835bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3836{ 3837 Buffer *pInBuffer; 3838 Buffer inBuffer; 3839 uint32_t channelCount = mChannelCount; 3840 bool outputBufferFull = false; 3841 inBuffer.frameCount = frames; 3842 inBuffer.i16 = data; 3843 3844 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3845 3846 if (!mActive && frames != 0) { 3847 start(); 3848 sp<ThreadBase> thread = mThread.promote(); 3849 if (thread != 0) { 3850 MixerThread *mixerThread = (MixerThread *)thread.get(); 3851 if (mCblk->frameCount > frames){ 3852 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3853 uint32_t startFrames = (mCblk->frameCount - frames); 3854 pInBuffer = new Buffer; 3855 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3856 pInBuffer->frameCount = startFrames; 3857 pInBuffer->i16 = pInBuffer->mBuffer; 3858 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3859 mBufferQueue.add(pInBuffer); 3860 } else { 3861 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3862 } 3863 } 3864 } 3865 } 3866 3867 while (waitTimeLeftMs) { 3868 // First write pending buffers, then new data 3869 if (mBufferQueue.size()) { 3870 pInBuffer = mBufferQueue.itemAt(0); 3871 } else { 3872 pInBuffer = &inBuffer; 3873 } 3874 3875 if (pInBuffer->frameCount == 0) { 3876 break; 3877 } 3878 3879 if (mOutBuffer.frameCount == 0) { 3880 mOutBuffer.frameCount = pInBuffer->frameCount; 3881 nsecs_t startTime = systemTime(); 3882 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3883 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3884 outputBufferFull = true; 3885 break; 3886 } 3887 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3888 if (waitTimeLeftMs >= waitTimeMs) { 3889 waitTimeLeftMs -= waitTimeMs; 3890 } else { 3891 waitTimeLeftMs = 0; 3892 } 3893 } 3894 3895 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3896 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3897 mCblk->stepUser(outFrames); 3898 pInBuffer->frameCount -= outFrames; 3899 pInBuffer->i16 += outFrames * channelCount; 3900 mOutBuffer.frameCount -= outFrames; 3901 mOutBuffer.i16 += outFrames * channelCount; 3902 3903 if (pInBuffer->frameCount == 0) { 3904 if (mBufferQueue.size()) { 3905 mBufferQueue.removeAt(0); 3906 delete [] pInBuffer->mBuffer; 3907 delete pInBuffer; 3908 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3909 } else { 3910 break; 3911 } 3912 } 3913 } 3914 3915 // If we could not write all frames, allocate a buffer and queue it for next time. 3916 if (inBuffer.frameCount) { 3917 sp<ThreadBase> thread = mThread.promote(); 3918 if (thread != 0 && !thread->standby()) { 3919 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3920 pInBuffer = new Buffer; 3921 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3922 pInBuffer->frameCount = inBuffer.frameCount; 3923 pInBuffer->i16 = pInBuffer->mBuffer; 3924 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3925 mBufferQueue.add(pInBuffer); 3926 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3927 } else { 3928 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3929 } 3930 } 3931 } 3932 3933 // Calling write() with a 0 length buffer, means that no more data will be written: 3934 // If no more buffers are pending, fill output track buffer to make sure it is started 3935 // by output mixer. 3936 if (frames == 0 && mBufferQueue.size() == 0) { 3937 if (mCblk->user < mCblk->frameCount) { 3938 frames = mCblk->frameCount - mCblk->user; 3939 pInBuffer = new Buffer; 3940 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3941 pInBuffer->frameCount = frames; 3942 pInBuffer->i16 = pInBuffer->mBuffer; 3943 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3944 mBufferQueue.add(pInBuffer); 3945 } else if (mActive) { 3946 stop(); 3947 } 3948 } 3949 3950 return outputBufferFull; 3951} 3952 3953status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3954{ 3955 int active; 3956 status_t result; 3957 audio_track_cblk_t* cblk = mCblk; 3958 uint32_t framesReq = buffer->frameCount; 3959 3960// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3961 buffer->frameCount = 0; 3962 3963 uint32_t framesAvail = cblk->framesAvailable(); 3964 3965 3966 if (framesAvail == 0) { 3967 Mutex::Autolock _l(cblk->lock); 3968 goto start_loop_here; 3969 while (framesAvail == 0) { 3970 active = mActive; 3971 if (CC_UNLIKELY(!active)) { 3972 ALOGV("Not active and NO_MORE_BUFFERS"); 3973 return NO_MORE_BUFFERS; 3974 } 3975 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3976 if (result != NO_ERROR) { 3977 return NO_MORE_BUFFERS; 3978 } 3979 // read the server count again 3980 start_loop_here: 3981 framesAvail = cblk->framesAvailable_l(); 3982 } 3983 } 3984 3985// if (framesAvail < framesReq) { 3986// return NO_MORE_BUFFERS; 3987// } 3988 3989 if (framesReq > framesAvail) { 3990 framesReq = framesAvail; 3991 } 3992 3993 uint32_t u = cblk->user; 3994 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3995 3996 if (u + framesReq > bufferEnd) { 3997 framesReq = bufferEnd - u; 3998 } 3999 4000 buffer->frameCount = framesReq; 4001 buffer->raw = (void *)cblk->buffer(u); 4002 return NO_ERROR; 4003} 4004 4005 4006void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4007{ 4008 size_t size = mBufferQueue.size(); 4009 Buffer *pBuffer; 4010 4011 for (size_t i = 0; i < size; i++) { 4012 pBuffer = mBufferQueue.itemAt(i); 4013 delete [] pBuffer->mBuffer; 4014 delete pBuffer; 4015 } 4016 mBufferQueue.clear(); 4017} 4018 4019// ---------------------------------------------------------------------------- 4020 4021AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4022 : RefBase(), 4023 mAudioFlinger(audioFlinger), 4024 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4025 mPid(pid) 4026{ 4027 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4028} 4029 4030// Client destructor must be called with AudioFlinger::mLock held 4031AudioFlinger::Client::~Client() 4032{ 4033 mAudioFlinger->removeClient_l(mPid); 4034} 4035 4036sp<MemoryDealer> AudioFlinger::Client::heap() const 4037{ 4038 return mMemoryDealer; 4039} 4040 4041// ---------------------------------------------------------------------------- 4042 4043AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4044 const sp<IAudioFlingerClient>& client, 4045 pid_t pid) 4046 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4047{ 4048} 4049 4050AudioFlinger::NotificationClient::~NotificationClient() 4051{ 4052} 4053 4054void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4055{ 4056 sp<NotificationClient> keep(this); 4057 { 4058 mAudioFlinger->removeNotificationClient(mPid); 4059 } 4060} 4061 4062// ---------------------------------------------------------------------------- 4063 4064AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4065 : BnAudioTrack(), 4066 mTrack(track) 4067{ 4068} 4069 4070AudioFlinger::TrackHandle::~TrackHandle() { 4071 // just stop the track on deletion, associated resources 4072 // will be freed from the main thread once all pending buffers have 4073 // been played. Unless it's not in the active track list, in which 4074 // case we free everything now... 4075 mTrack->destroy(); 4076} 4077 4078sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4079 return mTrack->getCblk(); 4080} 4081 4082status_t AudioFlinger::TrackHandle::start() { 4083 return mTrack->start(); 4084} 4085 4086void AudioFlinger::TrackHandle::stop() { 4087 mTrack->stop(); 4088} 4089 4090void AudioFlinger::TrackHandle::flush() { 4091 mTrack->flush(); 4092} 4093 4094void AudioFlinger::TrackHandle::mute(bool e) { 4095 mTrack->mute(e); 4096} 4097 4098void AudioFlinger::TrackHandle::pause() { 4099 mTrack->pause(); 4100} 4101 4102status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4103{ 4104 return mTrack->attachAuxEffect(EffectId); 4105} 4106 4107status_t AudioFlinger::TrackHandle::onTransact( 4108 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4109{ 4110 return BnAudioTrack::onTransact(code, data, reply, flags); 4111} 4112 4113// ---------------------------------------------------------------------------- 4114 4115sp<IAudioRecord> AudioFlinger::openRecord( 4116 pid_t pid, 4117 audio_io_handle_t input, 4118 uint32_t sampleRate, 4119 audio_format_t format, 4120 uint32_t channelMask, 4121 int frameCount, 4122 uint32_t flags, 4123 int *sessionId, 4124 status_t *status) 4125{ 4126 sp<RecordThread::RecordTrack> recordTrack; 4127 sp<RecordHandle> recordHandle; 4128 sp<Client> client; 4129 status_t lStatus; 4130 RecordThread *thread; 4131 size_t inFrameCount; 4132 int lSessionId; 4133 4134 // check calling permissions 4135 if (!recordingAllowed()) { 4136 lStatus = PERMISSION_DENIED; 4137 goto Exit; 4138 } 4139 4140 // add client to list 4141 { // scope for mLock 4142 Mutex::Autolock _l(mLock); 4143 thread = checkRecordThread_l(input); 4144 if (thread == NULL) { 4145 lStatus = BAD_VALUE; 4146 goto Exit; 4147 } 4148 4149 client = registerPid_l(pid); 4150 4151 // If no audio session id is provided, create one here 4152 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4153 lSessionId = *sessionId; 4154 } else { 4155 lSessionId = nextUniqueId(); 4156 if (sessionId != NULL) { 4157 *sessionId = lSessionId; 4158 } 4159 } 4160 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4161 recordTrack = thread->createRecordTrack_l(client, 4162 sampleRate, 4163 format, 4164 channelMask, 4165 frameCount, 4166 flags, 4167 lSessionId, 4168 &lStatus); 4169 } 4170 if (lStatus != NO_ERROR) { 4171 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4172 // destructor is called by the TrackBase destructor with mLock held 4173 client.clear(); 4174 recordTrack.clear(); 4175 goto Exit; 4176 } 4177 4178 // return to handle to client 4179 recordHandle = new RecordHandle(recordTrack); 4180 lStatus = NO_ERROR; 4181 4182Exit: 4183 if (status) { 4184 *status = lStatus; 4185 } 4186 return recordHandle; 4187} 4188 4189// ---------------------------------------------------------------------------- 4190 4191AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4192 : BnAudioRecord(), 4193 mRecordTrack(recordTrack) 4194{ 4195} 4196 4197AudioFlinger::RecordHandle::~RecordHandle() { 4198 stop(); 4199} 4200 4201sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4202 return mRecordTrack->getCblk(); 4203} 4204 4205status_t AudioFlinger::RecordHandle::start() { 4206 ALOGV("RecordHandle::start()"); 4207 return mRecordTrack->start(); 4208} 4209 4210void AudioFlinger::RecordHandle::stop() { 4211 ALOGV("RecordHandle::stop()"); 4212 mRecordTrack->stop(); 4213} 4214 4215status_t AudioFlinger::RecordHandle::onTransact( 4216 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4217{ 4218 return BnAudioRecord::onTransact(code, data, reply, flags); 4219} 4220 4221// ---------------------------------------------------------------------------- 4222 4223AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4224 AudioStreamIn *input, 4225 uint32_t sampleRate, 4226 uint32_t channels, 4227 audio_io_handle_t id, 4228 uint32_t device) : 4229 ThreadBase(audioFlinger, id, device, RECORD), 4230 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4231 // mRsmpInIndex and mInputBytes set by readInputParameters() 4232 mReqChannelCount(popcount(channels)), 4233 mReqSampleRate(sampleRate) 4234 // mBytesRead is only meaningful while active, and so is cleared in start() 4235 // (but might be better to also clear here for dump?) 4236{ 4237 snprintf(mName, kNameLength, "AudioIn_%d", id); 4238 4239 readInputParameters(); 4240} 4241 4242 4243AudioFlinger::RecordThread::~RecordThread() 4244{ 4245 delete[] mRsmpInBuffer; 4246 delete mResampler; 4247 delete[] mRsmpOutBuffer; 4248} 4249 4250void AudioFlinger::RecordThread::onFirstRef() 4251{ 4252 run(mName, PRIORITY_URGENT_AUDIO); 4253} 4254 4255status_t AudioFlinger::RecordThread::readyToRun() 4256{ 4257 status_t status = initCheck(); 4258 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4259 return status; 4260} 4261 4262bool AudioFlinger::RecordThread::threadLoop() 4263{ 4264 AudioBufferProvider::Buffer buffer; 4265 sp<RecordTrack> activeTrack; 4266 Vector< sp<EffectChain> > effectChains; 4267 4268 nsecs_t lastWarning = 0; 4269 4270 acquireWakeLock(); 4271 4272 // start recording 4273 while (!exitPending()) { 4274 4275 processConfigEvents(); 4276 4277 { // scope for mLock 4278 Mutex::Autolock _l(mLock); 4279 checkForNewParameters_l(); 4280 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4281 if (!mStandby) { 4282 mInput->stream->common.standby(&mInput->stream->common); 4283 mStandby = true; 4284 } 4285 4286 if (exitPending()) break; 4287 4288 releaseWakeLock_l(); 4289 ALOGV("RecordThread: loop stopping"); 4290 // go to sleep 4291 mWaitWorkCV.wait(mLock); 4292 ALOGV("RecordThread: loop starting"); 4293 acquireWakeLock_l(); 4294 continue; 4295 } 4296 if (mActiveTrack != 0) { 4297 if (mActiveTrack->mState == TrackBase::PAUSING) { 4298 if (!mStandby) { 4299 mInput->stream->common.standby(&mInput->stream->common); 4300 mStandby = true; 4301 } 4302 mActiveTrack.clear(); 4303 mStartStopCond.broadcast(); 4304 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4305 if (mReqChannelCount != mActiveTrack->channelCount()) { 4306 mActiveTrack.clear(); 4307 mStartStopCond.broadcast(); 4308 } else if (mBytesRead != 0) { 4309 // record start succeeds only if first read from audio input 4310 // succeeds 4311 if (mBytesRead > 0) { 4312 mActiveTrack->mState = TrackBase::ACTIVE; 4313 } else { 4314 mActiveTrack.clear(); 4315 } 4316 mStartStopCond.broadcast(); 4317 } 4318 mStandby = false; 4319 } 4320 } 4321 lockEffectChains_l(effectChains); 4322 } 4323 4324 if (mActiveTrack != 0) { 4325 if (mActiveTrack->mState != TrackBase::ACTIVE && 4326 mActiveTrack->mState != TrackBase::RESUMING) { 4327 unlockEffectChains(effectChains); 4328 usleep(kRecordThreadSleepUs); 4329 continue; 4330 } 4331 for (size_t i = 0; i < effectChains.size(); i ++) { 4332 effectChains[i]->process_l(); 4333 } 4334 4335 buffer.frameCount = mFrameCount; 4336 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4337 size_t framesOut = buffer.frameCount; 4338 if (mResampler == NULL) { 4339 // no resampling 4340 while (framesOut) { 4341 size_t framesIn = mFrameCount - mRsmpInIndex; 4342 if (framesIn) { 4343 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4344 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4345 if (framesIn > framesOut) 4346 framesIn = framesOut; 4347 mRsmpInIndex += framesIn; 4348 framesOut -= framesIn; 4349 if ((int)mChannelCount == mReqChannelCount || 4350 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4351 memcpy(dst, src, framesIn * mFrameSize); 4352 } else { 4353 int16_t *src16 = (int16_t *)src; 4354 int16_t *dst16 = (int16_t *)dst; 4355 if (mChannelCount == 1) { 4356 while (framesIn--) { 4357 *dst16++ = *src16; 4358 *dst16++ = *src16++; 4359 } 4360 } else { 4361 while (framesIn--) { 4362 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4363 src16 += 2; 4364 } 4365 } 4366 } 4367 } 4368 if (framesOut && mFrameCount == mRsmpInIndex) { 4369 if (framesOut == mFrameCount && 4370 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4371 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4372 framesOut = 0; 4373 } else { 4374 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4375 mRsmpInIndex = 0; 4376 } 4377 if (mBytesRead < 0) { 4378 ALOGE("Error reading audio input"); 4379 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4380 // Force input into standby so that it tries to 4381 // recover at next read attempt 4382 mInput->stream->common.standby(&mInput->stream->common); 4383 usleep(kRecordThreadSleepUs); 4384 } 4385 mRsmpInIndex = mFrameCount; 4386 framesOut = 0; 4387 buffer.frameCount = 0; 4388 } 4389 } 4390 } 4391 } else { 4392 // resampling 4393 4394 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4395 // alter output frame count as if we were expecting stereo samples 4396 if (mChannelCount == 1 && mReqChannelCount == 1) { 4397 framesOut >>= 1; 4398 } 4399 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4400 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4401 // are 32 bit aligned which should be always true. 4402 if (mChannelCount == 2 && mReqChannelCount == 1) { 4403 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4404 // the resampler always outputs stereo samples: do post stereo to mono conversion 4405 int16_t *src = (int16_t *)mRsmpOutBuffer; 4406 int16_t *dst = buffer.i16; 4407 while (framesOut--) { 4408 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4409 src += 2; 4410 } 4411 } else { 4412 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4413 } 4414 4415 } 4416 mActiveTrack->releaseBuffer(&buffer); 4417 mActiveTrack->overflow(); 4418 } 4419 // client isn't retrieving buffers fast enough 4420 else { 4421 if (!mActiveTrack->setOverflow()) { 4422 nsecs_t now = systemTime(); 4423 if ((now - lastWarning) > kWarningThrottleNs) { 4424 ALOGW("RecordThread: buffer overflow"); 4425 lastWarning = now; 4426 } 4427 } 4428 // Release the processor for a while before asking for a new buffer. 4429 // This will give the application more chance to read from the buffer and 4430 // clear the overflow. 4431 usleep(kRecordThreadSleepUs); 4432 } 4433 } 4434 // enable changes in effect chain 4435 unlockEffectChains(effectChains); 4436 effectChains.clear(); 4437 } 4438 4439 if (!mStandby) { 4440 mInput->stream->common.standby(&mInput->stream->common); 4441 } 4442 mActiveTrack.clear(); 4443 4444 mStartStopCond.broadcast(); 4445 4446 releaseWakeLock(); 4447 4448 ALOGV("RecordThread %p exiting", this); 4449 return false; 4450} 4451 4452 4453sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4454 const sp<AudioFlinger::Client>& client, 4455 uint32_t sampleRate, 4456 audio_format_t format, 4457 int channelMask, 4458 int frameCount, 4459 uint32_t flags, 4460 int sessionId, 4461 status_t *status) 4462{ 4463 sp<RecordTrack> track; 4464 status_t lStatus; 4465 4466 lStatus = initCheck(); 4467 if (lStatus != NO_ERROR) { 4468 ALOGE("Audio driver not initialized."); 4469 goto Exit; 4470 } 4471 4472 { // scope for mLock 4473 Mutex::Autolock _l(mLock); 4474 4475 track = new RecordTrack(this, client, sampleRate, 4476 format, channelMask, frameCount, flags, sessionId); 4477 4478 if (track->getCblk() == 0) { 4479 lStatus = NO_MEMORY; 4480 goto Exit; 4481 } 4482 4483 mTrack = track.get(); 4484 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4485 bool suspend = audio_is_bluetooth_sco_device( 4486 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4487 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4488 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4489 } 4490 lStatus = NO_ERROR; 4491 4492Exit: 4493 if (status) { 4494 *status = lStatus; 4495 } 4496 return track; 4497} 4498 4499status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4500{ 4501 ALOGV("RecordThread::start"); 4502 sp <ThreadBase> strongMe = this; 4503 status_t status = NO_ERROR; 4504 { 4505 AutoMutex lock(mLock); 4506 if (mActiveTrack != 0) { 4507 if (recordTrack != mActiveTrack.get()) { 4508 status = -EBUSY; 4509 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4510 mActiveTrack->mState = TrackBase::ACTIVE; 4511 } 4512 return status; 4513 } 4514 4515 recordTrack->mState = TrackBase::IDLE; 4516 mActiveTrack = recordTrack; 4517 mLock.unlock(); 4518 status_t status = AudioSystem::startInput(mId); 4519 mLock.lock(); 4520 if (status != NO_ERROR) { 4521 mActiveTrack.clear(); 4522 return status; 4523 } 4524 mRsmpInIndex = mFrameCount; 4525 mBytesRead = 0; 4526 if (mResampler != NULL) { 4527 mResampler->reset(); 4528 } 4529 mActiveTrack->mState = TrackBase::RESUMING; 4530 // signal thread to start 4531 ALOGV("Signal record thread"); 4532 mWaitWorkCV.signal(); 4533 // do not wait for mStartStopCond if exiting 4534 if (mExiting) { 4535 mActiveTrack.clear(); 4536 status = INVALID_OPERATION; 4537 goto startError; 4538 } 4539 mStartStopCond.wait(mLock); 4540 if (mActiveTrack == 0) { 4541 ALOGV("Record failed to start"); 4542 status = BAD_VALUE; 4543 goto startError; 4544 } 4545 ALOGV("Record started OK"); 4546 return status; 4547 } 4548startError: 4549 AudioSystem::stopInput(mId); 4550 return status; 4551} 4552 4553void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4554 ALOGV("RecordThread::stop"); 4555 sp <ThreadBase> strongMe = this; 4556 { 4557 AutoMutex lock(mLock); 4558 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4559 mActiveTrack->mState = TrackBase::PAUSING; 4560 // do not wait for mStartStopCond if exiting 4561 if (mExiting) { 4562 return; 4563 } 4564 mStartStopCond.wait(mLock); 4565 // if we have been restarted, recordTrack == mActiveTrack.get() here 4566 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4567 mLock.unlock(); 4568 AudioSystem::stopInput(mId); 4569 mLock.lock(); 4570 ALOGV("Record stopped OK"); 4571 } 4572 } 4573 } 4574} 4575 4576status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4577{ 4578 const size_t SIZE = 256; 4579 char buffer[SIZE]; 4580 String8 result; 4581 4582 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4583 result.append(buffer); 4584 4585 if (mActiveTrack != 0) { 4586 result.append("Active Track:\n"); 4587 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4588 mActiveTrack->dump(buffer, SIZE); 4589 result.append(buffer); 4590 4591 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4592 result.append(buffer); 4593 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4594 result.append(buffer); 4595 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4596 result.append(buffer); 4597 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4598 result.append(buffer); 4599 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4600 result.append(buffer); 4601 4602 4603 } else { 4604 result.append("No record client\n"); 4605 } 4606 write(fd, result.string(), result.size()); 4607 4608 dumpBase(fd, args); 4609 dumpEffectChains(fd, args); 4610 4611 return NO_ERROR; 4612} 4613 4614status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4615{ 4616 size_t framesReq = buffer->frameCount; 4617 size_t framesReady = mFrameCount - mRsmpInIndex; 4618 int channelCount; 4619 4620 if (framesReady == 0) { 4621 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4622 if (mBytesRead < 0) { 4623 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4624 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4625 // Force input into standby so that it tries to 4626 // recover at next read attempt 4627 mInput->stream->common.standby(&mInput->stream->common); 4628 usleep(kRecordThreadSleepUs); 4629 } 4630 buffer->raw = NULL; 4631 buffer->frameCount = 0; 4632 return NOT_ENOUGH_DATA; 4633 } 4634 mRsmpInIndex = 0; 4635 framesReady = mFrameCount; 4636 } 4637 4638 if (framesReq > framesReady) { 4639 framesReq = framesReady; 4640 } 4641 4642 if (mChannelCount == 1 && mReqChannelCount == 2) { 4643 channelCount = 1; 4644 } else { 4645 channelCount = 2; 4646 } 4647 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4648 buffer->frameCount = framesReq; 4649 return NO_ERROR; 4650} 4651 4652void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4653{ 4654 mRsmpInIndex += buffer->frameCount; 4655 buffer->frameCount = 0; 4656} 4657 4658bool AudioFlinger::RecordThread::checkForNewParameters_l() 4659{ 4660 bool reconfig = false; 4661 4662 while (!mNewParameters.isEmpty()) { 4663 status_t status = NO_ERROR; 4664 String8 keyValuePair = mNewParameters[0]; 4665 AudioParameter param = AudioParameter(keyValuePair); 4666 int value; 4667 audio_format_t reqFormat = mFormat; 4668 int reqSamplingRate = mReqSampleRate; 4669 int reqChannelCount = mReqChannelCount; 4670 4671 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4672 reqSamplingRate = value; 4673 reconfig = true; 4674 } 4675 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4676 reqFormat = (audio_format_t) value; 4677 reconfig = true; 4678 } 4679 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4680 reqChannelCount = popcount(value); 4681 reconfig = true; 4682 } 4683 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4684 // do not accept frame count changes if tracks are open as the track buffer 4685 // size depends on frame count and correct behavior would not be garantied 4686 // if frame count is changed after track creation 4687 if (mActiveTrack != 0) { 4688 status = INVALID_OPERATION; 4689 } else { 4690 reconfig = true; 4691 } 4692 } 4693 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4694 // forward device change to effects that have requested to be 4695 // aware of attached audio device. 4696 for (size_t i = 0; i < mEffectChains.size(); i++) { 4697 mEffectChains[i]->setDevice_l(value); 4698 } 4699 // store input device and output device but do not forward output device to audio HAL. 4700 // Note that status is ignored by the caller for output device 4701 // (see AudioFlinger::setParameters() 4702 if (value & AUDIO_DEVICE_OUT_ALL) { 4703 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4704 status = BAD_VALUE; 4705 } else { 4706 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4707 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4708 if (mTrack != NULL) { 4709 bool suspend = audio_is_bluetooth_sco_device( 4710 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4711 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4712 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4713 } 4714 } 4715 mDevice |= (uint32_t)value; 4716 } 4717 if (status == NO_ERROR) { 4718 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4719 if (status == INVALID_OPERATION) { 4720 mInput->stream->common.standby(&mInput->stream->common); 4721 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4722 } 4723 if (reconfig) { 4724 if (status == BAD_VALUE && 4725 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4726 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4727 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4728 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4729 (reqChannelCount < 3)) { 4730 status = NO_ERROR; 4731 } 4732 if (status == NO_ERROR) { 4733 readInputParameters(); 4734 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4735 } 4736 } 4737 } 4738 4739 mNewParameters.removeAt(0); 4740 4741 mParamStatus = status; 4742 mParamCond.signal(); 4743 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4744 // already timed out waiting for the status and will never signal the condition. 4745 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4746 } 4747 return reconfig; 4748} 4749 4750String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4751{ 4752 char *s; 4753 String8 out_s8 = String8(); 4754 4755 Mutex::Autolock _l(mLock); 4756 if (initCheck() != NO_ERROR) { 4757 return out_s8; 4758 } 4759 4760 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4761 out_s8 = String8(s); 4762 free(s); 4763 return out_s8; 4764} 4765 4766void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4767 AudioSystem::OutputDescriptor desc; 4768 void *param2 = NULL; 4769 4770 switch (event) { 4771 case AudioSystem::INPUT_OPENED: 4772 case AudioSystem::INPUT_CONFIG_CHANGED: 4773 desc.channels = mChannelMask; 4774 desc.samplingRate = mSampleRate; 4775 desc.format = mFormat; 4776 desc.frameCount = mFrameCount; 4777 desc.latency = 0; 4778 param2 = &desc; 4779 break; 4780 4781 case AudioSystem::INPUT_CLOSED: 4782 default: 4783 break; 4784 } 4785 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4786} 4787 4788void AudioFlinger::RecordThread::readInputParameters() 4789{ 4790 delete mRsmpInBuffer; 4791 // mRsmpInBuffer is always assigned a new[] below 4792 delete mRsmpOutBuffer; 4793 mRsmpOutBuffer = NULL; 4794 delete mResampler; 4795 mResampler = NULL; 4796 4797 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4798 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4799 mChannelCount = (uint16_t)popcount(mChannelMask); 4800 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4801 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4802 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4803 mFrameCount = mInputBytes / mFrameSize; 4804 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4805 4806 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4807 { 4808 int channelCount; 4809 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4810 // stereo to mono post process as the resampler always outputs stereo. 4811 if (mChannelCount == 1 && mReqChannelCount == 2) { 4812 channelCount = 1; 4813 } else { 4814 channelCount = 2; 4815 } 4816 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4817 mResampler->setSampleRate(mSampleRate); 4818 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4819 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4820 4821 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4822 if (mChannelCount == 1 && mReqChannelCount == 1) { 4823 mFrameCount >>= 1; 4824 } 4825 4826 } 4827 mRsmpInIndex = mFrameCount; 4828} 4829 4830unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4831{ 4832 Mutex::Autolock _l(mLock); 4833 if (initCheck() != NO_ERROR) { 4834 return 0; 4835 } 4836 4837 return mInput->stream->get_input_frames_lost(mInput->stream); 4838} 4839 4840uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4841{ 4842 Mutex::Autolock _l(mLock); 4843 uint32_t result = 0; 4844 if (getEffectChain_l(sessionId) != 0) { 4845 result = EFFECT_SESSION; 4846 } 4847 4848 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4849 result |= TRACK_SESSION; 4850 } 4851 4852 return result; 4853} 4854 4855AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4856{ 4857 Mutex::Autolock _l(mLock); 4858 return mTrack; 4859} 4860 4861AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4862{ 4863 Mutex::Autolock _l(mLock); 4864 return mInput; 4865} 4866 4867AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4868{ 4869 Mutex::Autolock _l(mLock); 4870 AudioStreamIn *input = mInput; 4871 mInput = NULL; 4872 return input; 4873} 4874 4875// this method must always be called either with ThreadBase mLock held or inside the thread loop 4876audio_stream_t* AudioFlinger::RecordThread::stream() 4877{ 4878 if (mInput == NULL) { 4879 return NULL; 4880 } 4881 return &mInput->stream->common; 4882} 4883 4884 4885// ---------------------------------------------------------------------------- 4886 4887audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 4888 uint32_t *pSamplingRate, 4889 audio_format_t *pFormat, 4890 uint32_t *pChannels, 4891 uint32_t *pLatencyMs, 4892 uint32_t flags) 4893{ 4894 status_t status; 4895 PlaybackThread *thread = NULL; 4896 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4897 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4898 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4899 uint32_t channels = pChannels ? *pChannels : 0; 4900 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4901 audio_stream_out_t *outStream; 4902 audio_hw_device_t *outHwDev; 4903 4904 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4905 pDevices ? *pDevices : 0, 4906 samplingRate, 4907 format, 4908 channels, 4909 flags); 4910 4911 if (pDevices == NULL || *pDevices == 0) { 4912 return 0; 4913 } 4914 4915 Mutex::Autolock _l(mLock); 4916 4917 outHwDev = findSuitableHwDev_l(*pDevices); 4918 if (outHwDev == NULL) 4919 return 0; 4920 4921 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4922 &channels, &samplingRate, &outStream); 4923 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4924 outStream, 4925 samplingRate, 4926 format, 4927 channels, 4928 status); 4929 4930 mHardwareStatus = AUDIO_HW_IDLE; 4931 if (outStream != NULL) { 4932 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4933 audio_io_handle_t id = nextUniqueId(); 4934 4935 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4936 (format != AUDIO_FORMAT_PCM_16_BIT) || 4937 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4938 thread = new DirectOutputThread(this, output, id, *pDevices); 4939 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4940 } else { 4941 thread = new MixerThread(this, output, id, *pDevices); 4942 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4943 } 4944 mPlaybackThreads.add(id, thread); 4945 4946 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 4947 if (pFormat != NULL) *pFormat = format; 4948 if (pChannels != NULL) *pChannels = channels; 4949 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 4950 4951 // notify client processes of the new output creation 4952 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4953 return id; 4954 } 4955 4956 return 0; 4957} 4958 4959audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 4960 audio_io_handle_t output2) 4961{ 4962 Mutex::Autolock _l(mLock); 4963 MixerThread *thread1 = checkMixerThread_l(output1); 4964 MixerThread *thread2 = checkMixerThread_l(output2); 4965 4966 if (thread1 == NULL || thread2 == NULL) { 4967 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4968 return 0; 4969 } 4970 4971 audio_io_handle_t id = nextUniqueId(); 4972 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4973 thread->addOutputTrack(thread2); 4974 mPlaybackThreads.add(id, thread); 4975 // notify client processes of the new output creation 4976 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4977 return id; 4978} 4979 4980status_t AudioFlinger::closeOutput(audio_io_handle_t output) 4981{ 4982 // keep strong reference on the playback thread so that 4983 // it is not destroyed while exit() is executed 4984 sp <PlaybackThread> thread; 4985 { 4986 Mutex::Autolock _l(mLock); 4987 thread = checkPlaybackThread_l(output); 4988 if (thread == NULL) { 4989 return BAD_VALUE; 4990 } 4991 4992 ALOGV("closeOutput() %d", output); 4993 4994 if (thread->type() == ThreadBase::MIXER) { 4995 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4996 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4997 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4998 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4999 } 5000 } 5001 } 5002 void *param2 = NULL; 5003 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5004 mPlaybackThreads.removeItem(output); 5005 } 5006 thread->exit(); 5007 5008 if (thread->type() != ThreadBase::DUPLICATING) { 5009 AudioStreamOut *out = thread->clearOutput(); 5010 assert(out != NULL); 5011 // from now on thread->mOutput is NULL 5012 out->hwDev->close_output_stream(out->hwDev, out->stream); 5013 delete out; 5014 } 5015 return NO_ERROR; 5016} 5017 5018status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5019{ 5020 Mutex::Autolock _l(mLock); 5021 PlaybackThread *thread = checkPlaybackThread_l(output); 5022 5023 if (thread == NULL) { 5024 return BAD_VALUE; 5025 } 5026 5027 ALOGV("suspendOutput() %d", output); 5028 thread->suspend(); 5029 5030 return NO_ERROR; 5031} 5032 5033status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5034{ 5035 Mutex::Autolock _l(mLock); 5036 PlaybackThread *thread = checkPlaybackThread_l(output); 5037 5038 if (thread == NULL) { 5039 return BAD_VALUE; 5040 } 5041 5042 ALOGV("restoreOutput() %d", output); 5043 5044 thread->restore(); 5045 5046 return NO_ERROR; 5047} 5048 5049audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5050 uint32_t *pSamplingRate, 5051 audio_format_t *pFormat, 5052 uint32_t *pChannels, 5053 audio_in_acoustics_t acoustics) 5054{ 5055 status_t status; 5056 RecordThread *thread = NULL; 5057 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5058 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5059 uint32_t channels = pChannels ? *pChannels : 0; 5060 uint32_t reqSamplingRate = samplingRate; 5061 audio_format_t reqFormat = format; 5062 uint32_t reqChannels = channels; 5063 audio_stream_in_t *inStream; 5064 audio_hw_device_t *inHwDev; 5065 5066 if (pDevices == NULL || *pDevices == 0) { 5067 return 0; 5068 } 5069 5070 Mutex::Autolock _l(mLock); 5071 5072 inHwDev = findSuitableHwDev_l(*pDevices); 5073 if (inHwDev == NULL) 5074 return 0; 5075 5076 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5077 &channels, &samplingRate, 5078 acoustics, 5079 &inStream); 5080 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5081 inStream, 5082 samplingRate, 5083 format, 5084 channels, 5085 acoustics, 5086 status); 5087 5088 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5089 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5090 // or stereo to mono conversions on 16 bit PCM inputs. 5091 if (inStream == NULL && status == BAD_VALUE && 5092 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5093 (samplingRate <= 2 * reqSamplingRate) && 5094 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5095 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5096 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5097 &channels, &samplingRate, 5098 acoustics, 5099 &inStream); 5100 } 5101 5102 if (inStream != NULL) { 5103 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5104 5105 audio_io_handle_t id = nextUniqueId(); 5106 // Start record thread 5107 // RecorThread require both input and output device indication to forward to audio 5108 // pre processing modules 5109 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5110 thread = new RecordThread(this, 5111 input, 5112 reqSamplingRate, 5113 reqChannels, 5114 id, 5115 device); 5116 mRecordThreads.add(id, thread); 5117 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5118 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5119 if (pFormat != NULL) *pFormat = format; 5120 if (pChannels != NULL) *pChannels = reqChannels; 5121 5122 input->stream->common.standby(&input->stream->common); 5123 5124 // notify client processes of the new input creation 5125 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5126 return id; 5127 } 5128 5129 return 0; 5130} 5131 5132status_t AudioFlinger::closeInput(audio_io_handle_t input) 5133{ 5134 // keep strong reference on the record thread so that 5135 // it is not destroyed while exit() is executed 5136 sp <RecordThread> thread; 5137 { 5138 Mutex::Autolock _l(mLock); 5139 thread = checkRecordThread_l(input); 5140 if (thread == NULL) { 5141 return BAD_VALUE; 5142 } 5143 5144 ALOGV("closeInput() %d", input); 5145 void *param2 = NULL; 5146 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5147 mRecordThreads.removeItem(input); 5148 } 5149 thread->exit(); 5150 5151 AudioStreamIn *in = thread->clearInput(); 5152 assert(in != NULL); 5153 // from now on thread->mInput is NULL 5154 in->hwDev->close_input_stream(in->hwDev, in->stream); 5155 delete in; 5156 5157 return NO_ERROR; 5158} 5159 5160status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5161{ 5162 Mutex::Autolock _l(mLock); 5163 MixerThread *dstThread = checkMixerThread_l(output); 5164 if (dstThread == NULL) { 5165 ALOGW("setStreamOutput() bad output id %d", output); 5166 return BAD_VALUE; 5167 } 5168 5169 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5170 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5171 5172 dstThread->setStreamValid(stream, true); 5173 5174 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5175 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5176 if (thread != dstThread && 5177 thread->type() != ThreadBase::DIRECT) { 5178 MixerThread *srcThread = (MixerThread *)thread; 5179 srcThread->setStreamValid(stream, false); 5180 srcThread->invalidateTracks(stream); 5181 } 5182 } 5183 5184 return NO_ERROR; 5185} 5186 5187 5188int AudioFlinger::newAudioSessionId() 5189{ 5190 return nextUniqueId(); 5191} 5192 5193void AudioFlinger::acquireAudioSessionId(int audioSession) 5194{ 5195 Mutex::Autolock _l(mLock); 5196 pid_t caller = IPCThreadState::self()->getCallingPid(); 5197 ALOGV("acquiring %d from %d", audioSession, caller); 5198 int num = mAudioSessionRefs.size(); 5199 for (int i = 0; i< num; i++) { 5200 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5201 if (ref->sessionid == audioSession && ref->pid == caller) { 5202 ref->cnt++; 5203 ALOGV(" incremented refcount to %d", ref->cnt); 5204 return; 5205 } 5206 } 5207 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5208 ALOGV(" added new entry for %d", audioSession); 5209} 5210 5211void AudioFlinger::releaseAudioSessionId(int audioSession) 5212{ 5213 Mutex::Autolock _l(mLock); 5214 pid_t caller = IPCThreadState::self()->getCallingPid(); 5215 ALOGV("releasing %d from %d", audioSession, caller); 5216 int num = mAudioSessionRefs.size(); 5217 for (int i = 0; i< num; i++) { 5218 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5219 if (ref->sessionid == audioSession && ref->pid == caller) { 5220 ref->cnt--; 5221 ALOGV(" decremented refcount to %d", ref->cnt); 5222 if (ref->cnt == 0) { 5223 mAudioSessionRefs.removeAt(i); 5224 delete ref; 5225 purgeStaleEffects_l(); 5226 } 5227 return; 5228 } 5229 } 5230 ALOGW("session id %d not found for pid %d", audioSession, caller); 5231} 5232 5233void AudioFlinger::purgeStaleEffects_l() { 5234 5235 ALOGV("purging stale effects"); 5236 5237 Vector< sp<EffectChain> > chains; 5238 5239 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5240 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5241 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5242 sp<EffectChain> ec = t->mEffectChains[j]; 5243 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5244 chains.push(ec); 5245 } 5246 } 5247 } 5248 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5249 sp<RecordThread> t = mRecordThreads.valueAt(i); 5250 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5251 sp<EffectChain> ec = t->mEffectChains[j]; 5252 chains.push(ec); 5253 } 5254 } 5255 5256 for (size_t i = 0; i < chains.size(); i++) { 5257 sp<EffectChain> ec = chains[i]; 5258 int sessionid = ec->sessionId(); 5259 sp<ThreadBase> t = ec->mThread.promote(); 5260 if (t == 0) { 5261 continue; 5262 } 5263 size_t numsessionrefs = mAudioSessionRefs.size(); 5264 bool found = false; 5265 for (size_t k = 0; k < numsessionrefs; k++) { 5266 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5267 if (ref->sessionid == sessionid) { 5268 ALOGV(" session %d still exists for %d with %d refs", 5269 sessionid, ref->pid, ref->cnt); 5270 found = true; 5271 break; 5272 } 5273 } 5274 if (!found) { 5275 // remove all effects from the chain 5276 while (ec->mEffects.size()) { 5277 sp<EffectModule> effect = ec->mEffects[0]; 5278 effect->unPin(); 5279 Mutex::Autolock _l (t->mLock); 5280 t->removeEffect_l(effect); 5281 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5282 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5283 if (handle != 0) { 5284 handle->mEffect.clear(); 5285 if (handle->mHasControl && handle->mEnabled) { 5286 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5287 } 5288 } 5289 } 5290 AudioSystem::unregisterEffect(effect->id()); 5291 } 5292 } 5293 } 5294 return; 5295} 5296 5297// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5298AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5299{ 5300 PlaybackThread *thread = NULL; 5301 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5302 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5303 } 5304 return thread; 5305} 5306 5307// checkMixerThread_l() must be called with AudioFlinger::mLock held 5308AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5309{ 5310 PlaybackThread *thread = checkPlaybackThread_l(output); 5311 if (thread != NULL) { 5312 if (thread->type() == ThreadBase::DIRECT) { 5313 thread = NULL; 5314 } 5315 } 5316 return (MixerThread *)thread; 5317} 5318 5319// checkRecordThread_l() must be called with AudioFlinger::mLock held 5320AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5321{ 5322 RecordThread *thread = NULL; 5323 if (mRecordThreads.indexOfKey(input) >= 0) { 5324 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5325 } 5326 return thread; 5327} 5328 5329uint32_t AudioFlinger::nextUniqueId() 5330{ 5331 return android_atomic_inc(&mNextUniqueId); 5332} 5333 5334AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5335{ 5336 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5337 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5338 AudioStreamOut *output = thread->getOutput(); 5339 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5340 return thread; 5341 } 5342 } 5343 return NULL; 5344} 5345 5346uint32_t AudioFlinger::primaryOutputDevice_l() 5347{ 5348 PlaybackThread *thread = primaryPlaybackThread_l(); 5349 5350 if (thread == NULL) { 5351 return 0; 5352 } 5353 5354 return thread->device(); 5355} 5356 5357 5358// ---------------------------------------------------------------------------- 5359// Effect management 5360// ---------------------------------------------------------------------------- 5361 5362 5363status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5364{ 5365 Mutex::Autolock _l(mLock); 5366 return EffectQueryNumberEffects(numEffects); 5367} 5368 5369status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5370{ 5371 Mutex::Autolock _l(mLock); 5372 return EffectQueryEffect(index, descriptor); 5373} 5374 5375status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5376 effect_descriptor_t *descriptor) const 5377{ 5378 Mutex::Autolock _l(mLock); 5379 return EffectGetDescriptor(pUuid, descriptor); 5380} 5381 5382 5383sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5384 effect_descriptor_t *pDesc, 5385 const sp<IEffectClient>& effectClient, 5386 int32_t priority, 5387 audio_io_handle_t io, 5388 int sessionId, 5389 status_t *status, 5390 int *id, 5391 int *enabled) 5392{ 5393 status_t lStatus = NO_ERROR; 5394 sp<EffectHandle> handle; 5395 effect_descriptor_t desc; 5396 5397 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5398 pid, effectClient.get(), priority, sessionId, io); 5399 5400 if (pDesc == NULL) { 5401 lStatus = BAD_VALUE; 5402 goto Exit; 5403 } 5404 5405 // check audio settings permission for global effects 5406 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5407 lStatus = PERMISSION_DENIED; 5408 goto Exit; 5409 } 5410 5411 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5412 // that can only be created by audio policy manager (running in same process) 5413 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5414 lStatus = PERMISSION_DENIED; 5415 goto Exit; 5416 } 5417 5418 if (io == 0) { 5419 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5420 // output must be specified by AudioPolicyManager when using session 5421 // AUDIO_SESSION_OUTPUT_STAGE 5422 lStatus = BAD_VALUE; 5423 goto Exit; 5424 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5425 // if the output returned by getOutputForEffect() is removed before we lock the 5426 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5427 // and we will exit safely 5428 io = AudioSystem::getOutputForEffect(&desc); 5429 } 5430 } 5431 5432 { 5433 Mutex::Autolock _l(mLock); 5434 5435 5436 if (!EffectIsNullUuid(&pDesc->uuid)) { 5437 // if uuid is specified, request effect descriptor 5438 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5439 if (lStatus < 0) { 5440 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5441 goto Exit; 5442 } 5443 } else { 5444 // if uuid is not specified, look for an available implementation 5445 // of the required type in effect factory 5446 if (EffectIsNullUuid(&pDesc->type)) { 5447 ALOGW("createEffect() no effect type"); 5448 lStatus = BAD_VALUE; 5449 goto Exit; 5450 } 5451 uint32_t numEffects = 0; 5452 effect_descriptor_t d; 5453 d.flags = 0; // prevent compiler warning 5454 bool found = false; 5455 5456 lStatus = EffectQueryNumberEffects(&numEffects); 5457 if (lStatus < 0) { 5458 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5459 goto Exit; 5460 } 5461 for (uint32_t i = 0; i < numEffects; i++) { 5462 lStatus = EffectQueryEffect(i, &desc); 5463 if (lStatus < 0) { 5464 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5465 continue; 5466 } 5467 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5468 // If matching type found save effect descriptor. If the session is 5469 // 0 and the effect is not auxiliary, continue enumeration in case 5470 // an auxiliary version of this effect type is available 5471 found = true; 5472 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5473 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5474 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5475 break; 5476 } 5477 } 5478 } 5479 if (!found) { 5480 lStatus = BAD_VALUE; 5481 ALOGW("createEffect() effect not found"); 5482 goto Exit; 5483 } 5484 // For same effect type, chose auxiliary version over insert version if 5485 // connect to output mix (Compliance to OpenSL ES) 5486 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5487 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5488 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5489 } 5490 } 5491 5492 // Do not allow auxiliary effects on a session different from 0 (output mix) 5493 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5494 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5495 lStatus = INVALID_OPERATION; 5496 goto Exit; 5497 } 5498 5499 // check recording permission for visualizer 5500 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5501 !recordingAllowed()) { 5502 lStatus = PERMISSION_DENIED; 5503 goto Exit; 5504 } 5505 5506 // return effect descriptor 5507 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5508 5509 // If output is not specified try to find a matching audio session ID in one of the 5510 // output threads. 5511 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5512 // because of code checking output when entering the function. 5513 // Note: io is never 0 when creating an effect on an input 5514 if (io == 0) { 5515 // look for the thread where the specified audio session is present 5516 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5517 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5518 io = mPlaybackThreads.keyAt(i); 5519 break; 5520 } 5521 } 5522 if (io == 0) { 5523 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5524 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5525 io = mRecordThreads.keyAt(i); 5526 break; 5527 } 5528 } 5529 } 5530 // If no output thread contains the requested session ID, default to 5531 // first output. The effect chain will be moved to the correct output 5532 // thread when a track with the same session ID is created 5533 if (io == 0 && mPlaybackThreads.size()) { 5534 io = mPlaybackThreads.keyAt(0); 5535 } 5536 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5537 } 5538 ThreadBase *thread = checkRecordThread_l(io); 5539 if (thread == NULL) { 5540 thread = checkPlaybackThread_l(io); 5541 if (thread == NULL) { 5542 ALOGE("createEffect() unknown output thread"); 5543 lStatus = BAD_VALUE; 5544 goto Exit; 5545 } 5546 } 5547 5548 sp<Client> client = registerPid_l(pid); 5549 5550 // create effect on selected output thread 5551 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5552 &desc, enabled, &lStatus); 5553 if (handle != 0 && id != NULL) { 5554 *id = handle->id(); 5555 } 5556 } 5557 5558Exit: 5559 if(status) { 5560 *status = lStatus; 5561 } 5562 return handle; 5563} 5564 5565status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 5566 audio_io_handle_t dstOutput) 5567{ 5568 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5569 sessionId, srcOutput, dstOutput); 5570 Mutex::Autolock _l(mLock); 5571 if (srcOutput == dstOutput) { 5572 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5573 return NO_ERROR; 5574 } 5575 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5576 if (srcThread == NULL) { 5577 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5578 return BAD_VALUE; 5579 } 5580 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5581 if (dstThread == NULL) { 5582 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5583 return BAD_VALUE; 5584 } 5585 5586 Mutex::Autolock _dl(dstThread->mLock); 5587 Mutex::Autolock _sl(srcThread->mLock); 5588 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5589 5590 return NO_ERROR; 5591} 5592 5593// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5594status_t AudioFlinger::moveEffectChain_l(int sessionId, 5595 AudioFlinger::PlaybackThread *srcThread, 5596 AudioFlinger::PlaybackThread *dstThread, 5597 bool reRegister) 5598{ 5599 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5600 sessionId, srcThread, dstThread); 5601 5602 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5603 if (chain == 0) { 5604 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5605 sessionId, srcThread); 5606 return INVALID_OPERATION; 5607 } 5608 5609 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5610 // so that a new chain is created with correct parameters when first effect is added. This is 5611 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5612 // removed. 5613 srcThread->removeEffectChain_l(chain); 5614 5615 // transfer all effects one by one so that new effect chain is created on new thread with 5616 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5617 audio_io_handle_t dstOutput = dstThread->id(); 5618 sp<EffectChain> dstChain; 5619 uint32_t strategy = 0; // prevent compiler warning 5620 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5621 while (effect != 0) { 5622 srcThread->removeEffect_l(effect); 5623 dstThread->addEffect_l(effect); 5624 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5625 if (effect->state() == EffectModule::ACTIVE || 5626 effect->state() == EffectModule::STOPPING) { 5627 effect->start(); 5628 } 5629 // if the move request is not received from audio policy manager, the effect must be 5630 // re-registered with the new strategy and output 5631 if (dstChain == 0) { 5632 dstChain = effect->chain().promote(); 5633 if (dstChain == 0) { 5634 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5635 srcThread->addEffect_l(effect); 5636 return NO_INIT; 5637 } 5638 strategy = dstChain->strategy(); 5639 } 5640 if (reRegister) { 5641 AudioSystem::unregisterEffect(effect->id()); 5642 AudioSystem::registerEffect(&effect->desc(), 5643 dstOutput, 5644 strategy, 5645 sessionId, 5646 effect->id()); 5647 } 5648 effect = chain->getEffectFromId_l(0); 5649 } 5650 5651 return NO_ERROR; 5652} 5653 5654 5655// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5656sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5657 const sp<AudioFlinger::Client>& client, 5658 const sp<IEffectClient>& effectClient, 5659 int32_t priority, 5660 int sessionId, 5661 effect_descriptor_t *desc, 5662 int *enabled, 5663 status_t *status 5664 ) 5665{ 5666 sp<EffectModule> effect; 5667 sp<EffectHandle> handle; 5668 status_t lStatus; 5669 sp<EffectChain> chain; 5670 bool chainCreated = false; 5671 bool effectCreated = false; 5672 bool effectRegistered = false; 5673 5674 lStatus = initCheck(); 5675 if (lStatus != NO_ERROR) { 5676 ALOGW("createEffect_l() Audio driver not initialized."); 5677 goto Exit; 5678 } 5679 5680 // Do not allow effects with session ID 0 on direct output or duplicating threads 5681 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5682 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5683 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5684 desc->name, sessionId); 5685 lStatus = BAD_VALUE; 5686 goto Exit; 5687 } 5688 // Only Pre processor effects are allowed on input threads and only on input threads 5689 if ((mType == RECORD && 5690 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5691 (mType != RECORD && 5692 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5693 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5694 desc->name, desc->flags, mType); 5695 lStatus = BAD_VALUE; 5696 goto Exit; 5697 } 5698 5699 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5700 5701 { // scope for mLock 5702 Mutex::Autolock _l(mLock); 5703 5704 // check for existing effect chain with the requested audio session 5705 chain = getEffectChain_l(sessionId); 5706 if (chain == 0) { 5707 // create a new chain for this session 5708 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5709 chain = new EffectChain(this, sessionId); 5710 addEffectChain_l(chain); 5711 chain->setStrategy(getStrategyForSession_l(sessionId)); 5712 chainCreated = true; 5713 } else { 5714 effect = chain->getEffectFromDesc_l(desc); 5715 } 5716 5717 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5718 5719 if (effect == 0) { 5720 int id = mAudioFlinger->nextUniqueId(); 5721 // Check CPU and memory usage 5722 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5723 if (lStatus != NO_ERROR) { 5724 goto Exit; 5725 } 5726 effectRegistered = true; 5727 // create a new effect module if none present in the chain 5728 effect = new EffectModule(this, chain, desc, id, sessionId); 5729 lStatus = effect->status(); 5730 if (lStatus != NO_ERROR) { 5731 goto Exit; 5732 } 5733 lStatus = chain->addEffect_l(effect); 5734 if (lStatus != NO_ERROR) { 5735 goto Exit; 5736 } 5737 effectCreated = true; 5738 5739 effect->setDevice(mDevice); 5740 effect->setMode(mAudioFlinger->getMode()); 5741 } 5742 // create effect handle and connect it to effect module 5743 handle = new EffectHandle(effect, client, effectClient, priority); 5744 lStatus = effect->addHandle(handle); 5745 if (enabled != NULL) { 5746 *enabled = (int)effect->isEnabled(); 5747 } 5748 } 5749 5750Exit: 5751 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5752 Mutex::Autolock _l(mLock); 5753 if (effectCreated) { 5754 chain->removeEffect_l(effect); 5755 } 5756 if (effectRegistered) { 5757 AudioSystem::unregisterEffect(effect->id()); 5758 } 5759 if (chainCreated) { 5760 removeEffectChain_l(chain); 5761 } 5762 handle.clear(); 5763 } 5764 5765 if(status) { 5766 *status = lStatus; 5767 } 5768 return handle; 5769} 5770 5771sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5772{ 5773 sp<EffectChain> chain = getEffectChain_l(sessionId); 5774 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 5775} 5776 5777// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5778// PlaybackThread::mLock held 5779status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5780{ 5781 // check for existing effect chain with the requested audio session 5782 int sessionId = effect->sessionId(); 5783 sp<EffectChain> chain = getEffectChain_l(sessionId); 5784 bool chainCreated = false; 5785 5786 if (chain == 0) { 5787 // create a new chain for this session 5788 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5789 chain = new EffectChain(this, sessionId); 5790 addEffectChain_l(chain); 5791 chain->setStrategy(getStrategyForSession_l(sessionId)); 5792 chainCreated = true; 5793 } 5794 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5795 5796 if (chain->getEffectFromId_l(effect->id()) != 0) { 5797 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5798 this, effect->desc().name, chain.get()); 5799 return BAD_VALUE; 5800 } 5801 5802 status_t status = chain->addEffect_l(effect); 5803 if (status != NO_ERROR) { 5804 if (chainCreated) { 5805 removeEffectChain_l(chain); 5806 } 5807 return status; 5808 } 5809 5810 effect->setDevice(mDevice); 5811 effect->setMode(mAudioFlinger->getMode()); 5812 return NO_ERROR; 5813} 5814 5815void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5816 5817 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5818 effect_descriptor_t desc = effect->desc(); 5819 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5820 detachAuxEffect_l(effect->id()); 5821 } 5822 5823 sp<EffectChain> chain = effect->chain().promote(); 5824 if (chain != 0) { 5825 // remove effect chain if removing last effect 5826 if (chain->removeEffect_l(effect) == 0) { 5827 removeEffectChain_l(chain); 5828 } 5829 } else { 5830 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5831 } 5832} 5833 5834void AudioFlinger::ThreadBase::lockEffectChains_l( 5835 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5836{ 5837 effectChains = mEffectChains; 5838 for (size_t i = 0; i < mEffectChains.size(); i++) { 5839 mEffectChains[i]->lock(); 5840 } 5841} 5842 5843void AudioFlinger::ThreadBase::unlockEffectChains( 5844 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5845{ 5846 for (size_t i = 0; i < effectChains.size(); i++) { 5847 effectChains[i]->unlock(); 5848 } 5849} 5850 5851sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5852{ 5853 Mutex::Autolock _l(mLock); 5854 return getEffectChain_l(sessionId); 5855} 5856 5857sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5858{ 5859 size_t size = mEffectChains.size(); 5860 for (size_t i = 0; i < size; i++) { 5861 if (mEffectChains[i]->sessionId() == sessionId) { 5862 return mEffectChains[i]; 5863 } 5864 } 5865 return 0; 5866} 5867 5868void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5869{ 5870 Mutex::Autolock _l(mLock); 5871 size_t size = mEffectChains.size(); 5872 for (size_t i = 0; i < size; i++) { 5873 mEffectChains[i]->setMode_l(mode); 5874 } 5875} 5876 5877void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5878 const wp<EffectHandle>& handle, 5879 bool unpiniflast) { 5880 5881 Mutex::Autolock _l(mLock); 5882 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5883 // delete the effect module if removing last handle on it 5884 if (effect->removeHandle(handle) == 0) { 5885 if (!effect->isPinned() || unpiniflast) { 5886 removeEffect_l(effect); 5887 AudioSystem::unregisterEffect(effect->id()); 5888 } 5889 } 5890} 5891 5892status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5893{ 5894 int session = chain->sessionId(); 5895 int16_t *buffer = mMixBuffer; 5896 bool ownsBuffer = false; 5897 5898 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5899 if (session > 0) { 5900 // Only one effect chain can be present in direct output thread and it uses 5901 // the mix buffer as input 5902 if (mType != DIRECT) { 5903 size_t numSamples = mFrameCount * mChannelCount; 5904 buffer = new int16_t[numSamples]; 5905 memset(buffer, 0, numSamples * sizeof(int16_t)); 5906 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5907 ownsBuffer = true; 5908 } 5909 5910 // Attach all tracks with same session ID to this chain. 5911 for (size_t i = 0; i < mTracks.size(); ++i) { 5912 sp<Track> track = mTracks[i]; 5913 if (session == track->sessionId()) { 5914 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5915 track->setMainBuffer(buffer); 5916 chain->incTrackCnt(); 5917 } 5918 } 5919 5920 // indicate all active tracks in the chain 5921 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5922 sp<Track> track = mActiveTracks[i].promote(); 5923 if (track == 0) continue; 5924 if (session == track->sessionId()) { 5925 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5926 chain->incActiveTrackCnt(); 5927 } 5928 } 5929 } 5930 5931 chain->setInBuffer(buffer, ownsBuffer); 5932 chain->setOutBuffer(mMixBuffer); 5933 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5934 // chains list in order to be processed last as it contains output stage effects 5935 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5936 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5937 // after track specific effects and before output stage 5938 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5939 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5940 // Effect chain for other sessions are inserted at beginning of effect 5941 // chains list to be processed before output mix effects. Relative order between other 5942 // sessions is not important 5943 size_t size = mEffectChains.size(); 5944 size_t i = 0; 5945 for (i = 0; i < size; i++) { 5946 if (mEffectChains[i]->sessionId() < session) break; 5947 } 5948 mEffectChains.insertAt(chain, i); 5949 checkSuspendOnAddEffectChain_l(chain); 5950 5951 return NO_ERROR; 5952} 5953 5954size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5955{ 5956 int session = chain->sessionId(); 5957 5958 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5959 5960 for (size_t i = 0; i < mEffectChains.size(); i++) { 5961 if (chain == mEffectChains[i]) { 5962 mEffectChains.removeAt(i); 5963 // detach all active tracks from the chain 5964 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5965 sp<Track> track = mActiveTracks[i].promote(); 5966 if (track == 0) continue; 5967 if (session == track->sessionId()) { 5968 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5969 chain.get(), session); 5970 chain->decActiveTrackCnt(); 5971 } 5972 } 5973 5974 // detach all tracks with same session ID from this chain 5975 for (size_t i = 0; i < mTracks.size(); ++i) { 5976 sp<Track> track = mTracks[i]; 5977 if (session == track->sessionId()) { 5978 track->setMainBuffer(mMixBuffer); 5979 chain->decTrackCnt(); 5980 } 5981 } 5982 break; 5983 } 5984 } 5985 return mEffectChains.size(); 5986} 5987 5988status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5989 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5990{ 5991 Mutex::Autolock _l(mLock); 5992 return attachAuxEffect_l(track, EffectId); 5993} 5994 5995status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5996 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5997{ 5998 status_t status = NO_ERROR; 5999 6000 if (EffectId == 0) { 6001 track->setAuxBuffer(0, NULL); 6002 } else { 6003 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6004 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6005 if (effect != 0) { 6006 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6007 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6008 } else { 6009 status = INVALID_OPERATION; 6010 } 6011 } else { 6012 status = BAD_VALUE; 6013 } 6014 } 6015 return status; 6016} 6017 6018void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6019{ 6020 for (size_t i = 0; i < mTracks.size(); ++i) { 6021 sp<Track> track = mTracks[i]; 6022 if (track->auxEffectId() == effectId) { 6023 attachAuxEffect_l(track, 0); 6024 } 6025 } 6026} 6027 6028status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6029{ 6030 // only one chain per input thread 6031 if (mEffectChains.size() != 0) { 6032 return INVALID_OPERATION; 6033 } 6034 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6035 6036 chain->setInBuffer(NULL); 6037 chain->setOutBuffer(NULL); 6038 6039 checkSuspendOnAddEffectChain_l(chain); 6040 6041 mEffectChains.add(chain); 6042 6043 return NO_ERROR; 6044} 6045 6046size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6047{ 6048 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6049 ALOGW_IF(mEffectChains.size() != 1, 6050 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6051 chain.get(), mEffectChains.size(), this); 6052 if (mEffectChains.size() == 1) { 6053 mEffectChains.removeAt(0); 6054 } 6055 return 0; 6056} 6057 6058// ---------------------------------------------------------------------------- 6059// EffectModule implementation 6060// ---------------------------------------------------------------------------- 6061 6062#undef LOG_TAG 6063#define LOG_TAG "AudioFlinger::EffectModule" 6064 6065AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6066 const wp<AudioFlinger::EffectChain>& chain, 6067 effect_descriptor_t *desc, 6068 int id, 6069 int sessionId) 6070 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6071 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6072{ 6073 ALOGV("Constructor %p", this); 6074 int lStatus; 6075 sp<ThreadBase> thread = mThread.promote(); 6076 if (thread == 0) { 6077 return; 6078 } 6079 6080 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6081 6082 // create effect engine from effect factory 6083 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6084 6085 if (mStatus != NO_ERROR) { 6086 return; 6087 } 6088 lStatus = init(); 6089 if (lStatus < 0) { 6090 mStatus = lStatus; 6091 goto Error; 6092 } 6093 6094 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6095 mPinned = true; 6096 } 6097 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6098 return; 6099Error: 6100 EffectRelease(mEffectInterface); 6101 mEffectInterface = NULL; 6102 ALOGV("Constructor Error %d", mStatus); 6103} 6104 6105AudioFlinger::EffectModule::~EffectModule() 6106{ 6107 ALOGV("Destructor %p", this); 6108 if (mEffectInterface != NULL) { 6109 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6110 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6111 sp<ThreadBase> thread = mThread.promote(); 6112 if (thread != 0) { 6113 audio_stream_t *stream = thread->stream(); 6114 if (stream != NULL) { 6115 stream->remove_audio_effect(stream, mEffectInterface); 6116 } 6117 } 6118 } 6119 // release effect engine 6120 EffectRelease(mEffectInterface); 6121 } 6122} 6123 6124status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6125{ 6126 status_t status; 6127 6128 Mutex::Autolock _l(mLock); 6129 // First handle in mHandles has highest priority and controls the effect module 6130 int priority = handle->priority(); 6131 size_t size = mHandles.size(); 6132 sp<EffectHandle> h; 6133 size_t i; 6134 for (i = 0; i < size; i++) { 6135 h = mHandles[i].promote(); 6136 if (h == 0) continue; 6137 if (h->priority() <= priority) break; 6138 } 6139 // if inserted in first place, move effect control from previous owner to this handle 6140 if (i == 0) { 6141 bool enabled = false; 6142 if (h != 0) { 6143 enabled = h->enabled(); 6144 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6145 } 6146 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6147 status = NO_ERROR; 6148 } else { 6149 status = ALREADY_EXISTS; 6150 } 6151 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6152 mHandles.insertAt(handle, i); 6153 return status; 6154} 6155 6156size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6157{ 6158 Mutex::Autolock _l(mLock); 6159 size_t size = mHandles.size(); 6160 size_t i; 6161 for (i = 0; i < size; i++) { 6162 if (mHandles[i] == handle) break; 6163 } 6164 if (i == size) { 6165 return size; 6166 } 6167 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6168 6169 bool enabled = false; 6170 EffectHandle *hdl = handle.unsafe_get(); 6171 if (hdl != NULL) { 6172 ALOGV("removeHandle() unsafe_get OK"); 6173 enabled = hdl->enabled(); 6174 } 6175 mHandles.removeAt(i); 6176 size = mHandles.size(); 6177 // if removed from first place, move effect control from this handle to next in line 6178 if (i == 0 && size != 0) { 6179 sp<EffectHandle> h = mHandles[0].promote(); 6180 if (h != 0) { 6181 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6182 } 6183 } 6184 6185 // Prevent calls to process() and other functions on effect interface from now on. 6186 // The effect engine will be released by the destructor when the last strong reference on 6187 // this object is released which can happen after next process is called. 6188 if (size == 0 && !mPinned) { 6189 mState = DESTROYED; 6190 } 6191 6192 return size; 6193} 6194 6195sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6196{ 6197 Mutex::Autolock _l(mLock); 6198 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6199} 6200 6201void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6202{ 6203 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6204 // keep a strong reference on this EffectModule to avoid calling the 6205 // destructor before we exit 6206 sp<EffectModule> keep(this); 6207 { 6208 sp<ThreadBase> thread = mThread.promote(); 6209 if (thread != 0) { 6210 thread->disconnectEffect(keep, handle, unpiniflast); 6211 } 6212 } 6213} 6214 6215void AudioFlinger::EffectModule::updateState() { 6216 Mutex::Autolock _l(mLock); 6217 6218 switch (mState) { 6219 case RESTART: 6220 reset_l(); 6221 // FALL THROUGH 6222 6223 case STARTING: 6224 // clear auxiliary effect input buffer for next accumulation 6225 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6226 memset(mConfig.inputCfg.buffer.raw, 6227 0, 6228 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6229 } 6230 start_l(); 6231 mState = ACTIVE; 6232 break; 6233 case STOPPING: 6234 stop_l(); 6235 mDisableWaitCnt = mMaxDisableWaitCnt; 6236 mState = STOPPED; 6237 break; 6238 case STOPPED: 6239 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6240 // turn off sequence. 6241 if (--mDisableWaitCnt == 0) { 6242 reset_l(); 6243 mState = IDLE; 6244 } 6245 break; 6246 default: //IDLE , ACTIVE, DESTROYED 6247 break; 6248 } 6249} 6250 6251void AudioFlinger::EffectModule::process() 6252{ 6253 Mutex::Autolock _l(mLock); 6254 6255 if (mState == DESTROYED || mEffectInterface == NULL || 6256 mConfig.inputCfg.buffer.raw == NULL || 6257 mConfig.outputCfg.buffer.raw == NULL) { 6258 return; 6259 } 6260 6261 if (isProcessEnabled()) { 6262 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6263 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6264 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6265 mConfig.inputCfg.buffer.s32, 6266 mConfig.inputCfg.buffer.frameCount/2); 6267 } 6268 6269 // do the actual processing in the effect engine 6270 int ret = (*mEffectInterface)->process(mEffectInterface, 6271 &mConfig.inputCfg.buffer, 6272 &mConfig.outputCfg.buffer); 6273 6274 // force transition to IDLE state when engine is ready 6275 if (mState == STOPPED && ret == -ENODATA) { 6276 mDisableWaitCnt = 1; 6277 } 6278 6279 // clear auxiliary effect input buffer for next accumulation 6280 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6281 memset(mConfig.inputCfg.buffer.raw, 0, 6282 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6283 } 6284 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6285 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6286 // If an insert effect is idle and input buffer is different from output buffer, 6287 // accumulate input onto output 6288 sp<EffectChain> chain = mChain.promote(); 6289 if (chain != 0 && chain->activeTrackCnt() != 0) { 6290 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6291 int16_t *in = mConfig.inputCfg.buffer.s16; 6292 int16_t *out = mConfig.outputCfg.buffer.s16; 6293 for (size_t i = 0; i < frameCnt; i++) { 6294 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6295 } 6296 } 6297 } 6298} 6299 6300void AudioFlinger::EffectModule::reset_l() 6301{ 6302 if (mEffectInterface == NULL) { 6303 return; 6304 } 6305 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6306} 6307 6308status_t AudioFlinger::EffectModule::configure() 6309{ 6310 uint32_t channels; 6311 if (mEffectInterface == NULL) { 6312 return NO_INIT; 6313 } 6314 6315 sp<ThreadBase> thread = mThread.promote(); 6316 if (thread == 0) { 6317 return DEAD_OBJECT; 6318 } 6319 6320 // TODO: handle configuration of effects replacing track process 6321 if (thread->channelCount() == 1) { 6322 channels = AUDIO_CHANNEL_OUT_MONO; 6323 } else { 6324 channels = AUDIO_CHANNEL_OUT_STEREO; 6325 } 6326 6327 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6328 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6329 } else { 6330 mConfig.inputCfg.channels = channels; 6331 } 6332 mConfig.outputCfg.channels = channels; 6333 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6334 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6335 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6336 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6337 mConfig.inputCfg.bufferProvider.cookie = NULL; 6338 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6339 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6340 mConfig.outputCfg.bufferProvider.cookie = NULL; 6341 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6342 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6343 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6344 // Insert effect: 6345 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6346 // always overwrites output buffer: input buffer == output buffer 6347 // - in other sessions: 6348 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6349 // other effect: overwrites output buffer: input buffer == output buffer 6350 // Auxiliary effect: 6351 // accumulates in output buffer: input buffer != output buffer 6352 // Therefore: accumulate <=> input buffer != output buffer 6353 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6354 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6355 } else { 6356 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6357 } 6358 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6359 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6360 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6361 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6362 6363 ALOGV("configure() %p thread %p buffer %p framecount %d", 6364 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6365 6366 status_t cmdStatus; 6367 uint32_t size = sizeof(int); 6368 status_t status = (*mEffectInterface)->command(mEffectInterface, 6369 EFFECT_CMD_SET_CONFIG, 6370 sizeof(effect_config_t), 6371 &mConfig, 6372 &size, 6373 &cmdStatus); 6374 if (status == 0) { 6375 status = cmdStatus; 6376 } 6377 6378 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6379 (1000 * mConfig.outputCfg.buffer.frameCount); 6380 6381 return status; 6382} 6383 6384status_t AudioFlinger::EffectModule::init() 6385{ 6386 Mutex::Autolock _l(mLock); 6387 if (mEffectInterface == NULL) { 6388 return NO_INIT; 6389 } 6390 status_t cmdStatus; 6391 uint32_t size = sizeof(status_t); 6392 status_t status = (*mEffectInterface)->command(mEffectInterface, 6393 EFFECT_CMD_INIT, 6394 0, 6395 NULL, 6396 &size, 6397 &cmdStatus); 6398 if (status == 0) { 6399 status = cmdStatus; 6400 } 6401 return status; 6402} 6403 6404status_t AudioFlinger::EffectModule::start() 6405{ 6406 Mutex::Autolock _l(mLock); 6407 return start_l(); 6408} 6409 6410status_t AudioFlinger::EffectModule::start_l() 6411{ 6412 if (mEffectInterface == NULL) { 6413 return NO_INIT; 6414 } 6415 status_t cmdStatus; 6416 uint32_t size = sizeof(status_t); 6417 status_t status = (*mEffectInterface)->command(mEffectInterface, 6418 EFFECT_CMD_ENABLE, 6419 0, 6420 NULL, 6421 &size, 6422 &cmdStatus); 6423 if (status == 0) { 6424 status = cmdStatus; 6425 } 6426 if (status == 0 && 6427 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6428 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6429 sp<ThreadBase> thread = mThread.promote(); 6430 if (thread != 0) { 6431 audio_stream_t *stream = thread->stream(); 6432 if (stream != NULL) { 6433 stream->add_audio_effect(stream, mEffectInterface); 6434 } 6435 } 6436 } 6437 return status; 6438} 6439 6440status_t AudioFlinger::EffectModule::stop() 6441{ 6442 Mutex::Autolock _l(mLock); 6443 return stop_l(); 6444} 6445 6446status_t AudioFlinger::EffectModule::stop_l() 6447{ 6448 if (mEffectInterface == NULL) { 6449 return NO_INIT; 6450 } 6451 status_t cmdStatus; 6452 uint32_t size = sizeof(status_t); 6453 status_t status = (*mEffectInterface)->command(mEffectInterface, 6454 EFFECT_CMD_DISABLE, 6455 0, 6456 NULL, 6457 &size, 6458 &cmdStatus); 6459 if (status == 0) { 6460 status = cmdStatus; 6461 } 6462 if (status == 0 && 6463 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6464 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6465 sp<ThreadBase> thread = mThread.promote(); 6466 if (thread != 0) { 6467 audio_stream_t *stream = thread->stream(); 6468 if (stream != NULL) { 6469 stream->remove_audio_effect(stream, mEffectInterface); 6470 } 6471 } 6472 } 6473 return status; 6474} 6475 6476status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6477 uint32_t cmdSize, 6478 void *pCmdData, 6479 uint32_t *replySize, 6480 void *pReplyData) 6481{ 6482 Mutex::Autolock _l(mLock); 6483// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6484 6485 if (mState == DESTROYED || mEffectInterface == NULL) { 6486 return NO_INIT; 6487 } 6488 status_t status = (*mEffectInterface)->command(mEffectInterface, 6489 cmdCode, 6490 cmdSize, 6491 pCmdData, 6492 replySize, 6493 pReplyData); 6494 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6495 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6496 for (size_t i = 1; i < mHandles.size(); i++) { 6497 sp<EffectHandle> h = mHandles[i].promote(); 6498 if (h != 0) { 6499 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6500 } 6501 } 6502 } 6503 return status; 6504} 6505 6506status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6507{ 6508 6509 Mutex::Autolock _l(mLock); 6510 ALOGV("setEnabled %p enabled %d", this, enabled); 6511 6512 if (enabled != isEnabled()) { 6513 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6514 if (enabled && status != NO_ERROR) { 6515 return status; 6516 } 6517 6518 switch (mState) { 6519 // going from disabled to enabled 6520 case IDLE: 6521 mState = STARTING; 6522 break; 6523 case STOPPED: 6524 mState = RESTART; 6525 break; 6526 case STOPPING: 6527 mState = ACTIVE; 6528 break; 6529 6530 // going from enabled to disabled 6531 case RESTART: 6532 mState = STOPPED; 6533 break; 6534 case STARTING: 6535 mState = IDLE; 6536 break; 6537 case ACTIVE: 6538 mState = STOPPING; 6539 break; 6540 case DESTROYED: 6541 return NO_ERROR; // simply ignore as we are being destroyed 6542 } 6543 for (size_t i = 1; i < mHandles.size(); i++) { 6544 sp<EffectHandle> h = mHandles[i].promote(); 6545 if (h != 0) { 6546 h->setEnabled(enabled); 6547 } 6548 } 6549 } 6550 return NO_ERROR; 6551} 6552 6553bool AudioFlinger::EffectModule::isEnabled() const 6554{ 6555 switch (mState) { 6556 case RESTART: 6557 case STARTING: 6558 case ACTIVE: 6559 return true; 6560 case IDLE: 6561 case STOPPING: 6562 case STOPPED: 6563 case DESTROYED: 6564 default: 6565 return false; 6566 } 6567} 6568 6569bool AudioFlinger::EffectModule::isProcessEnabled() const 6570{ 6571 switch (mState) { 6572 case RESTART: 6573 case ACTIVE: 6574 case STOPPING: 6575 case STOPPED: 6576 return true; 6577 case IDLE: 6578 case STARTING: 6579 case DESTROYED: 6580 default: 6581 return false; 6582 } 6583} 6584 6585status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6586{ 6587 Mutex::Autolock _l(mLock); 6588 status_t status = NO_ERROR; 6589 6590 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6591 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6592 if (isProcessEnabled() && 6593 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6594 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6595 status_t cmdStatus; 6596 uint32_t volume[2]; 6597 uint32_t *pVolume = NULL; 6598 uint32_t size = sizeof(volume); 6599 volume[0] = *left; 6600 volume[1] = *right; 6601 if (controller) { 6602 pVolume = volume; 6603 } 6604 status = (*mEffectInterface)->command(mEffectInterface, 6605 EFFECT_CMD_SET_VOLUME, 6606 size, 6607 volume, 6608 &size, 6609 pVolume); 6610 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6611 *left = volume[0]; 6612 *right = volume[1]; 6613 } 6614 } 6615 return status; 6616} 6617 6618status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6619{ 6620 Mutex::Autolock _l(mLock); 6621 status_t status = NO_ERROR; 6622 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6623 // audio pre processing modules on RecordThread can receive both output and 6624 // input device indication in the same call 6625 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6626 if (dev) { 6627 status_t cmdStatus; 6628 uint32_t size = sizeof(status_t); 6629 6630 status = (*mEffectInterface)->command(mEffectInterface, 6631 EFFECT_CMD_SET_DEVICE, 6632 sizeof(uint32_t), 6633 &dev, 6634 &size, 6635 &cmdStatus); 6636 if (status == NO_ERROR) { 6637 status = cmdStatus; 6638 } 6639 } 6640 dev = device & AUDIO_DEVICE_IN_ALL; 6641 if (dev) { 6642 status_t cmdStatus; 6643 uint32_t size = sizeof(status_t); 6644 6645 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6646 EFFECT_CMD_SET_INPUT_DEVICE, 6647 sizeof(uint32_t), 6648 &dev, 6649 &size, 6650 &cmdStatus); 6651 if (status2 == NO_ERROR) { 6652 status2 = cmdStatus; 6653 } 6654 if (status == NO_ERROR) { 6655 status = status2; 6656 } 6657 } 6658 } 6659 return status; 6660} 6661 6662status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6663{ 6664 Mutex::Autolock _l(mLock); 6665 status_t status = NO_ERROR; 6666 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6667 status_t cmdStatus; 6668 uint32_t size = sizeof(status_t); 6669 status = (*mEffectInterface)->command(mEffectInterface, 6670 EFFECT_CMD_SET_AUDIO_MODE, 6671 sizeof(audio_mode_t), 6672 &mode, 6673 &size, 6674 &cmdStatus); 6675 if (status == NO_ERROR) { 6676 status = cmdStatus; 6677 } 6678 } 6679 return status; 6680} 6681 6682void AudioFlinger::EffectModule::setSuspended(bool suspended) 6683{ 6684 Mutex::Autolock _l(mLock); 6685 mSuspended = suspended; 6686} 6687 6688bool AudioFlinger::EffectModule::suspended() const 6689{ 6690 Mutex::Autolock _l(mLock); 6691 return mSuspended; 6692} 6693 6694status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6695{ 6696 const size_t SIZE = 256; 6697 char buffer[SIZE]; 6698 String8 result; 6699 6700 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6701 result.append(buffer); 6702 6703 bool locked = tryLock(mLock); 6704 // failed to lock - AudioFlinger is probably deadlocked 6705 if (!locked) { 6706 result.append("\t\tCould not lock Fx mutex:\n"); 6707 } 6708 6709 result.append("\t\tSession Status State Engine:\n"); 6710 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6711 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6712 result.append(buffer); 6713 6714 result.append("\t\tDescriptor:\n"); 6715 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6716 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6717 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6718 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6719 result.append(buffer); 6720 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6721 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6722 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6723 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6724 result.append(buffer); 6725 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6726 mDescriptor.apiVersion, 6727 mDescriptor.flags); 6728 result.append(buffer); 6729 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6730 mDescriptor.name); 6731 result.append(buffer); 6732 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6733 mDescriptor.implementor); 6734 result.append(buffer); 6735 6736 result.append("\t\t- Input configuration:\n"); 6737 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6738 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6739 (uint32_t)mConfig.inputCfg.buffer.raw, 6740 mConfig.inputCfg.buffer.frameCount, 6741 mConfig.inputCfg.samplingRate, 6742 mConfig.inputCfg.channels, 6743 mConfig.inputCfg.format); 6744 result.append(buffer); 6745 6746 result.append("\t\t- Output configuration:\n"); 6747 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6748 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6749 (uint32_t)mConfig.outputCfg.buffer.raw, 6750 mConfig.outputCfg.buffer.frameCount, 6751 mConfig.outputCfg.samplingRate, 6752 mConfig.outputCfg.channels, 6753 mConfig.outputCfg.format); 6754 result.append(buffer); 6755 6756 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6757 result.append(buffer); 6758 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6759 for (size_t i = 0; i < mHandles.size(); ++i) { 6760 sp<EffectHandle> handle = mHandles[i].promote(); 6761 if (handle != 0) { 6762 handle->dump(buffer, SIZE); 6763 result.append(buffer); 6764 } 6765 } 6766 6767 result.append("\n"); 6768 6769 write(fd, result.string(), result.length()); 6770 6771 if (locked) { 6772 mLock.unlock(); 6773 } 6774 6775 return NO_ERROR; 6776} 6777 6778// ---------------------------------------------------------------------------- 6779// EffectHandle implementation 6780// ---------------------------------------------------------------------------- 6781 6782#undef LOG_TAG 6783#define LOG_TAG "AudioFlinger::EffectHandle" 6784 6785AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6786 const sp<AudioFlinger::Client>& client, 6787 const sp<IEffectClient>& effectClient, 6788 int32_t priority) 6789 : BnEffect(), 6790 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6791 mPriority(priority), mHasControl(false), mEnabled(false) 6792{ 6793 ALOGV("constructor %p", this); 6794 6795 if (client == 0) { 6796 return; 6797 } 6798 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6799 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6800 if (mCblkMemory != 0) { 6801 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6802 6803 if (mCblk != NULL) { 6804 new(mCblk) effect_param_cblk_t(); 6805 mBuffer = (uint8_t *)mCblk + bufOffset; 6806 } 6807 } else { 6808 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6809 return; 6810 } 6811} 6812 6813AudioFlinger::EffectHandle::~EffectHandle() 6814{ 6815 ALOGV("Destructor %p", this); 6816 disconnect(false); 6817 ALOGV("Destructor DONE %p", this); 6818} 6819 6820status_t AudioFlinger::EffectHandle::enable() 6821{ 6822 ALOGV("enable %p", this); 6823 if (!mHasControl) return INVALID_OPERATION; 6824 if (mEffect == 0) return DEAD_OBJECT; 6825 6826 if (mEnabled) { 6827 return NO_ERROR; 6828 } 6829 6830 mEnabled = true; 6831 6832 sp<ThreadBase> thread = mEffect->thread().promote(); 6833 if (thread != 0) { 6834 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6835 } 6836 6837 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6838 if (mEffect->suspended()) { 6839 return NO_ERROR; 6840 } 6841 6842 status_t status = mEffect->setEnabled(true); 6843 if (status != NO_ERROR) { 6844 if (thread != 0) { 6845 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6846 } 6847 mEnabled = false; 6848 } 6849 return status; 6850} 6851 6852status_t AudioFlinger::EffectHandle::disable() 6853{ 6854 ALOGV("disable %p", this); 6855 if (!mHasControl) return INVALID_OPERATION; 6856 if (mEffect == 0) return DEAD_OBJECT; 6857 6858 if (!mEnabled) { 6859 return NO_ERROR; 6860 } 6861 mEnabled = false; 6862 6863 if (mEffect->suspended()) { 6864 return NO_ERROR; 6865 } 6866 6867 status_t status = mEffect->setEnabled(false); 6868 6869 sp<ThreadBase> thread = mEffect->thread().promote(); 6870 if (thread != 0) { 6871 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6872 } 6873 6874 return status; 6875} 6876 6877void AudioFlinger::EffectHandle::disconnect() 6878{ 6879 disconnect(true); 6880} 6881 6882void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6883{ 6884 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6885 if (mEffect == 0) { 6886 return; 6887 } 6888 mEffect->disconnect(this, unpiniflast); 6889 6890 if (mHasControl && mEnabled) { 6891 sp<ThreadBase> thread = mEffect->thread().promote(); 6892 if (thread != 0) { 6893 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6894 } 6895 } 6896 6897 // release sp on module => module destructor can be called now 6898 mEffect.clear(); 6899 if (mClient != 0) { 6900 if (mCblk != NULL) { 6901 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 6902 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6903 } 6904 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 6905 // Client destructor must run with AudioFlinger mutex locked 6906 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6907 mClient.clear(); 6908 } 6909} 6910 6911status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6912 uint32_t cmdSize, 6913 void *pCmdData, 6914 uint32_t *replySize, 6915 void *pReplyData) 6916{ 6917// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6918// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6919 6920 // only get parameter command is permitted for applications not controlling the effect 6921 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6922 return INVALID_OPERATION; 6923 } 6924 if (mEffect == 0) return DEAD_OBJECT; 6925 if (mClient == 0) return INVALID_OPERATION; 6926 6927 // handle commands that are not forwarded transparently to effect engine 6928 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6929 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6930 // no risk to block the whole media server process or mixer threads is we are stuck here 6931 Mutex::Autolock _l(mCblk->lock); 6932 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6933 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6934 mCblk->serverIndex = 0; 6935 mCblk->clientIndex = 0; 6936 return BAD_VALUE; 6937 } 6938 status_t status = NO_ERROR; 6939 while (mCblk->serverIndex < mCblk->clientIndex) { 6940 int reply; 6941 uint32_t rsize = sizeof(int); 6942 int *p = (int *)(mBuffer + mCblk->serverIndex); 6943 int size = *p++; 6944 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6945 ALOGW("command(): invalid parameter block size"); 6946 break; 6947 } 6948 effect_param_t *param = (effect_param_t *)p; 6949 if (param->psize == 0 || param->vsize == 0) { 6950 ALOGW("command(): null parameter or value size"); 6951 mCblk->serverIndex += size; 6952 continue; 6953 } 6954 uint32_t psize = sizeof(effect_param_t) + 6955 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6956 param->vsize; 6957 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6958 psize, 6959 p, 6960 &rsize, 6961 &reply); 6962 // stop at first error encountered 6963 if (ret != NO_ERROR) { 6964 status = ret; 6965 *(int *)pReplyData = reply; 6966 break; 6967 } else if (reply != NO_ERROR) { 6968 *(int *)pReplyData = reply; 6969 break; 6970 } 6971 mCblk->serverIndex += size; 6972 } 6973 mCblk->serverIndex = 0; 6974 mCblk->clientIndex = 0; 6975 return status; 6976 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6977 *(int *)pReplyData = NO_ERROR; 6978 return enable(); 6979 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6980 *(int *)pReplyData = NO_ERROR; 6981 return disable(); 6982 } 6983 6984 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6985} 6986 6987void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 6988{ 6989 ALOGV("setControl %p control %d", this, hasControl); 6990 6991 mHasControl = hasControl; 6992 mEnabled = enabled; 6993 6994 if (signal && mEffectClient != 0) { 6995 mEffectClient->controlStatusChanged(hasControl); 6996 } 6997} 6998 6999void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7000 uint32_t cmdSize, 7001 void *pCmdData, 7002 uint32_t replySize, 7003 void *pReplyData) 7004{ 7005 if (mEffectClient != 0) { 7006 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7007 } 7008} 7009 7010 7011 7012void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7013{ 7014 if (mEffectClient != 0) { 7015 mEffectClient->enableStatusChanged(enabled); 7016 } 7017} 7018 7019status_t AudioFlinger::EffectHandle::onTransact( 7020 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7021{ 7022 return BnEffect::onTransact(code, data, reply, flags); 7023} 7024 7025 7026void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7027{ 7028 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7029 7030 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7031 (mClient == 0) ? getpid() : mClient->pid(), 7032 mPriority, 7033 mHasControl, 7034 !locked, 7035 mCblk ? mCblk->clientIndex : 0, 7036 mCblk ? mCblk->serverIndex : 0 7037 ); 7038 7039 if (locked) { 7040 mCblk->lock.unlock(); 7041 } 7042} 7043 7044#undef LOG_TAG 7045#define LOG_TAG "AudioFlinger::EffectChain" 7046 7047AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7048 int sessionId) 7049 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7050 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7051 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7052{ 7053 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7054 sp<ThreadBase> thread = mThread.promote(); 7055 if (thread == 0) { 7056 return; 7057 } 7058 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7059 thread->frameCount(); 7060} 7061 7062AudioFlinger::EffectChain::~EffectChain() 7063{ 7064 if (mOwnInBuffer) { 7065 delete mInBuffer; 7066 } 7067 7068} 7069 7070// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7071sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7072{ 7073 size_t size = mEffects.size(); 7074 7075 for (size_t i = 0; i < size; i++) { 7076 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7077 return mEffects[i]; 7078 } 7079 } 7080 return 0; 7081} 7082 7083// getEffectFromId_l() must be called with ThreadBase::mLock held 7084sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7085{ 7086 size_t size = mEffects.size(); 7087 7088 for (size_t i = 0; i < size; i++) { 7089 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7090 if (id == 0 || mEffects[i]->id() == id) { 7091 return mEffects[i]; 7092 } 7093 } 7094 return 0; 7095} 7096 7097// getEffectFromType_l() must be called with ThreadBase::mLock held 7098sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7099 const effect_uuid_t *type) 7100{ 7101 size_t size = mEffects.size(); 7102 7103 for (size_t i = 0; i < size; i++) { 7104 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7105 return mEffects[i]; 7106 } 7107 } 7108 return 0; 7109} 7110 7111// Must be called with EffectChain::mLock locked 7112void AudioFlinger::EffectChain::process_l() 7113{ 7114 sp<ThreadBase> thread = mThread.promote(); 7115 if (thread == 0) { 7116 ALOGW("process_l(): cannot promote mixer thread"); 7117 return; 7118 } 7119 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7120 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7121 // always process effects unless no more tracks are on the session and the effect tail 7122 // has been rendered 7123 bool doProcess = true; 7124 if (!isGlobalSession) { 7125 bool tracksOnSession = (trackCnt() != 0); 7126 7127 if (!tracksOnSession && mTailBufferCount == 0) { 7128 doProcess = false; 7129 } 7130 7131 if (activeTrackCnt() == 0) { 7132 // if no track is active and the effect tail has not been rendered, 7133 // the input buffer must be cleared here as the mixer process will not do it 7134 if (tracksOnSession || mTailBufferCount > 0) { 7135 size_t numSamples = thread->frameCount() * thread->channelCount(); 7136 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7137 if (mTailBufferCount > 0) { 7138 mTailBufferCount--; 7139 } 7140 } 7141 } 7142 } 7143 7144 size_t size = mEffects.size(); 7145 if (doProcess) { 7146 for (size_t i = 0; i < size; i++) { 7147 mEffects[i]->process(); 7148 } 7149 } 7150 for (size_t i = 0; i < size; i++) { 7151 mEffects[i]->updateState(); 7152 } 7153} 7154 7155// addEffect_l() must be called with PlaybackThread::mLock held 7156status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7157{ 7158 effect_descriptor_t desc = effect->desc(); 7159 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7160 7161 Mutex::Autolock _l(mLock); 7162 effect->setChain(this); 7163 sp<ThreadBase> thread = mThread.promote(); 7164 if (thread == 0) { 7165 return NO_INIT; 7166 } 7167 effect->setThread(thread); 7168 7169 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7170 // Auxiliary effects are inserted at the beginning of mEffects vector as 7171 // they are processed first and accumulated in chain input buffer 7172 mEffects.insertAt(effect, 0); 7173 7174 // the input buffer for auxiliary effect contains mono samples in 7175 // 32 bit format. This is to avoid saturation in AudoMixer 7176 // accumulation stage. Saturation is done in EffectModule::process() before 7177 // calling the process in effect engine 7178 size_t numSamples = thread->frameCount(); 7179 int32_t *buffer = new int32_t[numSamples]; 7180 memset(buffer, 0, numSamples * sizeof(int32_t)); 7181 effect->setInBuffer((int16_t *)buffer); 7182 // auxiliary effects output samples to chain input buffer for further processing 7183 // by insert effects 7184 effect->setOutBuffer(mInBuffer); 7185 } else { 7186 // Insert effects are inserted at the end of mEffects vector as they are processed 7187 // after track and auxiliary effects. 7188 // Insert effect order as a function of indicated preference: 7189 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7190 // another effect is present 7191 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7192 // last effect claiming first position 7193 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7194 // first effect claiming last position 7195 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7196 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7197 // already present 7198 7199 int size = (int)mEffects.size(); 7200 int idx_insert = size; 7201 int idx_insert_first = -1; 7202 int idx_insert_last = -1; 7203 7204 for (int i = 0; i < size; i++) { 7205 effect_descriptor_t d = mEffects[i]->desc(); 7206 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7207 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7208 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7209 // check invalid effect chaining combinations 7210 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7211 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7212 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7213 return INVALID_OPERATION; 7214 } 7215 // remember position of first insert effect and by default 7216 // select this as insert position for new effect 7217 if (idx_insert == size) { 7218 idx_insert = i; 7219 } 7220 // remember position of last insert effect claiming 7221 // first position 7222 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7223 idx_insert_first = i; 7224 } 7225 // remember position of first insert effect claiming 7226 // last position 7227 if (iPref == EFFECT_FLAG_INSERT_LAST && 7228 idx_insert_last == -1) { 7229 idx_insert_last = i; 7230 } 7231 } 7232 } 7233 7234 // modify idx_insert from first position if needed 7235 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7236 if (idx_insert_last != -1) { 7237 idx_insert = idx_insert_last; 7238 } else { 7239 idx_insert = size; 7240 } 7241 } else { 7242 if (idx_insert_first != -1) { 7243 idx_insert = idx_insert_first + 1; 7244 } 7245 } 7246 7247 // always read samples from chain input buffer 7248 effect->setInBuffer(mInBuffer); 7249 7250 // if last effect in the chain, output samples to chain 7251 // output buffer, otherwise to chain input buffer 7252 if (idx_insert == size) { 7253 if (idx_insert != 0) { 7254 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7255 mEffects[idx_insert-1]->configure(); 7256 } 7257 effect->setOutBuffer(mOutBuffer); 7258 } else { 7259 effect->setOutBuffer(mInBuffer); 7260 } 7261 mEffects.insertAt(effect, idx_insert); 7262 7263 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7264 } 7265 effect->configure(); 7266 return NO_ERROR; 7267} 7268 7269// removeEffect_l() must be called with PlaybackThread::mLock held 7270size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7271{ 7272 Mutex::Autolock _l(mLock); 7273 int size = (int)mEffects.size(); 7274 int i; 7275 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7276 7277 for (i = 0; i < size; i++) { 7278 if (effect == mEffects[i]) { 7279 // calling stop here will remove pre-processing effect from the audio HAL. 7280 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7281 // the middle of a read from audio HAL 7282 if (mEffects[i]->state() == EffectModule::ACTIVE || 7283 mEffects[i]->state() == EffectModule::STOPPING) { 7284 mEffects[i]->stop(); 7285 } 7286 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7287 delete[] effect->inBuffer(); 7288 } else { 7289 if (i == size - 1 && i != 0) { 7290 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7291 mEffects[i - 1]->configure(); 7292 } 7293 } 7294 mEffects.removeAt(i); 7295 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7296 break; 7297 } 7298 } 7299 7300 return mEffects.size(); 7301} 7302 7303// setDevice_l() must be called with PlaybackThread::mLock held 7304void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7305{ 7306 size_t size = mEffects.size(); 7307 for (size_t i = 0; i < size; i++) { 7308 mEffects[i]->setDevice(device); 7309 } 7310} 7311 7312// setMode_l() must be called with PlaybackThread::mLock held 7313void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7314{ 7315 size_t size = mEffects.size(); 7316 for (size_t i = 0; i < size; i++) { 7317 mEffects[i]->setMode(mode); 7318 } 7319} 7320 7321// setVolume_l() must be called with PlaybackThread::mLock held 7322bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7323{ 7324 uint32_t newLeft = *left; 7325 uint32_t newRight = *right; 7326 bool hasControl = false; 7327 int ctrlIdx = -1; 7328 size_t size = mEffects.size(); 7329 7330 // first update volume controller 7331 for (size_t i = size; i > 0; i--) { 7332 if (mEffects[i - 1]->isProcessEnabled() && 7333 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7334 ctrlIdx = i - 1; 7335 hasControl = true; 7336 break; 7337 } 7338 } 7339 7340 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7341 if (hasControl) { 7342 *left = mNewLeftVolume; 7343 *right = mNewRightVolume; 7344 } 7345 return hasControl; 7346 } 7347 7348 mVolumeCtrlIdx = ctrlIdx; 7349 mLeftVolume = newLeft; 7350 mRightVolume = newRight; 7351 7352 // second get volume update from volume controller 7353 if (ctrlIdx >= 0) { 7354 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7355 mNewLeftVolume = newLeft; 7356 mNewRightVolume = newRight; 7357 } 7358 // then indicate volume to all other effects in chain. 7359 // Pass altered volume to effects before volume controller 7360 // and requested volume to effects after controller 7361 uint32_t lVol = newLeft; 7362 uint32_t rVol = newRight; 7363 7364 for (size_t i = 0; i < size; i++) { 7365 if ((int)i == ctrlIdx) continue; 7366 // this also works for ctrlIdx == -1 when there is no volume controller 7367 if ((int)i > ctrlIdx) { 7368 lVol = *left; 7369 rVol = *right; 7370 } 7371 mEffects[i]->setVolume(&lVol, &rVol, false); 7372 } 7373 *left = newLeft; 7374 *right = newRight; 7375 7376 return hasControl; 7377} 7378 7379status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7380{ 7381 const size_t SIZE = 256; 7382 char buffer[SIZE]; 7383 String8 result; 7384 7385 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7386 result.append(buffer); 7387 7388 bool locked = tryLock(mLock); 7389 // failed to lock - AudioFlinger is probably deadlocked 7390 if (!locked) { 7391 result.append("\tCould not lock mutex:\n"); 7392 } 7393 7394 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7395 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7396 mEffects.size(), 7397 (uint32_t)mInBuffer, 7398 (uint32_t)mOutBuffer, 7399 mActiveTrackCnt); 7400 result.append(buffer); 7401 write(fd, result.string(), result.size()); 7402 7403 for (size_t i = 0; i < mEffects.size(); ++i) { 7404 sp<EffectModule> effect = mEffects[i]; 7405 if (effect != 0) { 7406 effect->dump(fd, args); 7407 } 7408 } 7409 7410 if (locked) { 7411 mLock.unlock(); 7412 } 7413 7414 return NO_ERROR; 7415} 7416 7417// must be called with ThreadBase::mLock held 7418void AudioFlinger::EffectChain::setEffectSuspended_l( 7419 const effect_uuid_t *type, bool suspend) 7420{ 7421 sp<SuspendedEffectDesc> desc; 7422 // use effect type UUID timelow as key as there is no real risk of identical 7423 // timeLow fields among effect type UUIDs. 7424 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7425 if (suspend) { 7426 if (index >= 0) { 7427 desc = mSuspendedEffects.valueAt(index); 7428 } else { 7429 desc = new SuspendedEffectDesc(); 7430 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7431 mSuspendedEffects.add(type->timeLow, desc); 7432 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7433 } 7434 if (desc->mRefCount++ == 0) { 7435 sp<EffectModule> effect = getEffectIfEnabled(type); 7436 if (effect != 0) { 7437 desc->mEffect = effect; 7438 effect->setSuspended(true); 7439 effect->setEnabled(false); 7440 } 7441 } 7442 } else { 7443 if (index < 0) { 7444 return; 7445 } 7446 desc = mSuspendedEffects.valueAt(index); 7447 if (desc->mRefCount <= 0) { 7448 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7449 desc->mRefCount = 1; 7450 } 7451 if (--desc->mRefCount == 0) { 7452 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7453 if (desc->mEffect != 0) { 7454 sp<EffectModule> effect = desc->mEffect.promote(); 7455 if (effect != 0) { 7456 effect->setSuspended(false); 7457 sp<EffectHandle> handle = effect->controlHandle(); 7458 if (handle != 0) { 7459 effect->setEnabled(handle->enabled()); 7460 } 7461 } 7462 desc->mEffect.clear(); 7463 } 7464 mSuspendedEffects.removeItemsAt(index); 7465 } 7466 } 7467} 7468 7469// must be called with ThreadBase::mLock held 7470void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7471{ 7472 sp<SuspendedEffectDesc> desc; 7473 7474 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7475 if (suspend) { 7476 if (index >= 0) { 7477 desc = mSuspendedEffects.valueAt(index); 7478 } else { 7479 desc = new SuspendedEffectDesc(); 7480 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7481 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7482 } 7483 if (desc->mRefCount++ == 0) { 7484 Vector< sp<EffectModule> > effects; 7485 getSuspendEligibleEffects(effects); 7486 for (size_t i = 0; i < effects.size(); i++) { 7487 setEffectSuspended_l(&effects[i]->desc().type, true); 7488 } 7489 } 7490 } else { 7491 if (index < 0) { 7492 return; 7493 } 7494 desc = mSuspendedEffects.valueAt(index); 7495 if (desc->mRefCount <= 0) { 7496 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7497 desc->mRefCount = 1; 7498 } 7499 if (--desc->mRefCount == 0) { 7500 Vector<const effect_uuid_t *> types; 7501 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7502 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7503 continue; 7504 } 7505 types.add(&mSuspendedEffects.valueAt(i)->mType); 7506 } 7507 for (size_t i = 0; i < types.size(); i++) { 7508 setEffectSuspended_l(types[i], false); 7509 } 7510 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7511 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7512 } 7513 } 7514} 7515 7516 7517// The volume effect is used for automated tests only 7518#ifndef OPENSL_ES_H_ 7519static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7520 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7521const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7522#endif //OPENSL_ES_H_ 7523 7524bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7525{ 7526 // auxiliary effects and visualizer are never suspended on output mix 7527 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7528 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7529 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7530 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7531 return false; 7532 } 7533 return true; 7534} 7535 7536void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 7537{ 7538 effects.clear(); 7539 for (size_t i = 0; i < mEffects.size(); i++) { 7540 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 7541 effects.add(mEffects[i]); 7542 } 7543 } 7544} 7545 7546sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7547 const effect_uuid_t *type) 7548{ 7549 sp<EffectModule> effect = getEffectFromType_l(type); 7550 return effect != 0 && effect->isEnabled() ? effect : 0; 7551} 7552 7553void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7554 bool enabled) 7555{ 7556 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7557 if (enabled) { 7558 if (index < 0) { 7559 // if the effect is not suspend check if all effects are suspended 7560 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7561 if (index < 0) { 7562 return; 7563 } 7564 if (!isEffectEligibleForSuspend(effect->desc())) { 7565 return; 7566 } 7567 setEffectSuspended_l(&effect->desc().type, enabled); 7568 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7569 if (index < 0) { 7570 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7571 return; 7572 } 7573 } 7574 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7575 effect->desc().type.timeLow); 7576 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7577 // if effect is requested to suspended but was not yet enabled, supend it now. 7578 if (desc->mEffect == 0) { 7579 desc->mEffect = effect; 7580 effect->setEnabled(false); 7581 effect->setSuspended(true); 7582 } 7583 } else { 7584 if (index < 0) { 7585 return; 7586 } 7587 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7588 effect->desc().type.timeLow); 7589 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7590 desc->mEffect.clear(); 7591 effect->setSuspended(false); 7592 } 7593} 7594 7595#undef LOG_TAG 7596#define LOG_TAG "AudioFlinger" 7597 7598// ---------------------------------------------------------------------------- 7599 7600status_t AudioFlinger::onTransact( 7601 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7602{ 7603 return BnAudioFlinger::onTransact(code, data, reply, flags); 7604} 7605 7606}; // namespace android 7607