AudioFlinger.cpp revision 83d86538c4c479a9225c75ab27938e8f05abb9c8
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/AudioTrack.h> 41#include <media/AudioRecord.h> 42#include <media/IMediaPlayerService.h> 43#include <media/IMediaDeathNotifier.h> 44 45#include <private/media/AudioTrackShared.h> 46#include <private/media/AudioEffectShared.h> 47 48#include <system/audio.h> 49#include <hardware/audio.h> 50 51#include "AudioMixer.h" 52#include "AudioFlinger.h" 53 54#include <media/EffectsFactoryApi.h> 55#include <audio_effects/effect_visualizer.h> 56#include <audio_effects/effect_ns.h> 57#include <audio_effects/effect_aec.h> 58 59#include <audio_utils/primitives.h> 60 61#include <cpustats/ThreadCpuUsage.h> 62#include <powermanager/PowerManager.h> 63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 64 65// ---------------------------------------------------------------------------- 66 67 68namespace android { 69 70static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 71static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 72 73//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 74static const float MAX_GAIN = 4096.0f; 75static const uint32_t MAX_GAIN_INT = 0x1000; 76 77// retry counts for buffer fill timeout 78// 50 * ~20msecs = 1 second 79static const int8_t kMaxTrackRetries = 50; 80static const int8_t kMaxTrackStartupRetries = 50; 81// allow less retry attempts on direct output thread. 82// direct outputs can be a scarce resource in audio hardware and should 83// be released as quickly as possible. 84static const int8_t kMaxTrackRetriesDirect = 2; 85 86static const int kDumpLockRetries = 50; 87static const int kDumpLockSleepUs = 20000; 88 89// don't warn about blocked writes or record buffer overflows more often than this 90static const nsecs_t kWarningThrottleNs = seconds(5); 91 92// RecordThread loop sleep time upon application overrun or audio HAL read error 93static const int kRecordThreadSleepUs = 5000; 94 95// maximum time to wait for setParameters to complete 96static const nsecs_t kSetParametersTimeoutNs = seconds(2); 97 98// minimum sleep time for the mixer thread loop when tracks are active but in underrun 99static const uint32_t kMinThreadSleepTimeUs = 5000; 100// maximum divider applied to the active sleep time in the mixer thread loop 101static const uint32_t kMaxThreadSleepTimeShift = 2; 102 103 104// ---------------------------------------------------------------------------- 105 106static bool recordingAllowed() { 107 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 108 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 109 if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO"); 110 return ok; 111} 112 113static bool settingsAllowed() { 114 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 115 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 116 if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 117 return ok; 118} 119 120// To collect the amplifier usage 121static void addBatteryData(uint32_t params) { 122 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 123 if (service == NULL) { 124 // it already logged 125 return; 126 } 127 128 service->addBatteryData(params); 129} 130 131static int load_audio_interface(const char *if_name, const hw_module_t **mod, 132 audio_hw_device_t **dev) 133{ 134 int rc; 135 136 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 137 if (rc) 138 goto out; 139 140 rc = audio_hw_device_open(*mod, dev); 141 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 142 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 143 if (rc) 144 goto out; 145 146 return 0; 147 148out: 149 *mod = NULL; 150 *dev = NULL; 151 return rc; 152} 153 154static const char * const audio_interfaces[] = { 155 "primary", 156 "a2dp", 157 "usb", 158}; 159#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 160 161// ---------------------------------------------------------------------------- 162 163AudioFlinger::AudioFlinger() 164 : BnAudioFlinger(), 165 mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 166 mBtNrecIsOff(false) 167{ 168} 169 170void AudioFlinger::onFirstRef() 171{ 172 int rc = 0; 173 174 Mutex::Autolock _l(mLock); 175 176 /* TODO: move all this work into an Init() function */ 177 mHardwareStatus = AUDIO_HW_IDLE; 178 179 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 180 const hw_module_t *mod; 181 audio_hw_device_t *dev; 182 183 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 184 if (rc) 185 continue; 186 187 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 mAudioHwDevs.clear(); 249} 250 251audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 252{ 253 /* first matching HW device is returned */ 254 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 255 audio_hw_device_t *dev = mAudioHwDevs[i]; 256 if ((dev->get_supported_devices(dev) & devices) == devices) 257 return dev; 258 } 259 return NULL; 260} 261 262status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 263{ 264 const size_t SIZE = 256; 265 char buffer[SIZE]; 266 String8 result; 267 268 result.append("Clients:\n"); 269 for (size_t i = 0; i < mClients.size(); ++i) { 270 wp<Client> wClient = mClients.valueAt(i); 271 if (wClient != 0) { 272 sp<Client> client = wClient.promote(); 273 if (client != 0) { 274 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 275 result.append(buffer); 276 } 277 } 278 } 279 280 result.append("Global session refs:\n"); 281 result.append(" session pid cnt\n"); 282 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 283 AudioSessionRef *r = mAudioSessionRefs[i]; 284 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 285 result.append(buffer); 286 } 287 write(fd, result.string(), result.size()); 288 return NO_ERROR; 289} 290 291 292status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 293{ 294 const size_t SIZE = 256; 295 char buffer[SIZE]; 296 String8 result; 297 hardware_call_state hardwareStatus = mHardwareStatus; 298 299 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 300 result.append(buffer); 301 write(fd, result.string(), result.size()); 302 return NO_ERROR; 303} 304 305status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 306{ 307 const size_t SIZE = 256; 308 char buffer[SIZE]; 309 String8 result; 310 snprintf(buffer, SIZE, "Permission Denial: " 311 "can't dump AudioFlinger from pid=%d, uid=%d\n", 312 IPCThreadState::self()->getCallingPid(), 313 IPCThreadState::self()->getCallingUid()); 314 result.append(buffer); 315 write(fd, result.string(), result.size()); 316 return NO_ERROR; 317} 318 319static bool tryLock(Mutex& mutex) 320{ 321 bool locked = false; 322 for (int i = 0; i < kDumpLockRetries; ++i) { 323 if (mutex.tryLock() == NO_ERROR) { 324 locked = true; 325 break; 326 } 327 usleep(kDumpLockSleepUs); 328 } 329 return locked; 330} 331 332status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 333{ 334 if (!checkCallingPermission(String16("android.permission.DUMP"))) { 335 dumpPermissionDenial(fd, args); 336 } else { 337 // get state of hardware lock 338 bool hardwareLocked = tryLock(mHardwareLock); 339 if (!hardwareLocked) { 340 String8 result(kHardwareLockedString); 341 write(fd, result.string(), result.size()); 342 } else { 343 mHardwareLock.unlock(); 344 } 345 346 bool locked = tryLock(mLock); 347 348 // failed to lock - AudioFlinger is probably deadlocked 349 if (!locked) { 350 String8 result(kDeadlockedString); 351 write(fd, result.string(), result.size()); 352 } 353 354 dumpClients(fd, args); 355 dumpInternals(fd, args); 356 357 // dump playback threads 358 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 359 mPlaybackThreads.valueAt(i)->dump(fd, args); 360 } 361 362 // dump record threads 363 for (size_t i = 0; i < mRecordThreads.size(); i++) { 364 mRecordThreads.valueAt(i)->dump(fd, args); 365 } 366 367 // dump all hardware devs 368 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 369 audio_hw_device_t *dev = mAudioHwDevs[i]; 370 dev->dump(dev, fd); 371 } 372 if (locked) mLock.unlock(); 373 } 374 return NO_ERROR; 375} 376 377 378// IAudioFlinger interface 379 380 381sp<IAudioTrack> AudioFlinger::createTrack( 382 pid_t pid, 383 audio_stream_type_t streamType, 384 uint32_t sampleRate, 385 audio_format_t format, 386 uint32_t channelMask, 387 int frameCount, 388 uint32_t flags, 389 const sp<IMemory>& sharedBuffer, 390 int output, 391 int *sessionId, 392 status_t *status) 393{ 394 sp<PlaybackThread::Track> track; 395 sp<TrackHandle> trackHandle; 396 sp<Client> client; 397 wp<Client> wclient; 398 status_t lStatus; 399 int lSessionId; 400 401 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 402 // but if someone uses binder directly they could bypass that and cause us to crash 403 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 404 ALOGE("createTrack() invalid stream type %d", streamType); 405 lStatus = BAD_VALUE; 406 goto Exit; 407 } 408 409 { 410 Mutex::Autolock _l(mLock); 411 PlaybackThread *thread = checkPlaybackThread_l(output); 412 PlaybackThread *effectThread = NULL; 413 if (thread == NULL) { 414 ALOGE("unknown output thread"); 415 lStatus = BAD_VALUE; 416 goto Exit; 417 } 418 419 wclient = mClients.valueFor(pid); 420 421 if (wclient != NULL) { 422 client = wclient.promote(); 423 } else { 424 client = new Client(this, pid); 425 mClients.add(pid, client); 426 } 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(int 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(int 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(int 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(int 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(int 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, int output) 662{ 663 // check calling permissions 664 if (!settingsAllowed()) { 665 return PERMISSION_DENIED; 666 } 667 668 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 669 ALOGE("setStreamVolume() invalid stream %d", stream); 670 return BAD_VALUE; 671 } 672 673 AutoMutex lock(mLock); 674 PlaybackThread *thread = NULL; 675 if (output) { 676 thread = checkPlaybackThread_l(output); 677 if (thread == NULL) { 678 return BAD_VALUE; 679 } 680 } 681 682 mStreamTypes[stream].volume = value; 683 684 if (thread == NULL) { 685 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 686 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 687 } 688 } else { 689 thread->setStreamVolume(stream, value); 690 } 691 692 return NO_ERROR; 693} 694 695status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 696{ 697 // check calling permissions 698 if (!settingsAllowed()) { 699 return PERMISSION_DENIED; 700 } 701 702 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 703 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 704 ALOGE("setStreamMute() invalid stream %d", stream); 705 return BAD_VALUE; 706 } 707 708 AutoMutex lock(mLock); 709 mStreamTypes[stream].mute = muted; 710 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 711 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 712 713 return NO_ERROR; 714} 715 716float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const 717{ 718 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 719 return 0.0f; 720 } 721 722 AutoMutex lock(mLock); 723 float volume; 724 if (output) { 725 PlaybackThread *thread = checkPlaybackThread_l(output); 726 if (thread == NULL) { 727 return 0.0f; 728 } 729 volume = thread->streamVolume(stream); 730 } else { 731 volume = mStreamTypes[stream].volume; 732 } 733 734 return volume; 735} 736 737bool AudioFlinger::streamMute(audio_stream_type_t stream) const 738{ 739 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 740 return true; 741 } 742 743 return mStreamTypes[stream].mute; 744} 745 746status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 747{ 748 status_t result; 749 750 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 751 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 752 // check calling permissions 753 if (!settingsAllowed()) { 754 return PERMISSION_DENIED; 755 } 756 757 // ioHandle == 0 means the parameters are global to the audio hardware interface 758 if (ioHandle == 0) { 759 AutoMutex lock(mHardwareLock); 760 mHardwareStatus = AUDIO_SET_PARAMETER; 761 status_t final_result = NO_ERROR; 762 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 763 audio_hw_device_t *dev = mAudioHwDevs[i]; 764 result = dev->set_parameters(dev, keyValuePairs.string()); 765 final_result = result ?: final_result; 766 } 767 mHardwareStatus = AUDIO_HW_IDLE; 768 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 769 AudioParameter param = AudioParameter(keyValuePairs); 770 String8 value; 771 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 772 Mutex::Autolock _l(mLock); 773 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 774 if (mBtNrecIsOff != btNrecIsOff) { 775 for (size_t i = 0; i < mRecordThreads.size(); i++) { 776 sp<RecordThread> thread = mRecordThreads.valueAt(i); 777 RecordThread::RecordTrack *track = thread->track(); 778 if (track != NULL) { 779 audio_devices_t device = (audio_devices_t)( 780 thread->device() & AUDIO_DEVICE_IN_ALL); 781 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 782 thread->setEffectSuspended(FX_IID_AEC, 783 suspend, 784 track->sessionId()); 785 thread->setEffectSuspended(FX_IID_NS, 786 suspend, 787 track->sessionId()); 788 } 789 } 790 mBtNrecIsOff = btNrecIsOff; 791 } 792 } 793 return final_result; 794 } 795 796 // hold a strong ref on thread in case closeOutput() or closeInput() is called 797 // and the thread is exited once the lock is released 798 sp<ThreadBase> thread; 799 { 800 Mutex::Autolock _l(mLock); 801 thread = checkPlaybackThread_l(ioHandle); 802 if (thread == NULL) { 803 thread = checkRecordThread_l(ioHandle); 804 } else if (thread == primaryPlaybackThread_l()) { 805 // indicate output device change to all input threads for pre processing 806 AudioParameter param = AudioParameter(keyValuePairs); 807 int value; 808 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 809 for (size_t i = 0; i < mRecordThreads.size(); i++) { 810 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 811 } 812 } 813 } 814 } 815 if (thread != NULL) { 816 result = thread->setParameters(keyValuePairs); 817 return result; 818 } 819 return BAD_VALUE; 820} 821 822String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 823{ 824// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %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) 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(int ioHandle) 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, int output) 898{ 899 status_t status; 900 901 Mutex::Autolock _l(mLock); 902 903 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 904 if (playbackThread != NULL) { 905 return playbackThread->getRenderPosition(halFrames, dspFrames); 906 } 907 908 return BAD_VALUE; 909} 910 911void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 912{ 913 914 Mutex::Autolock _l(mLock); 915 916 int pid = IPCThreadState::self()->getCallingPid(); 917 if (mNotificationClients.indexOfKey(pid) < 0) { 918 sp<NotificationClient> notificationClient = new NotificationClient(this, 919 client, 920 pid); 921 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 922 923 mNotificationClients.add(pid, notificationClient); 924 925 sp<IBinder> binder = client->asBinder(); 926 binder->linkToDeath(notificationClient); 927 928 // the config change is always sent from playback or record threads to avoid deadlock 929 // with AudioSystem::gLock 930 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 931 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 932 } 933 934 for (size_t i = 0; i < mRecordThreads.size(); i++) { 935 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 936 } 937 } 938} 939 940void AudioFlinger::removeNotificationClient(pid_t pid) 941{ 942 Mutex::Autolock _l(mLock); 943 944 int index = mNotificationClients.indexOfKey(pid); 945 if (index >= 0) { 946 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 947 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 948 mNotificationClients.removeItem(pid); 949 } 950 951 ALOGV("%d died, releasing its sessions", pid); 952 int num = mAudioSessionRefs.size(); 953 bool removed = false; 954 for (int i = 0; i< num; i++) { 955 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 956 ALOGV(" pid %d @ %d", ref->pid, i); 957 if (ref->pid == pid) { 958 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 959 mAudioSessionRefs.removeAt(i); 960 delete ref; 961 removed = true; 962 i--; 963 num--; 964 } 965 } 966 if (removed) { 967 purgeStaleEffects_l(); 968 } 969} 970 971// audioConfigChanged_l() must be called with AudioFlinger::mLock held 972void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 973{ 974 size_t size = mNotificationClients.size(); 975 for (size_t i = 0; i < size; i++) { 976 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 977 } 978} 979 980// removeClient_l() must be called with AudioFlinger::mLock held 981void AudioFlinger::removeClient_l(pid_t pid) 982{ 983 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 984 mClients.removeItem(pid); 985} 986 987 988// ---------------------------------------------------------------------------- 989 990AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 991 : Thread(false), 992 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 993 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false), 994 mDevice(device) 995{ 996 mDeathRecipient = new PMDeathRecipient(this); 997} 998 999AudioFlinger::ThreadBase::~ThreadBase() 1000{ 1001 mParamCond.broadcast(); 1002 // do not lock the mutex in destructor 1003 releaseWakeLock_l(); 1004 if (mPowerManager != 0) { 1005 sp<IBinder> binder = mPowerManager->asBinder(); 1006 binder->unlinkToDeath(mDeathRecipient); 1007 } 1008} 1009 1010void AudioFlinger::ThreadBase::exit() 1011{ 1012 // keep a strong ref on ourself so that we won't get 1013 // destroyed in the middle of requestExitAndWait() 1014 sp <ThreadBase> strongMe = this; 1015 1016 ALOGV("ThreadBase::exit"); 1017 { 1018 AutoMutex lock(mLock); 1019 mExiting = true; 1020 requestExit(); 1021 mWaitWorkCV.signal(); 1022 } 1023 requestExitAndWait(); 1024} 1025 1026uint32_t AudioFlinger::ThreadBase::sampleRate() const 1027{ 1028 return mSampleRate; 1029} 1030 1031int AudioFlinger::ThreadBase::channelCount() const 1032{ 1033 return (int)mChannelCount; 1034} 1035 1036audio_format_t AudioFlinger::ThreadBase::format() const 1037{ 1038 return mFormat; 1039} 1040 1041size_t AudioFlinger::ThreadBase::frameCount() const 1042{ 1043 return mFrameCount; 1044} 1045 1046status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1047{ 1048 status_t status; 1049 1050 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1051 Mutex::Autolock _l(mLock); 1052 1053 mNewParameters.add(keyValuePairs); 1054 mWaitWorkCV.signal(); 1055 // wait condition with timeout in case the thread loop has exited 1056 // before the request could be processed 1057 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1058 status = mParamStatus; 1059 mWaitWorkCV.signal(); 1060 } else { 1061 status = TIMED_OUT; 1062 } 1063 return status; 1064} 1065 1066void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1067{ 1068 Mutex::Autolock _l(mLock); 1069 sendConfigEvent_l(event, param); 1070} 1071 1072// sendConfigEvent_l() must be called with ThreadBase::mLock held 1073void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1074{ 1075 ConfigEvent configEvent; 1076 configEvent.mEvent = event; 1077 configEvent.mParam = param; 1078 mConfigEvents.add(configEvent); 1079 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1080 mWaitWorkCV.signal(); 1081} 1082 1083void AudioFlinger::ThreadBase::processConfigEvents() 1084{ 1085 mLock.lock(); 1086 while(!mConfigEvents.isEmpty()) { 1087 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1088 ConfigEvent configEvent = mConfigEvents[0]; 1089 mConfigEvents.removeAt(0); 1090 // release mLock before locking AudioFlinger mLock: lock order is always 1091 // AudioFlinger then ThreadBase to avoid cross deadlock 1092 mLock.unlock(); 1093 mAudioFlinger->mLock.lock(); 1094 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1095 mAudioFlinger->mLock.unlock(); 1096 mLock.lock(); 1097 } 1098 mLock.unlock(); 1099} 1100 1101status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1102{ 1103 const size_t SIZE = 256; 1104 char buffer[SIZE]; 1105 String8 result; 1106 1107 bool locked = tryLock(mLock); 1108 if (!locked) { 1109 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1110 write(fd, buffer, strlen(buffer)); 1111 } 1112 1113 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1114 result.append(buffer); 1115 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1116 result.append(buffer); 1117 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1122 result.append(buffer); 1123 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1124 result.append(buffer); 1125 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1126 result.append(buffer); 1127 1128 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1129 result.append(buffer); 1130 result.append(" Index Command"); 1131 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1132 snprintf(buffer, SIZE, "\n %02d ", i); 1133 result.append(buffer); 1134 result.append(mNewParameters[i]); 1135 } 1136 1137 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1138 result.append(buffer); 1139 snprintf(buffer, SIZE, " Index event param\n"); 1140 result.append(buffer); 1141 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1142 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1143 result.append(buffer); 1144 } 1145 result.append("\n"); 1146 1147 write(fd, result.string(), result.size()); 1148 1149 if (locked) { 1150 mLock.unlock(); 1151 } 1152 return NO_ERROR; 1153} 1154 1155status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1156{ 1157 const size_t SIZE = 256; 1158 char buffer[SIZE]; 1159 String8 result; 1160 1161 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1162 write(fd, buffer, strlen(buffer)); 1163 1164 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1165 sp<EffectChain> chain = mEffectChains[i]; 1166 if (chain != 0) { 1167 chain->dump(fd, args); 1168 } 1169 } 1170 return NO_ERROR; 1171} 1172 1173void AudioFlinger::ThreadBase::acquireWakeLock() 1174{ 1175 Mutex::Autolock _l(mLock); 1176 acquireWakeLock_l(); 1177} 1178 1179void AudioFlinger::ThreadBase::acquireWakeLock_l() 1180{ 1181 if (mPowerManager == 0) { 1182 // use checkService() to avoid blocking if power service is not up yet 1183 sp<IBinder> binder = 1184 defaultServiceManager()->checkService(String16("power")); 1185 if (binder == 0) { 1186 ALOGW("Thread %s cannot connect to the power manager service", mName); 1187 } else { 1188 mPowerManager = interface_cast<IPowerManager>(binder); 1189 binder->linkToDeath(mDeathRecipient); 1190 } 1191 } 1192 if (mPowerManager != 0) { 1193 sp<IBinder> binder = new BBinder(); 1194 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1195 binder, 1196 String16(mName)); 1197 if (status == NO_ERROR) { 1198 mWakeLockToken = binder; 1199 } 1200 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1201 } 1202} 1203 1204void AudioFlinger::ThreadBase::releaseWakeLock() 1205{ 1206 Mutex::Autolock _l(mLock); 1207 releaseWakeLock_l(); 1208} 1209 1210void AudioFlinger::ThreadBase::releaseWakeLock_l() 1211{ 1212 if (mWakeLockToken != 0) { 1213 ALOGV("releaseWakeLock_l() %s", mName); 1214 if (mPowerManager != 0) { 1215 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1216 } 1217 mWakeLockToken.clear(); 1218 } 1219} 1220 1221void AudioFlinger::ThreadBase::clearPowerManager() 1222{ 1223 Mutex::Autolock _l(mLock); 1224 releaseWakeLock_l(); 1225 mPowerManager.clear(); 1226} 1227 1228void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1229{ 1230 sp<ThreadBase> thread = mThread.promote(); 1231 if (thread != 0) { 1232 thread->clearPowerManager(); 1233 } 1234 ALOGW("power manager service died !!!"); 1235} 1236 1237void AudioFlinger::ThreadBase::setEffectSuspended( 1238 const effect_uuid_t *type, bool suspend, int sessionId) 1239{ 1240 Mutex::Autolock _l(mLock); 1241 setEffectSuspended_l(type, suspend, sessionId); 1242} 1243 1244void AudioFlinger::ThreadBase::setEffectSuspended_l( 1245 const effect_uuid_t *type, bool suspend, int sessionId) 1246{ 1247 sp<EffectChain> chain; 1248 chain = getEffectChain_l(sessionId); 1249 if (chain != 0) { 1250 if (type != NULL) { 1251 chain->setEffectSuspended_l(type, suspend); 1252 } else { 1253 chain->setEffectSuspendedAll_l(suspend); 1254 } 1255 } 1256 1257 updateSuspendedSessions_l(type, suspend, sessionId); 1258} 1259 1260void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1261{ 1262 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1263 if (index < 0) { 1264 return; 1265 } 1266 1267 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1268 mSuspendedSessions.editValueAt(index); 1269 1270 for (size_t i = 0; i < sessionEffects.size(); i++) { 1271 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1272 for (int j = 0; j < desc->mRefCount; j++) { 1273 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1274 chain->setEffectSuspendedAll_l(true); 1275 } else { 1276 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1277 desc->mType.timeLow); 1278 chain->setEffectSuspended_l(&desc->mType, true); 1279 } 1280 } 1281 } 1282} 1283 1284void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1285 bool suspend, 1286 int sessionId) 1287{ 1288 int index = mSuspendedSessions.indexOfKey(sessionId); 1289 1290 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1291 1292 if (suspend) { 1293 if (index >= 0) { 1294 sessionEffects = mSuspendedSessions.editValueAt(index); 1295 } else { 1296 mSuspendedSessions.add(sessionId, sessionEffects); 1297 } 1298 } else { 1299 if (index < 0) { 1300 return; 1301 } 1302 sessionEffects = mSuspendedSessions.editValueAt(index); 1303 } 1304 1305 1306 int key = EffectChain::kKeyForSuspendAll; 1307 if (type != NULL) { 1308 key = type->timeLow; 1309 } 1310 index = sessionEffects.indexOfKey(key); 1311 1312 sp <SuspendedSessionDesc> desc; 1313 if (suspend) { 1314 if (index >= 0) { 1315 desc = sessionEffects.valueAt(index); 1316 } else { 1317 desc = new SuspendedSessionDesc(); 1318 if (type != NULL) { 1319 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1320 } 1321 sessionEffects.add(key, desc); 1322 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1323 } 1324 desc->mRefCount++; 1325 } else { 1326 if (index < 0) { 1327 return; 1328 } 1329 desc = sessionEffects.valueAt(index); 1330 if (--desc->mRefCount == 0) { 1331 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1332 sessionEffects.removeItemsAt(index); 1333 if (sessionEffects.isEmpty()) { 1334 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1335 sessionId); 1336 mSuspendedSessions.removeItem(sessionId); 1337 } 1338 } 1339 } 1340 if (!sessionEffects.isEmpty()) { 1341 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1342 } 1343} 1344 1345void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1346 bool enabled, 1347 int sessionId) 1348{ 1349 Mutex::Autolock _l(mLock); 1350 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1351} 1352 1353void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1354 bool enabled, 1355 int sessionId) 1356{ 1357 if (mType != RECORD) { 1358 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1359 // another session. This gives the priority to well behaved effect control panels 1360 // and applications not using global effects. 1361 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1362 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1363 } 1364 } 1365 1366 sp<EffectChain> chain = getEffectChain_l(sessionId); 1367 if (chain != 0) { 1368 chain->checkSuspendOnEffectEnabled(effect, enabled); 1369 } 1370} 1371 1372// ---------------------------------------------------------------------------- 1373 1374AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1375 AudioStreamOut* output, 1376 int id, 1377 uint32_t device) 1378 : ThreadBase(audioFlinger, id, device), 1379 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1380 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1381{ 1382 snprintf(mName, kNameLength, "AudioOut_%d", id); 1383 1384 readOutputParameters(); 1385 1386 // Assumes constructor is called by AudioFlinger with it's mLock held, 1387 // but it would be safer to explicitly pass these as parameters 1388 mMasterVolume = mAudioFlinger->masterVolume_l(); 1389 mMasterMute = mAudioFlinger->masterMute_l(); 1390 1391 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1392 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1393 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1394 stream = (audio_stream_type_t) (stream + 1)) { 1395 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1396 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1397 // initialized by stream_type_t default constructor 1398 // mStreamTypes[stream].valid = true; 1399 } 1400} 1401 1402AudioFlinger::PlaybackThread::~PlaybackThread() 1403{ 1404 delete [] mMixBuffer; 1405} 1406 1407status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1408{ 1409 dumpInternals(fd, args); 1410 dumpTracks(fd, args); 1411 dumpEffectChains(fd, args); 1412 return NO_ERROR; 1413} 1414 1415status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1416{ 1417 const size_t SIZE = 256; 1418 char buffer[SIZE]; 1419 String8 result; 1420 1421 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1422 result.append(buffer); 1423 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1424 for (size_t i = 0; i < mTracks.size(); ++i) { 1425 sp<Track> track = mTracks[i]; 1426 if (track != 0) { 1427 track->dump(buffer, SIZE); 1428 result.append(buffer); 1429 } 1430 } 1431 1432 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1433 result.append(buffer); 1434 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1435 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1436 wp<Track> wTrack = mActiveTracks[i]; 1437 if (wTrack != 0) { 1438 sp<Track> track = wTrack.promote(); 1439 if (track != 0) { 1440 track->dump(buffer, SIZE); 1441 result.append(buffer); 1442 } 1443 } 1444 } 1445 write(fd, result.string(), result.size()); 1446 return NO_ERROR; 1447} 1448 1449status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1450{ 1451 const size_t SIZE = 256; 1452 char buffer[SIZE]; 1453 String8 result; 1454 1455 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1456 result.append(buffer); 1457 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1458 result.append(buffer); 1459 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1460 result.append(buffer); 1461 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1462 result.append(buffer); 1463 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1464 result.append(buffer); 1465 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1466 result.append(buffer); 1467 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1468 result.append(buffer); 1469 write(fd, result.string(), result.size()); 1470 1471 dumpBase(fd, args); 1472 1473 return NO_ERROR; 1474} 1475 1476// Thread virtuals 1477status_t AudioFlinger::PlaybackThread::readyToRun() 1478{ 1479 status_t status = initCheck(); 1480 if (status == NO_ERROR) { 1481 ALOGI("AudioFlinger's thread %p ready to run", this); 1482 } else { 1483 ALOGE("No working audio driver found."); 1484 } 1485 return status; 1486} 1487 1488void AudioFlinger::PlaybackThread::onFirstRef() 1489{ 1490 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1491} 1492 1493// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1494sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1495 const sp<AudioFlinger::Client>& client, 1496 audio_stream_type_t streamType, 1497 uint32_t sampleRate, 1498 audio_format_t format, 1499 uint32_t channelMask, 1500 int frameCount, 1501 const sp<IMemory>& sharedBuffer, 1502 int sessionId, 1503 status_t *status) 1504{ 1505 sp<Track> track; 1506 status_t lStatus; 1507 1508 if (mType == DIRECT) { 1509 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1510 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1511 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1512 "for output %p with format %d", 1513 sampleRate, format, channelMask, mOutput, mFormat); 1514 lStatus = BAD_VALUE; 1515 goto Exit; 1516 } 1517 } 1518 } else { 1519 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1520 if (sampleRate > mSampleRate*2) { 1521 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1522 lStatus = BAD_VALUE; 1523 goto Exit; 1524 } 1525 } 1526 1527 lStatus = initCheck(); 1528 if (lStatus != NO_ERROR) { 1529 ALOGE("Audio driver not initialized."); 1530 goto Exit; 1531 } 1532 1533 { // scope for mLock 1534 Mutex::Autolock _l(mLock); 1535 1536 // all tracks in same audio session must share the same routing strategy otherwise 1537 // conflicts will happen when tracks are moved from one output to another by audio policy 1538 // manager 1539 uint32_t strategy = 1540 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1541 for (size_t i = 0; i < mTracks.size(); ++i) { 1542 sp<Track> t = mTracks[i]; 1543 if (t != 0) { 1544 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1545 if (sessionId == t->sessionId() && strategy != actual) { 1546 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1547 strategy, actual); 1548 lStatus = BAD_VALUE; 1549 goto Exit; 1550 } 1551 } 1552 } 1553 1554 track = new Track(this, client, streamType, sampleRate, format, 1555 channelMask, frameCount, sharedBuffer, sessionId); 1556 if (track->getCblk() == NULL || track->name() < 0) { 1557 lStatus = NO_MEMORY; 1558 goto Exit; 1559 } 1560 mTracks.add(track); 1561 1562 sp<EffectChain> chain = getEffectChain_l(sessionId); 1563 if (chain != 0) { 1564 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1565 track->setMainBuffer(chain->inBuffer()); 1566 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1567 chain->incTrackCnt(); 1568 } 1569 1570 // invalidate track immediately if the stream type was moved to another thread since 1571 // createTrack() was called by the client process. 1572 if (!mStreamTypes[streamType].valid) { 1573 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1574 this, streamType); 1575 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1576 } 1577 } 1578 lStatus = NO_ERROR; 1579 1580Exit: 1581 if(status) { 1582 *status = lStatus; 1583 } 1584 return track; 1585} 1586 1587uint32_t AudioFlinger::PlaybackThread::latency() const 1588{ 1589 Mutex::Autolock _l(mLock); 1590 if (initCheck() == NO_ERROR) { 1591 return mOutput->stream->get_latency(mOutput->stream); 1592 } else { 1593 return 0; 1594 } 1595} 1596 1597status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1598{ 1599 mMasterVolume = value; 1600 return NO_ERROR; 1601} 1602 1603status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1604{ 1605 mMasterMute = muted; 1606 return NO_ERROR; 1607} 1608 1609float AudioFlinger::PlaybackThread::masterVolume() const 1610{ 1611 return mMasterVolume; 1612} 1613 1614bool AudioFlinger::PlaybackThread::masterMute() const 1615{ 1616 return mMasterMute; 1617} 1618 1619status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1620{ 1621 mStreamTypes[stream].volume = value; 1622 return NO_ERROR; 1623} 1624 1625status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1626{ 1627 mStreamTypes[stream].mute = muted; 1628 return NO_ERROR; 1629} 1630 1631float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1632{ 1633 return mStreamTypes[stream].volume; 1634} 1635 1636bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1637{ 1638 return mStreamTypes[stream].mute; 1639} 1640 1641// addTrack_l() must be called with ThreadBase::mLock held 1642status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1643{ 1644 status_t status = ALREADY_EXISTS; 1645 1646 // set retry count for buffer fill 1647 track->mRetryCount = kMaxTrackStartupRetries; 1648 if (mActiveTracks.indexOf(track) < 0) { 1649 // the track is newly added, make sure it fills up all its 1650 // buffers before playing. This is to ensure the client will 1651 // effectively get the latency it requested. 1652 track->mFillingUpStatus = Track::FS_FILLING; 1653 track->mResetDone = false; 1654 mActiveTracks.add(track); 1655 if (track->mainBuffer() != mMixBuffer) { 1656 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1657 if (chain != 0) { 1658 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1659 chain->incActiveTrackCnt(); 1660 } 1661 } 1662 1663 status = NO_ERROR; 1664 } 1665 1666 ALOGV("mWaitWorkCV.broadcast"); 1667 mWaitWorkCV.broadcast(); 1668 1669 return status; 1670} 1671 1672// destroyTrack_l() must be called with ThreadBase::mLock held 1673void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1674{ 1675 track->mState = TrackBase::TERMINATED; 1676 if (mActiveTracks.indexOf(track) < 0) { 1677 removeTrack_l(track); 1678 } 1679} 1680 1681void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1682{ 1683 mTracks.remove(track); 1684 deleteTrackName_l(track->name()); 1685 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1686 if (chain != 0) { 1687 chain->decTrackCnt(); 1688 } 1689} 1690 1691String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1692{ 1693 String8 out_s8 = String8(""); 1694 char *s; 1695 1696 Mutex::Autolock _l(mLock); 1697 if (initCheck() != NO_ERROR) { 1698 return out_s8; 1699 } 1700 1701 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1702 out_s8 = String8(s); 1703 free(s); 1704 return out_s8; 1705} 1706 1707// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1708void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1709 AudioSystem::OutputDescriptor desc; 1710 void *param2 = 0; 1711 1712 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1713 1714 switch (event) { 1715 case AudioSystem::OUTPUT_OPENED: 1716 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1717 desc.channels = mChannelMask; 1718 desc.samplingRate = mSampleRate; 1719 desc.format = mFormat; 1720 desc.frameCount = mFrameCount; 1721 desc.latency = latency(); 1722 param2 = &desc; 1723 break; 1724 1725 case AudioSystem::STREAM_CONFIG_CHANGED: 1726 param2 = ¶m; 1727 case AudioSystem::OUTPUT_CLOSED: 1728 default: 1729 break; 1730 } 1731 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1732} 1733 1734void AudioFlinger::PlaybackThread::readOutputParameters() 1735{ 1736 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1737 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1738 mChannelCount = (uint16_t)popcount(mChannelMask); 1739 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1740 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1741 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1742 1743 // FIXME - Current mixer implementation only supports stereo output: Always 1744 // Allocate a stereo buffer even if HW output is mono. 1745 if (mMixBuffer != NULL) delete[] mMixBuffer; 1746 mMixBuffer = new int16_t[mFrameCount * 2]; 1747 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1748 1749 // force reconfiguration of effect chains and engines to take new buffer size and audio 1750 // parameters into account 1751 // Note that mLock is not held when readOutputParameters() is called from the constructor 1752 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1753 // matter. 1754 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1755 Vector< sp<EffectChain> > effectChains = mEffectChains; 1756 for (size_t i = 0; i < effectChains.size(); i ++) { 1757 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1758 } 1759} 1760 1761status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1762{ 1763 if (halFrames == 0 || dspFrames == 0) { 1764 return BAD_VALUE; 1765 } 1766 Mutex::Autolock _l(mLock); 1767 if (initCheck() != NO_ERROR) { 1768 return INVALID_OPERATION; 1769 } 1770 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1771 1772 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1773} 1774 1775uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1776{ 1777 Mutex::Autolock _l(mLock); 1778 uint32_t result = 0; 1779 if (getEffectChain_l(sessionId) != 0) { 1780 result = EFFECT_SESSION; 1781 } 1782 1783 for (size_t i = 0; i < mTracks.size(); ++i) { 1784 sp<Track> track = mTracks[i]; 1785 if (sessionId == track->sessionId() && 1786 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1787 result |= TRACK_SESSION; 1788 break; 1789 } 1790 } 1791 1792 return result; 1793} 1794 1795uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1796{ 1797 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1798 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1799 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1800 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1801 } 1802 for (size_t i = 0; i < mTracks.size(); i++) { 1803 sp<Track> track = mTracks[i]; 1804 if (sessionId == track->sessionId() && 1805 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1806 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1807 } 1808 } 1809 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1810} 1811 1812 1813AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() 1814{ 1815 Mutex::Autolock _l(mLock); 1816 return mOutput; 1817} 1818 1819AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1820{ 1821 Mutex::Autolock _l(mLock); 1822 AudioStreamOut *output = mOutput; 1823 mOutput = NULL; 1824 return output; 1825} 1826 1827// this method must always be called either with ThreadBase mLock held or inside the thread loop 1828audio_stream_t* AudioFlinger::PlaybackThread::stream() 1829{ 1830 if (mOutput == NULL) { 1831 return NULL; 1832 } 1833 return &mOutput->stream->common; 1834} 1835 1836uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1837{ 1838 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1839 // decoding and transfer time. So sleeping for half of the latency would likely cause 1840 // underruns 1841 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1842 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1843 } else { 1844 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1845 } 1846} 1847 1848// ---------------------------------------------------------------------------- 1849 1850AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1851 : PlaybackThread(audioFlinger, output, id, device), 1852 mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE) 1853{ 1854 mType = ThreadBase::MIXER; 1855 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1856 1857 // FIXME - Current mixer implementation only supports stereo output 1858 if (mChannelCount == 1) { 1859 ALOGE("Invalid audio hardware channel count"); 1860 } 1861} 1862 1863AudioFlinger::MixerThread::~MixerThread() 1864{ 1865 delete mAudioMixer; 1866} 1867 1868bool AudioFlinger::MixerThread::threadLoop() 1869{ 1870 Vector< sp<Track> > tracksToRemove; 1871 uint32_t mixerStatus = MIXER_IDLE; 1872 nsecs_t standbyTime = systemTime(); 1873 size_t mixBufferSize = mFrameCount * mFrameSize; 1874 // FIXME: Relaxed timing because of a certain device that can't meet latency 1875 // Should be reduced to 2x after the vendor fixes the driver issue 1876 // increase threshold again due to low power audio mode. The way this warning threshold is 1877 // calculated and its usefulness should be reconsidered anyway. 1878 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1879 nsecs_t lastWarning = 0; 1880 bool longStandbyExit = false; 1881 uint32_t activeSleepTime = activeSleepTimeUs(); 1882 uint32_t idleSleepTime = idleSleepTimeUs(); 1883 uint32_t sleepTime = idleSleepTime; 1884 uint32_t sleepTimeShift = 0; 1885 Vector< sp<EffectChain> > effectChains; 1886#ifdef DEBUG_CPU_USAGE 1887 ThreadCpuUsage cpu; 1888 const CentralTendencyStatistics& stats = cpu.statistics(); 1889#endif 1890 1891 acquireWakeLock(); 1892 1893 while (!exitPending()) 1894 { 1895#ifdef DEBUG_CPU_USAGE 1896 cpu.sampleAndEnable(); 1897 unsigned n = stats.n(); 1898 // cpu.elapsed() is expensive, so don't call it every loop 1899 if ((n & 127) == 1) { 1900 long long elapsed = cpu.elapsed(); 1901 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1902 double perLoop = elapsed / (double) n; 1903 double perLoop100 = perLoop * 0.01; 1904 double mean = stats.mean(); 1905 double stddev = stats.stddev(); 1906 double minimum = stats.minimum(); 1907 double maximum = stats.maximum(); 1908 cpu.resetStatistics(); 1909 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", 1910 elapsed * .000000001, n, perLoop * .000001, 1911 mean * .001, 1912 stddev * .001, 1913 minimum * .001, 1914 maximum * .001, 1915 mean / perLoop100, 1916 stddev / perLoop100, 1917 minimum / perLoop100, 1918 maximum / perLoop100); 1919 } 1920 } 1921#endif 1922 processConfigEvents(); 1923 1924 mixerStatus = MIXER_IDLE; 1925 { // scope for mLock 1926 1927 Mutex::Autolock _l(mLock); 1928 1929 if (checkForNewParameters_l()) { 1930 mixBufferSize = mFrameCount * mFrameSize; 1931 // FIXME: Relaxed timing because of a certain device that can't meet latency 1932 // Should be reduced to 2x after the vendor fixes the driver issue 1933 // increase threshold again due to low power audio mode. The way this warning 1934 // threshold is calculated and its usefulness should be reconsidered anyway. 1935 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1936 activeSleepTime = activeSleepTimeUs(); 1937 idleSleepTime = idleSleepTimeUs(); 1938 } 1939 1940 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1941 1942 // put audio hardware into standby after short delay 1943 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1944 mSuspended)) { 1945 if (!mStandby) { 1946 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1947 mOutput->stream->common.standby(&mOutput->stream->common); 1948 mStandby = true; 1949 mBytesWritten = 0; 1950 } 1951 1952 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1953 // we're about to wait, flush the binder command buffer 1954 IPCThreadState::self()->flushCommands(); 1955 1956 if (exitPending()) break; 1957 1958 releaseWakeLock_l(); 1959 // wait until we have something to do... 1960 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1961 mWaitWorkCV.wait(mLock); 1962 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1963 acquireWakeLock_l(); 1964 1965 mPrevMixerStatus = MIXER_IDLE; 1966 if (!mMasterMute) { 1967 char value[PROPERTY_VALUE_MAX]; 1968 property_get("ro.audio.silent", value, "0"); 1969 if (atoi(value)) { 1970 ALOGD("Silence is golden"); 1971 setMasterMute(true); 1972 } 1973 } 1974 1975 standbyTime = systemTime() + kStandbyTimeInNsecs; 1976 sleepTime = idleSleepTime; 1977 sleepTimeShift = 0; 1978 continue; 1979 } 1980 } 1981 1982 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1983 1984 // prevent any changes in effect chain list and in each effect chain 1985 // during mixing and effect process as the audio buffers could be deleted 1986 // or modified if an effect is created or deleted 1987 lockEffectChains_l(effectChains); 1988 } 1989 1990 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1991 // mix buffers... 1992 mAudioMixer->process(); 1993 sleepTime = 0; 1994 // increase sleep time progressively when application underrun condition clears 1995 if (sleepTimeShift > 0) { 1996 sleepTimeShift--; 1997 } 1998 standbyTime = systemTime() + kStandbyTimeInNsecs; 1999 //TODO: delay standby when effects have a tail 2000 } else { 2001 // If no tracks are ready, sleep once for the duration of an output 2002 // buffer size, then write 0s to the output 2003 if (sleepTime == 0) { 2004 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2005 sleepTime = activeSleepTime >> sleepTimeShift; 2006 if (sleepTime < kMinThreadSleepTimeUs) { 2007 sleepTime = kMinThreadSleepTimeUs; 2008 } 2009 // reduce sleep time in case of consecutive application underruns to avoid 2010 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2011 // duration we would end up writing less data than needed by the audio HAL if 2012 // the condition persists. 2013 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2014 sleepTimeShift++; 2015 } 2016 } else { 2017 sleepTime = idleSleepTime; 2018 } 2019 } else if (mBytesWritten != 0 || 2020 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2021 memset (mMixBuffer, 0, mixBufferSize); 2022 sleepTime = 0; 2023 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2024 } 2025 // TODO add standby time extension fct of effect tail 2026 } 2027 2028 if (mSuspended) { 2029 sleepTime = suspendSleepTimeUs(); 2030 } 2031 // sleepTime == 0 means we must write to audio hardware 2032 if (sleepTime == 0) { 2033 for (size_t i = 0; i < effectChains.size(); i ++) { 2034 effectChains[i]->process_l(); 2035 } 2036 // enable changes in effect chain 2037 unlockEffectChains(effectChains); 2038 mLastWriteTime = systemTime(); 2039 mInWrite = true; 2040 mBytesWritten += mixBufferSize; 2041 2042 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2043 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2044 mNumWrites++; 2045 mInWrite = false; 2046 nsecs_t now = systemTime(); 2047 nsecs_t delta = now - mLastWriteTime; 2048 if (!mStandby && delta > maxPeriod) { 2049 mNumDelayedWrites++; 2050 if ((now - lastWarning) > kWarningThrottleNs) { 2051 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2052 ns2ms(delta), mNumDelayedWrites, this); 2053 lastWarning = now; 2054 } 2055 if (mStandby) { 2056 longStandbyExit = true; 2057 } 2058 } 2059 mStandby = false; 2060 } else { 2061 // enable changes in effect chain 2062 unlockEffectChains(effectChains); 2063 usleep(sleepTime); 2064 } 2065 2066 // finally let go of all our tracks, without the lock held 2067 // since we can't guarantee the destructors won't acquire that 2068 // same lock. 2069 tracksToRemove.clear(); 2070 2071 // Effect chains will be actually deleted here if they were removed from 2072 // mEffectChains list during mixing or effects processing 2073 effectChains.clear(); 2074 } 2075 2076 if (!mStandby) { 2077 mOutput->stream->common.standby(&mOutput->stream->common); 2078 } 2079 2080 releaseWakeLock(); 2081 2082 ALOGV("MixerThread %p exiting", this); 2083 return false; 2084} 2085 2086// prepareTracks_l() must be called with ThreadBase::mLock held 2087uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2088{ 2089 2090 uint32_t mixerStatus = MIXER_IDLE; 2091 // find out which tracks need to be processed 2092 size_t count = activeTracks.size(); 2093 size_t mixedTracks = 0; 2094 size_t tracksWithEffect = 0; 2095 2096 float masterVolume = mMasterVolume; 2097 bool masterMute = mMasterMute; 2098 2099 if (masterMute) { 2100 masterVolume = 0; 2101 } 2102 // Delegate master volume control to effect in output mix effect chain if needed 2103 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2104 if (chain != 0) { 2105 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2106 chain->setVolume_l(&v, &v); 2107 masterVolume = (float)((v + (1 << 23)) >> 24); 2108 chain.clear(); 2109 } 2110 2111 for (size_t i=0 ; i<count ; i++) { 2112 sp<Track> t = activeTracks[i].promote(); 2113 if (t == 0) continue; 2114 2115 // this const just means the local variable doesn't change 2116 Track* const track = t.get(); 2117 audio_track_cblk_t* cblk = track->cblk(); 2118 2119 // The first time a track is added we wait 2120 // for all its buffers to be filled before processing it 2121 int name = track->name(); 2122 // make sure that we have enough frames to mix one full buffer. 2123 // enforce this condition only once to enable draining the buffer in case the client 2124 // app does not call stop() and relies on underrun to stop: 2125 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2126 // during last round 2127 uint32_t minFrames = 1; 2128 if (!track->isStopped() && !track->isPausing() && 2129 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2130 if (t->sampleRate() == (int)mSampleRate) { 2131 minFrames = mFrameCount; 2132 } else { 2133 // +1 for rounding and +1 for additional sample needed for interpolation 2134 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2135 // add frames already consumed but not yet released by the resampler 2136 // because cblk->framesReady() will include these frames 2137 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2138 // the minimum track buffer size is normally twice the number of frames necessary 2139 // to fill one buffer and the resampler should not leave more than one buffer worth 2140 // of unreleased frames after each pass, but just in case... 2141 ALOG_ASSERT(minFrames <= cblk->frameCount); 2142 } 2143 } 2144 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2145 !track->isPaused() && !track->isTerminated()) 2146 { 2147 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2148 2149 mixedTracks++; 2150 2151 // track->mainBuffer() != mMixBuffer means there is an effect chain 2152 // connected to the track 2153 chain.clear(); 2154 if (track->mainBuffer() != mMixBuffer) { 2155 chain = getEffectChain_l(track->sessionId()); 2156 // Delegate volume control to effect in track effect chain if needed 2157 if (chain != 0) { 2158 tracksWithEffect++; 2159 } else { 2160 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2161 name, track->sessionId()); 2162 } 2163 } 2164 2165 2166 int param = AudioMixer::VOLUME; 2167 if (track->mFillingUpStatus == Track::FS_FILLED) { 2168 // no ramp for the first volume setting 2169 track->mFillingUpStatus = Track::FS_ACTIVE; 2170 if (track->mState == TrackBase::RESUMING) { 2171 track->mState = TrackBase::ACTIVE; 2172 param = AudioMixer::RAMP_VOLUME; 2173 } 2174 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2175 } else if (cblk->server != 0) { 2176 // If the track is stopped before the first frame was mixed, 2177 // do not apply ramp 2178 param = AudioMixer::RAMP_VOLUME; 2179 } 2180 2181 // compute volume for this track 2182 uint32_t vl, vr, va; 2183 if (track->isMuted() || track->isPausing() || 2184 mStreamTypes[track->type()].mute) { 2185 vl = vr = va = 0; 2186 if (track->isPausing()) { 2187 track->setPaused(); 2188 } 2189 } else { 2190 2191 // read original volumes with volume control 2192 float typeVolume = mStreamTypes[track->type()].volume; 2193 float v = masterVolume * typeVolume; 2194 uint32_t vlr = cblk->getVolumeLR(); 2195 vl = vlr & 0xFFFF; 2196 vr = vlr >> 16; 2197 // track volumes come from shared memory, so can't be trusted and must be clamped 2198 if (vl > MAX_GAIN_INT) { 2199 ALOGV("Track left volume out of range: %04X", vl); 2200 vl = MAX_GAIN_INT; 2201 } 2202 if (vr > MAX_GAIN_INT) { 2203 ALOGV("Track right volume out of range: %04X", vr); 2204 vr = MAX_GAIN_INT; 2205 } 2206 // now apply the master volume and stream type volume 2207 vl = (uint32_t)(v * vl) << 12; 2208 vr = (uint32_t)(v * vr) << 12; 2209 // assuming master volume and stream type volume each go up to 1.0, 2210 // vl and vr are now in 8.24 format 2211 2212 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2213 // send level comes from shared memory and so may be corrupt 2214 if (sendLevel >= MAX_GAIN_INT) { 2215 ALOGV("Track send level out of range: %04X", sendLevel); 2216 sendLevel = MAX_GAIN_INT; 2217 } 2218 va = (uint32_t)(v * sendLevel); 2219 } 2220 // Delegate volume control to effect in track effect chain if needed 2221 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2222 // Do not ramp volume if volume is controlled by effect 2223 param = AudioMixer::VOLUME; 2224 track->mHasVolumeController = true; 2225 } else { 2226 // force no volume ramp when volume controller was just disabled or removed 2227 // from effect chain to avoid volume spike 2228 if (track->mHasVolumeController) { 2229 param = AudioMixer::VOLUME; 2230 } 2231 track->mHasVolumeController = false; 2232 } 2233 2234 // Convert volumes from 8.24 to 4.12 format 2235 int16_t left, right, aux; 2236 // This additional clamping is needed in case chain->setVolume_l() overshot 2237 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2238 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2239 left = int16_t(v_clamped); 2240 v_clamped = (vr + (1 << 11)) >> 12; 2241 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2242 right = int16_t(v_clamped); 2243 2244 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2245 aux = int16_t(va); 2246 2247 // XXX: these things DON'T need to be done each time 2248 mAudioMixer->setBufferProvider(name, track); 2249 mAudioMixer->enable(name); 2250 2251 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2252 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2253 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2254 mAudioMixer->setParameter( 2255 name, 2256 AudioMixer::TRACK, 2257 AudioMixer::FORMAT, (void *)track->format()); 2258 mAudioMixer->setParameter( 2259 name, 2260 AudioMixer::TRACK, 2261 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2262 mAudioMixer->setParameter( 2263 name, 2264 AudioMixer::RESAMPLE, 2265 AudioMixer::SAMPLE_RATE, 2266 (void *)(cblk->sampleRate)); 2267 mAudioMixer->setParameter( 2268 name, 2269 AudioMixer::TRACK, 2270 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2271 mAudioMixer->setParameter( 2272 name, 2273 AudioMixer::TRACK, 2274 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2275 2276 // reset retry count 2277 track->mRetryCount = kMaxTrackRetries; 2278 // If one track is ready, set the mixer ready if: 2279 // - the mixer was not ready during previous round OR 2280 // - no other track is not ready 2281 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2282 mixerStatus != MIXER_TRACKS_ENABLED) { 2283 mixerStatus = MIXER_TRACKS_READY; 2284 } 2285 } else { 2286 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2287 if (track->isStopped()) { 2288 track->reset(); 2289 } 2290 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2291 // We have consumed all the buffers of this track. 2292 // Remove it from the list of active tracks. 2293 tracksToRemove->add(track); 2294 } else { 2295 // No buffers for this track. Give it a few chances to 2296 // fill a buffer, then remove it from active list. 2297 if (--(track->mRetryCount) <= 0) { 2298 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2299 tracksToRemove->add(track); 2300 // indicate to client process that the track was disabled because of underrun 2301 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2302 // If one track is not ready, mark the mixer also not ready if: 2303 // - the mixer was ready during previous round OR 2304 // - no other track is ready 2305 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2306 mixerStatus != MIXER_TRACKS_READY) { 2307 mixerStatus = MIXER_TRACKS_ENABLED; 2308 } 2309 } 2310 mAudioMixer->disable(name); 2311 } 2312 } 2313 2314 // remove all the tracks that need to be... 2315 count = tracksToRemove->size(); 2316 if (CC_UNLIKELY(count)) { 2317 for (size_t i=0 ; i<count ; i++) { 2318 const sp<Track>& track = tracksToRemove->itemAt(i); 2319 mActiveTracks.remove(track); 2320 if (track->mainBuffer() != mMixBuffer) { 2321 chain = getEffectChain_l(track->sessionId()); 2322 if (chain != 0) { 2323 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2324 chain->decActiveTrackCnt(); 2325 } 2326 } 2327 if (track->isTerminated()) { 2328 removeTrack_l(track); 2329 } 2330 } 2331 } 2332 2333 // mix buffer must be cleared if all tracks are connected to an 2334 // effect chain as in this case the mixer will not write to 2335 // mix buffer and track effects will accumulate into it 2336 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2337 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2338 } 2339 2340 mPrevMixerStatus = mixerStatus; 2341 return mixerStatus; 2342} 2343 2344void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2345{ 2346 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2347 this, streamType, mTracks.size()); 2348 Mutex::Autolock _l(mLock); 2349 2350 size_t size = mTracks.size(); 2351 for (size_t i = 0; i < size; i++) { 2352 sp<Track> t = mTracks[i]; 2353 if (t->type() == streamType) { 2354 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2355 t->mCblk->cv.signal(); 2356 } 2357 } 2358} 2359 2360void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2361{ 2362 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2363 this, streamType, valid); 2364 Mutex::Autolock _l(mLock); 2365 2366 mStreamTypes[streamType].valid = valid; 2367} 2368 2369// getTrackName_l() must be called with ThreadBase::mLock held 2370int AudioFlinger::MixerThread::getTrackName_l() 2371{ 2372 return mAudioMixer->getTrackName(); 2373} 2374 2375// deleteTrackName_l() must be called with ThreadBase::mLock held 2376void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2377{ 2378 ALOGV("remove track (%d) and delete from mixer", name); 2379 mAudioMixer->deleteTrackName(name); 2380} 2381 2382// checkForNewParameters_l() must be called with ThreadBase::mLock held 2383bool AudioFlinger::MixerThread::checkForNewParameters_l() 2384{ 2385 bool reconfig = false; 2386 2387 while (!mNewParameters.isEmpty()) { 2388 status_t status = NO_ERROR; 2389 String8 keyValuePair = mNewParameters[0]; 2390 AudioParameter param = AudioParameter(keyValuePair); 2391 int value; 2392 2393 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2394 reconfig = true; 2395 } 2396 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2397 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2398 status = BAD_VALUE; 2399 } else { 2400 reconfig = true; 2401 } 2402 } 2403 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2404 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2405 status = BAD_VALUE; 2406 } else { 2407 reconfig = true; 2408 } 2409 } 2410 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2411 // do not accept frame count changes if tracks are open as the track buffer 2412 // size depends on frame count and correct behavior would not be guaranteed 2413 // if frame count is changed after track creation 2414 if (!mTracks.isEmpty()) { 2415 status = INVALID_OPERATION; 2416 } else { 2417 reconfig = true; 2418 } 2419 } 2420 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2421 // when changing the audio output device, call addBatteryData to notify 2422 // the change 2423 if ((int)mDevice != value) { 2424 uint32_t params = 0; 2425 // check whether speaker is on 2426 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2427 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2428 } 2429 2430 int deviceWithoutSpeaker 2431 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2432 // check if any other device (except speaker) is on 2433 if (value & deviceWithoutSpeaker ) { 2434 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2435 } 2436 2437 if (params != 0) { 2438 addBatteryData(params); 2439 } 2440 } 2441 2442 // forward device change to effects that have requested to be 2443 // aware of attached audio device. 2444 mDevice = (uint32_t)value; 2445 for (size_t i = 0; i < mEffectChains.size(); i++) { 2446 mEffectChains[i]->setDevice_l(mDevice); 2447 } 2448 } 2449 2450 if (status == NO_ERROR) { 2451 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2452 keyValuePair.string()); 2453 if (!mStandby && status == INVALID_OPERATION) { 2454 mOutput->stream->common.standby(&mOutput->stream->common); 2455 mStandby = true; 2456 mBytesWritten = 0; 2457 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2458 keyValuePair.string()); 2459 } 2460 if (status == NO_ERROR && reconfig) { 2461 delete mAudioMixer; 2462 readOutputParameters(); 2463 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2464 for (size_t i = 0; i < mTracks.size() ; i++) { 2465 int name = getTrackName_l(); 2466 if (name < 0) break; 2467 mTracks[i]->mName = name; 2468 // limit track sample rate to 2 x new output sample rate 2469 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2470 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2471 } 2472 } 2473 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2474 } 2475 } 2476 2477 mNewParameters.removeAt(0); 2478 2479 mParamStatus = status; 2480 mParamCond.signal(); 2481 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2482 // already timed out waiting for the status and will never signal the condition. 2483 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2484 } 2485 return reconfig; 2486} 2487 2488status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2489{ 2490 const size_t SIZE = 256; 2491 char buffer[SIZE]; 2492 String8 result; 2493 2494 PlaybackThread::dumpInternals(fd, args); 2495 2496 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2497 result.append(buffer); 2498 write(fd, result.string(), result.size()); 2499 return NO_ERROR; 2500} 2501 2502uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2503{ 2504 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2505} 2506 2507uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2508{ 2509 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2510} 2511 2512// ---------------------------------------------------------------------------- 2513AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2514 : PlaybackThread(audioFlinger, output, id, device) 2515{ 2516 mType = ThreadBase::DIRECT; 2517} 2518 2519AudioFlinger::DirectOutputThread::~DirectOutputThread() 2520{ 2521} 2522 2523static inline 2524int32_t mul(int16_t in, int16_t v) 2525{ 2526#if defined(__arm__) && !defined(__thumb__) 2527 int32_t out; 2528 asm( "smulbb %[out], %[in], %[v] \n" 2529 : [out]"=r"(out) 2530 : [in]"%r"(in), [v]"r"(v) 2531 : ); 2532 return out; 2533#else 2534 return in * int32_t(v); 2535#endif 2536} 2537 2538void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2539{ 2540 // Do not apply volume on compressed audio 2541 if (!audio_is_linear_pcm(mFormat)) { 2542 return; 2543 } 2544 2545 // convert to signed 16 bit before volume calculation 2546 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2547 size_t count = mFrameCount * mChannelCount; 2548 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2549 int16_t *dst = mMixBuffer + count-1; 2550 while(count--) { 2551 *dst-- = (int16_t)(*src--^0x80) << 8; 2552 } 2553 } 2554 2555 size_t frameCount = mFrameCount; 2556 int16_t *out = mMixBuffer; 2557 if (ramp) { 2558 if (mChannelCount == 1) { 2559 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2560 int32_t vlInc = d / (int32_t)frameCount; 2561 int32_t vl = ((int32_t)mLeftVolShort << 16); 2562 do { 2563 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2564 out++; 2565 vl += vlInc; 2566 } while (--frameCount); 2567 2568 } else { 2569 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2570 int32_t vlInc = d / (int32_t)frameCount; 2571 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2572 int32_t vrInc = d / (int32_t)frameCount; 2573 int32_t vl = ((int32_t)mLeftVolShort << 16); 2574 int32_t vr = ((int32_t)mRightVolShort << 16); 2575 do { 2576 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2577 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2578 out += 2; 2579 vl += vlInc; 2580 vr += vrInc; 2581 } while (--frameCount); 2582 } 2583 } else { 2584 if (mChannelCount == 1) { 2585 do { 2586 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2587 out++; 2588 } while (--frameCount); 2589 } else { 2590 do { 2591 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2592 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2593 out += 2; 2594 } while (--frameCount); 2595 } 2596 } 2597 2598 // convert back to unsigned 8 bit after volume calculation 2599 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2600 size_t count = mFrameCount * mChannelCount; 2601 int16_t *src = mMixBuffer; 2602 uint8_t *dst = (uint8_t *)mMixBuffer; 2603 while(count--) { 2604 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2605 } 2606 } 2607 2608 mLeftVolShort = leftVol; 2609 mRightVolShort = rightVol; 2610} 2611 2612bool AudioFlinger::DirectOutputThread::threadLoop() 2613{ 2614 uint32_t mixerStatus = MIXER_IDLE; 2615 sp<Track> trackToRemove; 2616 sp<Track> activeTrack; 2617 nsecs_t standbyTime = systemTime(); 2618 int8_t *curBuf; 2619 size_t mixBufferSize = mFrameCount*mFrameSize; 2620 uint32_t activeSleepTime = activeSleepTimeUs(); 2621 uint32_t idleSleepTime = idleSleepTimeUs(); 2622 uint32_t sleepTime = idleSleepTime; 2623 // use shorter standby delay as on normal output to release 2624 // hardware resources as soon as possible 2625 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2626 2627 acquireWakeLock(); 2628 2629 while (!exitPending()) 2630 { 2631 bool rampVolume; 2632 uint16_t leftVol; 2633 uint16_t rightVol; 2634 Vector< sp<EffectChain> > effectChains; 2635 2636 processConfigEvents(); 2637 2638 mixerStatus = MIXER_IDLE; 2639 2640 { // scope for the mLock 2641 2642 Mutex::Autolock _l(mLock); 2643 2644 if (checkForNewParameters_l()) { 2645 mixBufferSize = mFrameCount*mFrameSize; 2646 activeSleepTime = activeSleepTimeUs(); 2647 idleSleepTime = idleSleepTimeUs(); 2648 standbyDelay = microseconds(activeSleepTime*2); 2649 } 2650 2651 // put audio hardware into standby after short delay 2652 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2653 mSuspended)) { 2654 // wait until we have something to do... 2655 if (!mStandby) { 2656 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2657 mOutput->stream->common.standby(&mOutput->stream->common); 2658 mStandby = true; 2659 mBytesWritten = 0; 2660 } 2661 2662 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2663 // we're about to wait, flush the binder command buffer 2664 IPCThreadState::self()->flushCommands(); 2665 2666 if (exitPending()) break; 2667 2668 releaseWakeLock_l(); 2669 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2670 mWaitWorkCV.wait(mLock); 2671 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2672 acquireWakeLock_l(); 2673 2674 if (!mMasterMute) { 2675 char value[PROPERTY_VALUE_MAX]; 2676 property_get("ro.audio.silent", value, "0"); 2677 if (atoi(value)) { 2678 ALOGD("Silence is golden"); 2679 setMasterMute(true); 2680 } 2681 } 2682 2683 standbyTime = systemTime() + standbyDelay; 2684 sleepTime = idleSleepTime; 2685 continue; 2686 } 2687 } 2688 2689 effectChains = mEffectChains; 2690 2691 // find out which tracks need to be processed 2692 if (mActiveTracks.size() != 0) { 2693 sp<Track> t = mActiveTracks[0].promote(); 2694 if (t == 0) continue; 2695 2696 Track* const track = t.get(); 2697 audio_track_cblk_t* cblk = track->cblk(); 2698 2699 // The first time a track is added we wait 2700 // for all its buffers to be filled before processing it 2701 if (cblk->framesReady() && track->isReady() && 2702 !track->isPaused() && !track->isTerminated()) 2703 { 2704 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2705 2706 if (track->mFillingUpStatus == Track::FS_FILLED) { 2707 track->mFillingUpStatus = Track::FS_ACTIVE; 2708 mLeftVolFloat = mRightVolFloat = 0; 2709 mLeftVolShort = mRightVolShort = 0; 2710 if (track->mState == TrackBase::RESUMING) { 2711 track->mState = TrackBase::ACTIVE; 2712 rampVolume = true; 2713 } 2714 } else if (cblk->server != 0) { 2715 // If the track is stopped before the first frame was mixed, 2716 // do not apply ramp 2717 rampVolume = true; 2718 } 2719 // compute volume for this track 2720 float left, right; 2721 if (track->isMuted() || mMasterMute || track->isPausing() || 2722 mStreamTypes[track->type()].mute) { 2723 left = right = 0; 2724 if (track->isPausing()) { 2725 track->setPaused(); 2726 } 2727 } else { 2728 float typeVolume = mStreamTypes[track->type()].volume; 2729 float v = mMasterVolume * typeVolume; 2730 uint32_t vlr = cblk->getVolumeLR(); 2731 float v_clamped = v * (vlr & 0xFFFF); 2732 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2733 left = v_clamped/MAX_GAIN; 2734 v_clamped = v * (vlr >> 16); 2735 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2736 right = v_clamped/MAX_GAIN; 2737 } 2738 2739 if (left != mLeftVolFloat || right != mRightVolFloat) { 2740 mLeftVolFloat = left; 2741 mRightVolFloat = right; 2742 2743 // If audio HAL implements volume control, 2744 // force software volume to nominal value 2745 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2746 left = 1.0f; 2747 right = 1.0f; 2748 } 2749 2750 // Convert volumes from float to 8.24 2751 uint32_t vl = (uint32_t)(left * (1 << 24)); 2752 uint32_t vr = (uint32_t)(right * (1 << 24)); 2753 2754 // Delegate volume control to effect in track effect chain if needed 2755 // only one effect chain can be present on DirectOutputThread, so if 2756 // there is one, the track is connected to it 2757 if (!effectChains.isEmpty()) { 2758 // Do not ramp volume if volume is controlled by effect 2759 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2760 rampVolume = false; 2761 } 2762 } 2763 2764 // Convert volumes from 8.24 to 4.12 format 2765 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2766 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2767 leftVol = (uint16_t)v_clamped; 2768 v_clamped = (vr + (1 << 11)) >> 12; 2769 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2770 rightVol = (uint16_t)v_clamped; 2771 } else { 2772 leftVol = mLeftVolShort; 2773 rightVol = mRightVolShort; 2774 rampVolume = false; 2775 } 2776 2777 // reset retry count 2778 track->mRetryCount = kMaxTrackRetriesDirect; 2779 activeTrack = t; 2780 mixerStatus = MIXER_TRACKS_READY; 2781 } else { 2782 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2783 if (track->isStopped()) { 2784 track->reset(); 2785 } 2786 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2787 // We have consumed all the buffers of this track. 2788 // Remove it from the list of active tracks. 2789 trackToRemove = track; 2790 } else { 2791 // No buffers for this track. Give it a few chances to 2792 // fill a buffer, then remove it from active list. 2793 if (--(track->mRetryCount) <= 0) { 2794 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2795 trackToRemove = track; 2796 } else { 2797 mixerStatus = MIXER_TRACKS_ENABLED; 2798 } 2799 } 2800 } 2801 } 2802 2803 // remove all the tracks that need to be... 2804 if (CC_UNLIKELY(trackToRemove != 0)) { 2805 mActiveTracks.remove(trackToRemove); 2806 if (!effectChains.isEmpty()) { 2807 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2808 trackToRemove->sessionId()); 2809 effectChains[0]->decActiveTrackCnt(); 2810 } 2811 if (trackToRemove->isTerminated()) { 2812 removeTrack_l(trackToRemove); 2813 } 2814 } 2815 2816 lockEffectChains_l(effectChains); 2817 } 2818 2819 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2820 AudioBufferProvider::Buffer buffer; 2821 size_t frameCount = mFrameCount; 2822 curBuf = (int8_t *)mMixBuffer; 2823 // output audio to hardware 2824 while (frameCount) { 2825 buffer.frameCount = frameCount; 2826 activeTrack->getNextBuffer(&buffer); 2827 if (CC_UNLIKELY(buffer.raw == NULL)) { 2828 memset(curBuf, 0, frameCount * mFrameSize); 2829 break; 2830 } 2831 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2832 frameCount -= buffer.frameCount; 2833 curBuf += buffer.frameCount * mFrameSize; 2834 activeTrack->releaseBuffer(&buffer); 2835 } 2836 sleepTime = 0; 2837 standbyTime = systemTime() + standbyDelay; 2838 } else { 2839 if (sleepTime == 0) { 2840 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2841 sleepTime = activeSleepTime; 2842 } else { 2843 sleepTime = idleSleepTime; 2844 } 2845 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2846 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2847 sleepTime = 0; 2848 } 2849 } 2850 2851 if (mSuspended) { 2852 sleepTime = suspendSleepTimeUs(); 2853 } 2854 // sleepTime == 0 means we must write to audio hardware 2855 if (sleepTime == 0) { 2856 if (mixerStatus == MIXER_TRACKS_READY) { 2857 applyVolume(leftVol, rightVol, rampVolume); 2858 } 2859 for (size_t i = 0; i < effectChains.size(); i ++) { 2860 effectChains[i]->process_l(); 2861 } 2862 unlockEffectChains(effectChains); 2863 2864 mLastWriteTime = systemTime(); 2865 mInWrite = true; 2866 mBytesWritten += mixBufferSize; 2867 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2868 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2869 mNumWrites++; 2870 mInWrite = false; 2871 mStandby = false; 2872 } else { 2873 unlockEffectChains(effectChains); 2874 usleep(sleepTime); 2875 } 2876 2877 // finally let go of removed track, without the lock held 2878 // since we can't guarantee the destructors won't acquire that 2879 // same lock. 2880 trackToRemove.clear(); 2881 activeTrack.clear(); 2882 2883 // Effect chains will be actually deleted here if they were removed from 2884 // mEffectChains list during mixing or effects processing 2885 effectChains.clear(); 2886 } 2887 2888 if (!mStandby) { 2889 mOutput->stream->common.standby(&mOutput->stream->common); 2890 } 2891 2892 releaseWakeLock(); 2893 2894 ALOGV("DirectOutputThread %p exiting", this); 2895 return false; 2896} 2897 2898// getTrackName_l() must be called with ThreadBase::mLock held 2899int AudioFlinger::DirectOutputThread::getTrackName_l() 2900{ 2901 return 0; 2902} 2903 2904// deleteTrackName_l() must be called with ThreadBase::mLock held 2905void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2906{ 2907} 2908 2909// checkForNewParameters_l() must be called with ThreadBase::mLock held 2910bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2911{ 2912 bool reconfig = false; 2913 2914 while (!mNewParameters.isEmpty()) { 2915 status_t status = NO_ERROR; 2916 String8 keyValuePair = mNewParameters[0]; 2917 AudioParameter param = AudioParameter(keyValuePair); 2918 int value; 2919 2920 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2921 // do not accept frame count changes if tracks are open as the track buffer 2922 // size depends on frame count and correct behavior would not be garantied 2923 // if frame count is changed after track creation 2924 if (!mTracks.isEmpty()) { 2925 status = INVALID_OPERATION; 2926 } else { 2927 reconfig = true; 2928 } 2929 } 2930 if (status == NO_ERROR) { 2931 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2932 keyValuePair.string()); 2933 if (!mStandby && status == INVALID_OPERATION) { 2934 mOutput->stream->common.standby(&mOutput->stream->common); 2935 mStandby = true; 2936 mBytesWritten = 0; 2937 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2938 keyValuePair.string()); 2939 } 2940 if (status == NO_ERROR && reconfig) { 2941 readOutputParameters(); 2942 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2943 } 2944 } 2945 2946 mNewParameters.removeAt(0); 2947 2948 mParamStatus = status; 2949 mParamCond.signal(); 2950 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2951 // already timed out waiting for the status and will never signal the condition. 2952 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2953 } 2954 return reconfig; 2955} 2956 2957uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2958{ 2959 uint32_t time; 2960 if (audio_is_linear_pcm(mFormat)) { 2961 time = PlaybackThread::activeSleepTimeUs(); 2962 } else { 2963 time = 10000; 2964 } 2965 return time; 2966} 2967 2968uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2969{ 2970 uint32_t time; 2971 if (audio_is_linear_pcm(mFormat)) { 2972 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2973 } else { 2974 time = 10000; 2975 } 2976 return time; 2977} 2978 2979uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2980{ 2981 uint32_t time; 2982 if (audio_is_linear_pcm(mFormat)) { 2983 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2984 } else { 2985 time = 10000; 2986 } 2987 return time; 2988} 2989 2990 2991// ---------------------------------------------------------------------------- 2992 2993AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2994 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2995{ 2996 mType = ThreadBase::DUPLICATING; 2997 addOutputTrack(mainThread); 2998} 2999 3000AudioFlinger::DuplicatingThread::~DuplicatingThread() 3001{ 3002 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3003 mOutputTracks[i]->destroy(); 3004 } 3005 mOutputTracks.clear(); 3006} 3007 3008bool AudioFlinger::DuplicatingThread::threadLoop() 3009{ 3010 Vector< sp<Track> > tracksToRemove; 3011 uint32_t mixerStatus = MIXER_IDLE; 3012 nsecs_t standbyTime = systemTime(); 3013 size_t mixBufferSize = mFrameCount*mFrameSize; 3014 SortedVector< sp<OutputTrack> > outputTracks; 3015 uint32_t writeFrames = 0; 3016 uint32_t activeSleepTime = activeSleepTimeUs(); 3017 uint32_t idleSleepTime = idleSleepTimeUs(); 3018 uint32_t sleepTime = idleSleepTime; 3019 Vector< sp<EffectChain> > effectChains; 3020 3021 acquireWakeLock(); 3022 3023 while (!exitPending()) 3024 { 3025 processConfigEvents(); 3026 3027 mixerStatus = MIXER_IDLE; 3028 { // scope for the mLock 3029 3030 Mutex::Autolock _l(mLock); 3031 3032 if (checkForNewParameters_l()) { 3033 mixBufferSize = mFrameCount*mFrameSize; 3034 updateWaitTime(); 3035 activeSleepTime = activeSleepTimeUs(); 3036 idleSleepTime = idleSleepTimeUs(); 3037 } 3038 3039 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3040 3041 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3042 outputTracks.add(mOutputTracks[i]); 3043 } 3044 3045 // put audio hardware into standby after short delay 3046 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3047 mSuspended)) { 3048 if (!mStandby) { 3049 for (size_t i = 0; i < outputTracks.size(); i++) { 3050 outputTracks[i]->stop(); 3051 } 3052 mStandby = true; 3053 mBytesWritten = 0; 3054 } 3055 3056 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3057 // we're about to wait, flush the binder command buffer 3058 IPCThreadState::self()->flushCommands(); 3059 outputTracks.clear(); 3060 3061 if (exitPending()) break; 3062 3063 releaseWakeLock_l(); 3064 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3065 mWaitWorkCV.wait(mLock); 3066 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3067 acquireWakeLock_l(); 3068 3069 mPrevMixerStatus = MIXER_IDLE; 3070 if (!mMasterMute) { 3071 char value[PROPERTY_VALUE_MAX]; 3072 property_get("ro.audio.silent", value, "0"); 3073 if (atoi(value)) { 3074 ALOGD("Silence is golden"); 3075 setMasterMute(true); 3076 } 3077 } 3078 3079 standbyTime = systemTime() + kStandbyTimeInNsecs; 3080 sleepTime = idleSleepTime; 3081 continue; 3082 } 3083 } 3084 3085 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3086 3087 // prevent any changes in effect chain list and in each effect chain 3088 // during mixing and effect process as the audio buffers could be deleted 3089 // or modified if an effect is created or deleted 3090 lockEffectChains_l(effectChains); 3091 } 3092 3093 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3094 // mix buffers... 3095 if (outputsReady(outputTracks)) { 3096 mAudioMixer->process(); 3097 } else { 3098 memset(mMixBuffer, 0, mixBufferSize); 3099 } 3100 sleepTime = 0; 3101 writeFrames = mFrameCount; 3102 } else { 3103 if (sleepTime == 0) { 3104 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3105 sleepTime = activeSleepTime; 3106 } else { 3107 sleepTime = idleSleepTime; 3108 } 3109 } else if (mBytesWritten != 0) { 3110 // flush remaining overflow buffers in output tracks 3111 for (size_t i = 0; i < outputTracks.size(); i++) { 3112 if (outputTracks[i]->isActive()) { 3113 sleepTime = 0; 3114 writeFrames = 0; 3115 memset(mMixBuffer, 0, mixBufferSize); 3116 break; 3117 } 3118 } 3119 } 3120 } 3121 3122 if (mSuspended) { 3123 sleepTime = suspendSleepTimeUs(); 3124 } 3125 // sleepTime == 0 means we must write to audio hardware 3126 if (sleepTime == 0) { 3127 for (size_t i = 0; i < effectChains.size(); i ++) { 3128 effectChains[i]->process_l(); 3129 } 3130 // enable changes in effect chain 3131 unlockEffectChains(effectChains); 3132 3133 standbyTime = systemTime() + kStandbyTimeInNsecs; 3134 for (size_t i = 0; i < outputTracks.size(); i++) { 3135 outputTracks[i]->write(mMixBuffer, writeFrames); 3136 } 3137 mStandby = false; 3138 mBytesWritten += mixBufferSize; 3139 } else { 3140 // enable changes in effect chain 3141 unlockEffectChains(effectChains); 3142 usleep(sleepTime); 3143 } 3144 3145 // finally let go of all our tracks, without the lock held 3146 // since we can't guarantee the destructors won't acquire that 3147 // same lock. 3148 tracksToRemove.clear(); 3149 outputTracks.clear(); 3150 3151 // Effect chains will be actually deleted here if they were removed from 3152 // mEffectChains list during mixing or effects processing 3153 effectChains.clear(); 3154 } 3155 3156 releaseWakeLock(); 3157 3158 return false; 3159} 3160 3161void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3162{ 3163 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3164 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3165 this, 3166 mSampleRate, 3167 mFormat, 3168 mChannelMask, 3169 frameCount); 3170 if (outputTrack->cblk() != NULL) { 3171 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3172 mOutputTracks.add(outputTrack); 3173 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3174 updateWaitTime(); 3175 } 3176} 3177 3178void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3179{ 3180 Mutex::Autolock _l(mLock); 3181 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3182 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3183 mOutputTracks[i]->destroy(); 3184 mOutputTracks.removeAt(i); 3185 updateWaitTime(); 3186 return; 3187 } 3188 } 3189 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3190} 3191 3192void AudioFlinger::DuplicatingThread::updateWaitTime() 3193{ 3194 mWaitTimeMs = UINT_MAX; 3195 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3196 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3197 if (strong != NULL) { 3198 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3199 if (waitTimeMs < mWaitTimeMs) { 3200 mWaitTimeMs = waitTimeMs; 3201 } 3202 } 3203 } 3204} 3205 3206 3207bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3208{ 3209 for (size_t i = 0; i < outputTracks.size(); i++) { 3210 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3211 if (thread == 0) { 3212 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3213 return false; 3214 } 3215 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3216 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3217 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3218 return false; 3219 } 3220 } 3221 return true; 3222} 3223 3224uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3225{ 3226 return (mWaitTimeMs * 1000) / 2; 3227} 3228 3229// ---------------------------------------------------------------------------- 3230 3231// TrackBase constructor must be called with AudioFlinger::mLock held 3232AudioFlinger::ThreadBase::TrackBase::TrackBase( 3233 const wp<ThreadBase>& thread, 3234 const sp<Client>& client, 3235 uint32_t sampleRate, 3236 audio_format_t format, 3237 uint32_t channelMask, 3238 int frameCount, 3239 uint32_t flags, 3240 const sp<IMemory>& sharedBuffer, 3241 int sessionId) 3242 : RefBase(), 3243 mThread(thread), 3244 mClient(client), 3245 mCblk(0), 3246 mFrameCount(0), 3247 mState(IDLE), 3248 mClientTid(-1), 3249 mFormat(format), 3250 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3251 mSessionId(sessionId) 3252{ 3253 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3254 3255 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3256 size_t size = sizeof(audio_track_cblk_t); 3257 uint8_t channelCount = popcount(channelMask); 3258 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3259 if (sharedBuffer == 0) { 3260 size += bufferSize; 3261 } 3262 3263 if (client != NULL) { 3264 mCblkMemory = client->heap()->allocate(size); 3265 if (mCblkMemory != 0) { 3266 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3267 if (mCblk) { // construct the shared structure in-place. 3268 new(mCblk) audio_track_cblk_t(); 3269 // clear all buffers 3270 mCblk->frameCount = frameCount; 3271 mCblk->sampleRate = sampleRate; 3272 mChannelCount = channelCount; 3273 mChannelMask = channelMask; 3274 if (sharedBuffer == 0) { 3275 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3276 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3277 // Force underrun condition to avoid false underrun callback until first data is 3278 // written to buffer (other flags are cleared) 3279 mCblk->flags = CBLK_UNDERRUN_ON; 3280 } else { 3281 mBuffer = sharedBuffer->pointer(); 3282 } 3283 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3284 } 3285 } else { 3286 ALOGE("not enough memory for AudioTrack size=%u", size); 3287 client->heap()->dump("AudioTrack"); 3288 return; 3289 } 3290 } else { 3291 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3292 // construct the shared structure in-place. 3293 new(mCblk) audio_track_cblk_t(); 3294 // clear all buffers 3295 mCblk->frameCount = frameCount; 3296 mCblk->sampleRate = sampleRate; 3297 mChannelCount = channelCount; 3298 mChannelMask = channelMask; 3299 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3300 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3301 // Force underrun condition to avoid false underrun callback until first data is 3302 // written to buffer (other flags are cleared) 3303 mCblk->flags = CBLK_UNDERRUN_ON; 3304 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3305 } 3306} 3307 3308AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3309{ 3310 if (mCblk) { 3311 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3312 if (mClient == NULL) { 3313 delete mCblk; 3314 } 3315 } 3316 mCblkMemory.clear(); // and free the shared memory 3317 if (mClient != NULL) { 3318 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3319 mClient.clear(); 3320 } 3321} 3322 3323void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3324{ 3325 buffer->raw = NULL; 3326 mFrameCount = buffer->frameCount; 3327 step(); 3328 buffer->frameCount = 0; 3329} 3330 3331bool AudioFlinger::ThreadBase::TrackBase::step() { 3332 bool result; 3333 audio_track_cblk_t* cblk = this->cblk(); 3334 3335 result = cblk->stepServer(mFrameCount); 3336 if (!result) { 3337 ALOGV("stepServer failed acquiring cblk mutex"); 3338 mFlags |= STEPSERVER_FAILED; 3339 } 3340 return result; 3341} 3342 3343void AudioFlinger::ThreadBase::TrackBase::reset() { 3344 audio_track_cblk_t* cblk = this->cblk(); 3345 3346 cblk->user = 0; 3347 cblk->server = 0; 3348 cblk->userBase = 0; 3349 cblk->serverBase = 0; 3350 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3351 ALOGV("TrackBase::reset"); 3352} 3353 3354sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3355{ 3356 return mCblkMemory; 3357} 3358 3359int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3360 return (int)mCblk->sampleRate; 3361} 3362 3363int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3364 return (const int)mChannelCount; 3365} 3366 3367uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3368 return mChannelMask; 3369} 3370 3371void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3372 audio_track_cblk_t* cblk = this->cblk(); 3373 size_t frameSize = cblk->frameSize; 3374 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3375 int8_t *bufferEnd = bufferStart + frames * frameSize; 3376 3377 // Check validity of returned pointer in case the track control block would have been corrupted. 3378 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3379 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3380 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3381 server %d, serverBase %d, user %d, userBase %d", 3382 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3383 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3384 return 0; 3385 } 3386 3387 return bufferStart; 3388} 3389 3390// ---------------------------------------------------------------------------- 3391 3392// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3393AudioFlinger::PlaybackThread::Track::Track( 3394 const wp<ThreadBase>& thread, 3395 const sp<Client>& client, 3396 audio_stream_type_t streamType, 3397 uint32_t sampleRate, 3398 audio_format_t format, 3399 uint32_t channelMask, 3400 int frameCount, 3401 const sp<IMemory>& sharedBuffer, 3402 int sessionId) 3403 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3404 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3405 mAuxEffectId(0), mHasVolumeController(false) 3406{ 3407 if (mCblk != NULL) { 3408 sp<ThreadBase> baseThread = thread.promote(); 3409 if (baseThread != 0) { 3410 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3411 mName = playbackThread->getTrackName_l(); 3412 mMainBuffer = playbackThread->mixBuffer(); 3413 } 3414 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3415 if (mName < 0) { 3416 ALOGE("no more track names available"); 3417 } 3418 mStreamType = streamType; 3419 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3420 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3421 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3422 } 3423} 3424 3425AudioFlinger::PlaybackThread::Track::~Track() 3426{ 3427 ALOGV("PlaybackThread::Track destructor"); 3428 sp<ThreadBase> thread = mThread.promote(); 3429 if (thread != 0) { 3430 Mutex::Autolock _l(thread->mLock); 3431 mState = TERMINATED; 3432 } 3433} 3434 3435void AudioFlinger::PlaybackThread::Track::destroy() 3436{ 3437 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3438 // by removing it from mTracks vector, so there is a risk that this Tracks's 3439 // desctructor is called. As the destructor needs to lock mLock, 3440 // we must acquire a strong reference on this Track before locking mLock 3441 // here so that the destructor is called only when exiting this function. 3442 // On the other hand, as long as Track::destroy() is only called by 3443 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3444 // this Track with its member mTrack. 3445 sp<Track> keep(this); 3446 { // scope for mLock 3447 sp<ThreadBase> thread = mThread.promote(); 3448 if (thread != 0) { 3449 if (!isOutputTrack()) { 3450 if (mState == ACTIVE || mState == RESUMING) { 3451 AudioSystem::stopOutput(thread->id(), 3452 (audio_stream_type_t)mStreamType, 3453 mSessionId); 3454 3455 // to track the speaker usage 3456 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3457 } 3458 AudioSystem::releaseOutput(thread->id()); 3459 } 3460 Mutex::Autolock _l(thread->mLock); 3461 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3462 playbackThread->destroyTrack_l(this); 3463 } 3464 } 3465} 3466 3467void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3468{ 3469 uint32_t vlr = mCblk->getVolumeLR(); 3470 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", 3471 mName - AudioMixer::TRACK0, 3472 (mClient == NULL) ? getpid() : mClient->pid(), 3473 mStreamType, 3474 mFormat, 3475 mChannelMask, 3476 mSessionId, 3477 mFrameCount, 3478 mState, 3479 mMute, 3480 mFillingUpStatus, 3481 mCblk->sampleRate, 3482 vlr & 0xFFFF, 3483 vlr >> 16, 3484 mCblk->server, 3485 mCblk->user, 3486 (int)mMainBuffer, 3487 (int)mAuxBuffer); 3488} 3489 3490status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3491{ 3492 audio_track_cblk_t* cblk = this->cblk(); 3493 uint32_t framesReady; 3494 uint32_t framesReq = buffer->frameCount; 3495 3496 // Check if last stepServer failed, try to step now 3497 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3498 if (!step()) goto getNextBuffer_exit; 3499 ALOGV("stepServer recovered"); 3500 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3501 } 3502 3503 framesReady = cblk->framesReady(); 3504 3505 if (CC_LIKELY(framesReady)) { 3506 uint32_t s = cblk->server; 3507 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3508 3509 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3510 if (framesReq > framesReady) { 3511 framesReq = framesReady; 3512 } 3513 if (s + framesReq > bufferEnd) { 3514 framesReq = bufferEnd - s; 3515 } 3516 3517 buffer->raw = getBuffer(s, framesReq); 3518 if (buffer->raw == NULL) goto getNextBuffer_exit; 3519 3520 buffer->frameCount = framesReq; 3521 return NO_ERROR; 3522 } 3523 3524getNextBuffer_exit: 3525 buffer->raw = NULL; 3526 buffer->frameCount = 0; 3527 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3528 return NOT_ENOUGH_DATA; 3529} 3530 3531bool AudioFlinger::PlaybackThread::Track::isReady() const { 3532 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3533 3534 if (mCblk->framesReady() >= mCblk->frameCount || 3535 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3536 mFillingUpStatus = FS_FILLED; 3537 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3538 return true; 3539 } 3540 return false; 3541} 3542 3543status_t AudioFlinger::PlaybackThread::Track::start() 3544{ 3545 status_t status = NO_ERROR; 3546 ALOGV("start(%d), calling thread %d session %d", 3547 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3548 sp<ThreadBase> thread = mThread.promote(); 3549 if (thread != 0) { 3550 Mutex::Autolock _l(thread->mLock); 3551 int state = mState; 3552 // here the track could be either new, or restarted 3553 // in both cases "unstop" the track 3554 if (mState == PAUSED) { 3555 mState = TrackBase::RESUMING; 3556 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3557 } else { 3558 mState = TrackBase::ACTIVE; 3559 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3560 } 3561 3562 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3563 thread->mLock.unlock(); 3564 status = AudioSystem::startOutput(thread->id(), 3565 (audio_stream_type_t)mStreamType, 3566 mSessionId); 3567 thread->mLock.lock(); 3568 3569 // to track the speaker usage 3570 if (status == NO_ERROR) { 3571 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3572 } 3573 } 3574 if (status == NO_ERROR) { 3575 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3576 playbackThread->addTrack_l(this); 3577 } else { 3578 mState = state; 3579 } 3580 } else { 3581 status = BAD_VALUE; 3582 } 3583 return status; 3584} 3585 3586void AudioFlinger::PlaybackThread::Track::stop() 3587{ 3588 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3589 sp<ThreadBase> thread = mThread.promote(); 3590 if (thread != 0) { 3591 Mutex::Autolock _l(thread->mLock); 3592 int state = mState; 3593 if (mState > STOPPED) { 3594 mState = STOPPED; 3595 // If the track is not active (PAUSED and buffers full), flush buffers 3596 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3597 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3598 reset(); 3599 } 3600 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3601 } 3602 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3603 thread->mLock.unlock(); 3604 AudioSystem::stopOutput(thread->id(), 3605 (audio_stream_type_t)mStreamType, 3606 mSessionId); 3607 thread->mLock.lock(); 3608 3609 // to track the speaker usage 3610 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3611 } 3612 } 3613} 3614 3615void AudioFlinger::PlaybackThread::Track::pause() 3616{ 3617 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3618 sp<ThreadBase> thread = mThread.promote(); 3619 if (thread != 0) { 3620 Mutex::Autolock _l(thread->mLock); 3621 if (mState == ACTIVE || mState == RESUMING) { 3622 mState = PAUSING; 3623 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3624 if (!isOutputTrack()) { 3625 thread->mLock.unlock(); 3626 AudioSystem::stopOutput(thread->id(), 3627 (audio_stream_type_t)mStreamType, 3628 mSessionId); 3629 thread->mLock.lock(); 3630 3631 // to track the speaker usage 3632 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3633 } 3634 } 3635 } 3636} 3637 3638void AudioFlinger::PlaybackThread::Track::flush() 3639{ 3640 ALOGV("flush(%d)", mName); 3641 sp<ThreadBase> thread = mThread.promote(); 3642 if (thread != 0) { 3643 Mutex::Autolock _l(thread->mLock); 3644 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3645 return; 3646 } 3647 // No point remaining in PAUSED state after a flush => go to 3648 // STOPPED state 3649 mState = STOPPED; 3650 3651 // do not reset the track if it is still in the process of being stopped or paused. 3652 // this will be done by prepareTracks_l() when the track is stopped. 3653 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3654 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3655 reset(); 3656 } 3657 } 3658} 3659 3660void AudioFlinger::PlaybackThread::Track::reset() 3661{ 3662 // Do not reset twice to avoid discarding data written just after a flush and before 3663 // the audioflinger thread detects the track is stopped. 3664 if (!mResetDone) { 3665 TrackBase::reset(); 3666 // Force underrun condition to avoid false underrun callback until first data is 3667 // written to buffer 3668 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3669 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3670 mFillingUpStatus = FS_FILLING; 3671 mResetDone = true; 3672 } 3673} 3674 3675void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3676{ 3677 mMute = muted; 3678} 3679 3680status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3681{ 3682 status_t status = DEAD_OBJECT; 3683 sp<ThreadBase> thread = mThread.promote(); 3684 if (thread != 0) { 3685 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3686 status = playbackThread->attachAuxEffect(this, EffectId); 3687 } 3688 return status; 3689} 3690 3691void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3692{ 3693 mAuxEffectId = EffectId; 3694 mAuxBuffer = buffer; 3695} 3696 3697// ---------------------------------------------------------------------------- 3698 3699// RecordTrack constructor must be called with AudioFlinger::mLock held 3700AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3701 const wp<ThreadBase>& thread, 3702 const sp<Client>& client, 3703 uint32_t sampleRate, 3704 audio_format_t format, 3705 uint32_t channelMask, 3706 int frameCount, 3707 uint32_t flags, 3708 int sessionId) 3709 : TrackBase(thread, client, sampleRate, format, 3710 channelMask, frameCount, flags, 0, sessionId), 3711 mOverflow(false) 3712{ 3713 if (mCblk != NULL) { 3714 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3715 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3716 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3717 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3718 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3719 } else { 3720 mCblk->frameSize = sizeof(int8_t); 3721 } 3722 } 3723} 3724 3725AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3726{ 3727 sp<ThreadBase> thread = mThread.promote(); 3728 if (thread != 0) { 3729 AudioSystem::releaseInput(thread->id()); 3730 } 3731} 3732 3733status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3734{ 3735 audio_track_cblk_t* cblk = this->cblk(); 3736 uint32_t framesAvail; 3737 uint32_t framesReq = buffer->frameCount; 3738 3739 // Check if last stepServer failed, try to step now 3740 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3741 if (!step()) goto getNextBuffer_exit; 3742 ALOGV("stepServer recovered"); 3743 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3744 } 3745 3746 framesAvail = cblk->framesAvailable_l(); 3747 3748 if (CC_LIKELY(framesAvail)) { 3749 uint32_t s = cblk->server; 3750 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3751 3752 if (framesReq > framesAvail) { 3753 framesReq = framesAvail; 3754 } 3755 if (s + framesReq > bufferEnd) { 3756 framesReq = bufferEnd - s; 3757 } 3758 3759 buffer->raw = getBuffer(s, framesReq); 3760 if (buffer->raw == NULL) goto getNextBuffer_exit; 3761 3762 buffer->frameCount = framesReq; 3763 return NO_ERROR; 3764 } 3765 3766getNextBuffer_exit: 3767 buffer->raw = NULL; 3768 buffer->frameCount = 0; 3769 return NOT_ENOUGH_DATA; 3770} 3771 3772status_t AudioFlinger::RecordThread::RecordTrack::start() 3773{ 3774 sp<ThreadBase> thread = mThread.promote(); 3775 if (thread != 0) { 3776 RecordThread *recordThread = (RecordThread *)thread.get(); 3777 return recordThread->start(this); 3778 } else { 3779 return BAD_VALUE; 3780 } 3781} 3782 3783void AudioFlinger::RecordThread::RecordTrack::stop() 3784{ 3785 sp<ThreadBase> thread = mThread.promote(); 3786 if (thread != 0) { 3787 RecordThread *recordThread = (RecordThread *)thread.get(); 3788 recordThread->stop(this); 3789 TrackBase::reset(); 3790 // Force overerrun condition to avoid false overrun callback until first data is 3791 // read from buffer 3792 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3793 } 3794} 3795 3796void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3797{ 3798 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3799 (mClient == NULL) ? getpid() : mClient->pid(), 3800 mFormat, 3801 mChannelMask, 3802 mSessionId, 3803 mFrameCount, 3804 mState, 3805 mCblk->sampleRate, 3806 mCblk->server, 3807 mCblk->user); 3808} 3809 3810 3811// ---------------------------------------------------------------------------- 3812 3813AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3814 const wp<ThreadBase>& thread, 3815 DuplicatingThread *sourceThread, 3816 uint32_t sampleRate, 3817 audio_format_t format, 3818 uint32_t channelMask, 3819 int frameCount) 3820 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3821 mActive(false), mSourceThread(sourceThread) 3822{ 3823 3824 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3825 if (mCblk != NULL) { 3826 mCblk->flags |= CBLK_DIRECTION_OUT; 3827 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3828 mOutBuffer.frameCount = 0; 3829 playbackThread->mTracks.add(this); 3830 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3831 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3832 mCblk, mBuffer, mCblk->buffers, 3833 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3834 } else { 3835 ALOGW("Error creating output track on thread %p", playbackThread); 3836 } 3837} 3838 3839AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3840{ 3841 clearBufferQueue(); 3842} 3843 3844status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3845{ 3846 status_t status = Track::start(); 3847 if (status != NO_ERROR) { 3848 return status; 3849 } 3850 3851 mActive = true; 3852 mRetryCount = 127; 3853 return status; 3854} 3855 3856void AudioFlinger::PlaybackThread::OutputTrack::stop() 3857{ 3858 Track::stop(); 3859 clearBufferQueue(); 3860 mOutBuffer.frameCount = 0; 3861 mActive = false; 3862} 3863 3864bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3865{ 3866 Buffer *pInBuffer; 3867 Buffer inBuffer; 3868 uint32_t channelCount = mChannelCount; 3869 bool outputBufferFull = false; 3870 inBuffer.frameCount = frames; 3871 inBuffer.i16 = data; 3872 3873 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3874 3875 if (!mActive && frames != 0) { 3876 start(); 3877 sp<ThreadBase> thread = mThread.promote(); 3878 if (thread != 0) { 3879 MixerThread *mixerThread = (MixerThread *)thread.get(); 3880 if (mCblk->frameCount > frames){ 3881 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3882 uint32_t startFrames = (mCblk->frameCount - frames); 3883 pInBuffer = new Buffer; 3884 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3885 pInBuffer->frameCount = startFrames; 3886 pInBuffer->i16 = pInBuffer->mBuffer; 3887 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3888 mBufferQueue.add(pInBuffer); 3889 } else { 3890 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3891 } 3892 } 3893 } 3894 } 3895 3896 while (waitTimeLeftMs) { 3897 // First write pending buffers, then new data 3898 if (mBufferQueue.size()) { 3899 pInBuffer = mBufferQueue.itemAt(0); 3900 } else { 3901 pInBuffer = &inBuffer; 3902 } 3903 3904 if (pInBuffer->frameCount == 0) { 3905 break; 3906 } 3907 3908 if (mOutBuffer.frameCount == 0) { 3909 mOutBuffer.frameCount = pInBuffer->frameCount; 3910 nsecs_t startTime = systemTime(); 3911 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3912 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3913 outputBufferFull = true; 3914 break; 3915 } 3916 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3917 if (waitTimeLeftMs >= waitTimeMs) { 3918 waitTimeLeftMs -= waitTimeMs; 3919 } else { 3920 waitTimeLeftMs = 0; 3921 } 3922 } 3923 3924 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3925 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3926 mCblk->stepUser(outFrames); 3927 pInBuffer->frameCount -= outFrames; 3928 pInBuffer->i16 += outFrames * channelCount; 3929 mOutBuffer.frameCount -= outFrames; 3930 mOutBuffer.i16 += outFrames * channelCount; 3931 3932 if (pInBuffer->frameCount == 0) { 3933 if (mBufferQueue.size()) { 3934 mBufferQueue.removeAt(0); 3935 delete [] pInBuffer->mBuffer; 3936 delete pInBuffer; 3937 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3938 } else { 3939 break; 3940 } 3941 } 3942 } 3943 3944 // If we could not write all frames, allocate a buffer and queue it for next time. 3945 if (inBuffer.frameCount) { 3946 sp<ThreadBase> thread = mThread.promote(); 3947 if (thread != 0 && !thread->standby()) { 3948 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3949 pInBuffer = new Buffer; 3950 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3951 pInBuffer->frameCount = inBuffer.frameCount; 3952 pInBuffer->i16 = pInBuffer->mBuffer; 3953 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3954 mBufferQueue.add(pInBuffer); 3955 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3956 } else { 3957 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3958 } 3959 } 3960 } 3961 3962 // Calling write() with a 0 length buffer, means that no more data will be written: 3963 // If no more buffers are pending, fill output track buffer to make sure it is started 3964 // by output mixer. 3965 if (frames == 0 && mBufferQueue.size() == 0) { 3966 if (mCblk->user < mCblk->frameCount) { 3967 frames = mCblk->frameCount - mCblk->user; 3968 pInBuffer = new Buffer; 3969 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3970 pInBuffer->frameCount = frames; 3971 pInBuffer->i16 = pInBuffer->mBuffer; 3972 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3973 mBufferQueue.add(pInBuffer); 3974 } else if (mActive) { 3975 stop(); 3976 } 3977 } 3978 3979 return outputBufferFull; 3980} 3981 3982status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3983{ 3984 int active; 3985 status_t result; 3986 audio_track_cblk_t* cblk = mCblk; 3987 uint32_t framesReq = buffer->frameCount; 3988 3989// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3990 buffer->frameCount = 0; 3991 3992 uint32_t framesAvail = cblk->framesAvailable(); 3993 3994 3995 if (framesAvail == 0) { 3996 Mutex::Autolock _l(cblk->lock); 3997 goto start_loop_here; 3998 while (framesAvail == 0) { 3999 active = mActive; 4000 if (CC_UNLIKELY(!active)) { 4001 ALOGV("Not active and NO_MORE_BUFFERS"); 4002 return AudioTrack::NO_MORE_BUFFERS; 4003 } 4004 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4005 if (result != NO_ERROR) { 4006 return AudioTrack::NO_MORE_BUFFERS; 4007 } 4008 // read the server count again 4009 start_loop_here: 4010 framesAvail = cblk->framesAvailable_l(); 4011 } 4012 } 4013 4014// if (framesAvail < framesReq) { 4015// return AudioTrack::NO_MORE_BUFFERS; 4016// } 4017 4018 if (framesReq > framesAvail) { 4019 framesReq = framesAvail; 4020 } 4021 4022 uint32_t u = cblk->user; 4023 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4024 4025 if (u + framesReq > bufferEnd) { 4026 framesReq = bufferEnd - u; 4027 } 4028 4029 buffer->frameCount = framesReq; 4030 buffer->raw = (void *)cblk->buffer(u); 4031 return NO_ERROR; 4032} 4033 4034 4035void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4036{ 4037 size_t size = mBufferQueue.size(); 4038 Buffer *pBuffer; 4039 4040 for (size_t i = 0; i < size; i++) { 4041 pBuffer = mBufferQueue.itemAt(i); 4042 delete [] pBuffer->mBuffer; 4043 delete pBuffer; 4044 } 4045 mBufferQueue.clear(); 4046} 4047 4048// ---------------------------------------------------------------------------- 4049 4050AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4051 : RefBase(), 4052 mAudioFlinger(audioFlinger), 4053 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4054 mPid(pid) 4055{ 4056 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4057} 4058 4059// Client destructor must be called with AudioFlinger::mLock held 4060AudioFlinger::Client::~Client() 4061{ 4062 mAudioFlinger->removeClient_l(mPid); 4063} 4064 4065const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4066{ 4067 return mMemoryDealer; 4068} 4069 4070// ---------------------------------------------------------------------------- 4071 4072AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4073 const sp<IAudioFlingerClient>& client, 4074 pid_t pid) 4075 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4076{ 4077} 4078 4079AudioFlinger::NotificationClient::~NotificationClient() 4080{ 4081 mClient.clear(); 4082} 4083 4084void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4085{ 4086 sp<NotificationClient> keep(this); 4087 { 4088 mAudioFlinger->removeNotificationClient(mPid); 4089 } 4090} 4091 4092// ---------------------------------------------------------------------------- 4093 4094AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4095 : BnAudioTrack(), 4096 mTrack(track) 4097{ 4098} 4099 4100AudioFlinger::TrackHandle::~TrackHandle() { 4101 // just stop the track on deletion, associated resources 4102 // will be freed from the main thread once all pending buffers have 4103 // been played. Unless it's not in the active track list, in which 4104 // case we free everything now... 4105 mTrack->destroy(); 4106} 4107 4108status_t AudioFlinger::TrackHandle::start() { 4109 return mTrack->start(); 4110} 4111 4112void AudioFlinger::TrackHandle::stop() { 4113 mTrack->stop(); 4114} 4115 4116void AudioFlinger::TrackHandle::flush() { 4117 mTrack->flush(); 4118} 4119 4120void AudioFlinger::TrackHandle::mute(bool e) { 4121 mTrack->mute(e); 4122} 4123 4124void AudioFlinger::TrackHandle::pause() { 4125 mTrack->pause(); 4126} 4127 4128sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4129 return mTrack->getCblk(); 4130} 4131 4132status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4133{ 4134 return mTrack->attachAuxEffect(EffectId); 4135} 4136 4137status_t AudioFlinger::TrackHandle::onTransact( 4138 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4139{ 4140 return BnAudioTrack::onTransact(code, data, reply, flags); 4141} 4142 4143// ---------------------------------------------------------------------------- 4144 4145sp<IAudioRecord> AudioFlinger::openRecord( 4146 pid_t pid, 4147 int input, 4148 uint32_t sampleRate, 4149 audio_format_t format, 4150 uint32_t channelMask, 4151 int frameCount, 4152 uint32_t flags, 4153 int *sessionId, 4154 status_t *status) 4155{ 4156 sp<RecordThread::RecordTrack> recordTrack; 4157 sp<RecordHandle> recordHandle; 4158 sp<Client> client; 4159 wp<Client> wclient; 4160 status_t lStatus; 4161 RecordThread *thread; 4162 size_t inFrameCount; 4163 int lSessionId; 4164 4165 // check calling permissions 4166 if (!recordingAllowed()) { 4167 lStatus = PERMISSION_DENIED; 4168 goto Exit; 4169 } 4170 4171 // add client to list 4172 { // scope for mLock 4173 Mutex::Autolock _l(mLock); 4174 thread = checkRecordThread_l(input); 4175 if (thread == NULL) { 4176 lStatus = BAD_VALUE; 4177 goto Exit; 4178 } 4179 4180 wclient = mClients.valueFor(pid); 4181 if (wclient != NULL) { 4182 client = wclient.promote(); 4183 } else { 4184 client = new Client(this, pid); 4185 mClients.add(pid, client); 4186 } 4187 4188 // If no audio session id is provided, create one here 4189 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4190 lSessionId = *sessionId; 4191 } else { 4192 lSessionId = nextUniqueId(); 4193 if (sessionId != NULL) { 4194 *sessionId = lSessionId; 4195 } 4196 } 4197 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4198 recordTrack = thread->createRecordTrack_l(client, 4199 sampleRate, 4200 format, 4201 channelMask, 4202 frameCount, 4203 flags, 4204 lSessionId, 4205 &lStatus); 4206 } 4207 if (lStatus != NO_ERROR) { 4208 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4209 // destructor is called by the TrackBase destructor with mLock held 4210 client.clear(); 4211 recordTrack.clear(); 4212 goto Exit; 4213 } 4214 4215 // return to handle to client 4216 recordHandle = new RecordHandle(recordTrack); 4217 lStatus = NO_ERROR; 4218 4219Exit: 4220 if (status) { 4221 *status = lStatus; 4222 } 4223 return recordHandle; 4224} 4225 4226// ---------------------------------------------------------------------------- 4227 4228AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4229 : BnAudioRecord(), 4230 mRecordTrack(recordTrack) 4231{ 4232} 4233 4234AudioFlinger::RecordHandle::~RecordHandle() { 4235 stop(); 4236} 4237 4238status_t AudioFlinger::RecordHandle::start() { 4239 ALOGV("RecordHandle::start()"); 4240 return mRecordTrack->start(); 4241} 4242 4243void AudioFlinger::RecordHandle::stop() { 4244 ALOGV("RecordHandle::stop()"); 4245 mRecordTrack->stop(); 4246} 4247 4248sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4249 return mRecordTrack->getCblk(); 4250} 4251 4252status_t AudioFlinger::RecordHandle::onTransact( 4253 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4254{ 4255 return BnAudioRecord::onTransact(code, data, reply, flags); 4256} 4257 4258// ---------------------------------------------------------------------------- 4259 4260AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4261 AudioStreamIn *input, 4262 uint32_t sampleRate, 4263 uint32_t channels, 4264 int id, 4265 uint32_t device) : 4266 ThreadBase(audioFlinger, id, device), 4267 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4268{ 4269 mType = ThreadBase::RECORD; 4270 4271 snprintf(mName, kNameLength, "AudioIn_%d", id); 4272 4273 mReqChannelCount = popcount(channels); 4274 mReqSampleRate = sampleRate; 4275 readInputParameters(); 4276} 4277 4278 4279AudioFlinger::RecordThread::~RecordThread() 4280{ 4281 delete[] mRsmpInBuffer; 4282 if (mResampler != NULL) { 4283 delete mResampler; 4284 delete[] mRsmpOutBuffer; 4285 } 4286} 4287 4288void AudioFlinger::RecordThread::onFirstRef() 4289{ 4290 run(mName, PRIORITY_URGENT_AUDIO); 4291} 4292 4293status_t AudioFlinger::RecordThread::readyToRun() 4294{ 4295 status_t status = initCheck(); 4296 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4297 return status; 4298} 4299 4300bool AudioFlinger::RecordThread::threadLoop() 4301{ 4302 AudioBufferProvider::Buffer buffer; 4303 sp<RecordTrack> activeTrack; 4304 Vector< sp<EffectChain> > effectChains; 4305 4306 nsecs_t lastWarning = 0; 4307 4308 acquireWakeLock(); 4309 4310 // start recording 4311 while (!exitPending()) { 4312 4313 processConfigEvents(); 4314 4315 { // scope for mLock 4316 Mutex::Autolock _l(mLock); 4317 checkForNewParameters_l(); 4318 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4319 if (!mStandby) { 4320 mInput->stream->common.standby(&mInput->stream->common); 4321 mStandby = true; 4322 } 4323 4324 if (exitPending()) break; 4325 4326 releaseWakeLock_l(); 4327 ALOGV("RecordThread: loop stopping"); 4328 // go to sleep 4329 mWaitWorkCV.wait(mLock); 4330 ALOGV("RecordThread: loop starting"); 4331 acquireWakeLock_l(); 4332 continue; 4333 } 4334 if (mActiveTrack != 0) { 4335 if (mActiveTrack->mState == TrackBase::PAUSING) { 4336 if (!mStandby) { 4337 mInput->stream->common.standby(&mInput->stream->common); 4338 mStandby = true; 4339 } 4340 mActiveTrack.clear(); 4341 mStartStopCond.broadcast(); 4342 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4343 if (mReqChannelCount != mActiveTrack->channelCount()) { 4344 mActiveTrack.clear(); 4345 mStartStopCond.broadcast(); 4346 } else if (mBytesRead != 0) { 4347 // record start succeeds only if first read from audio input 4348 // succeeds 4349 if (mBytesRead > 0) { 4350 mActiveTrack->mState = TrackBase::ACTIVE; 4351 } else { 4352 mActiveTrack.clear(); 4353 } 4354 mStartStopCond.broadcast(); 4355 } 4356 mStandby = false; 4357 } 4358 } 4359 lockEffectChains_l(effectChains); 4360 } 4361 4362 if (mActiveTrack != 0) { 4363 if (mActiveTrack->mState != TrackBase::ACTIVE && 4364 mActiveTrack->mState != TrackBase::RESUMING) { 4365 unlockEffectChains(effectChains); 4366 usleep(kRecordThreadSleepUs); 4367 continue; 4368 } 4369 for (size_t i = 0; i < effectChains.size(); i ++) { 4370 effectChains[i]->process_l(); 4371 } 4372 4373 buffer.frameCount = mFrameCount; 4374 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4375 size_t framesOut = buffer.frameCount; 4376 if (mResampler == NULL) { 4377 // no resampling 4378 while (framesOut) { 4379 size_t framesIn = mFrameCount - mRsmpInIndex; 4380 if (framesIn) { 4381 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4382 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4383 if (framesIn > framesOut) 4384 framesIn = framesOut; 4385 mRsmpInIndex += framesIn; 4386 framesOut -= framesIn; 4387 if ((int)mChannelCount == mReqChannelCount || 4388 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4389 memcpy(dst, src, framesIn * mFrameSize); 4390 } else { 4391 int16_t *src16 = (int16_t *)src; 4392 int16_t *dst16 = (int16_t *)dst; 4393 if (mChannelCount == 1) { 4394 while (framesIn--) { 4395 *dst16++ = *src16; 4396 *dst16++ = *src16++; 4397 } 4398 } else { 4399 while (framesIn--) { 4400 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4401 src16 += 2; 4402 } 4403 } 4404 } 4405 } 4406 if (framesOut && mFrameCount == mRsmpInIndex) { 4407 if (framesOut == mFrameCount && 4408 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4409 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4410 framesOut = 0; 4411 } else { 4412 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4413 mRsmpInIndex = 0; 4414 } 4415 if (mBytesRead < 0) { 4416 ALOGE("Error reading audio input"); 4417 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4418 // Force input into standby so that it tries to 4419 // recover at next read attempt 4420 mInput->stream->common.standby(&mInput->stream->common); 4421 usleep(kRecordThreadSleepUs); 4422 } 4423 mRsmpInIndex = mFrameCount; 4424 framesOut = 0; 4425 buffer.frameCount = 0; 4426 } 4427 } 4428 } 4429 } else { 4430 // resampling 4431 4432 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4433 // alter output frame count as if we were expecting stereo samples 4434 if (mChannelCount == 1 && mReqChannelCount == 1) { 4435 framesOut >>= 1; 4436 } 4437 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4438 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4439 // are 32 bit aligned which should be always true. 4440 if (mChannelCount == 2 && mReqChannelCount == 1) { 4441 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4442 // the resampler always outputs stereo samples: do post stereo to mono conversion 4443 int16_t *src = (int16_t *)mRsmpOutBuffer; 4444 int16_t *dst = buffer.i16; 4445 while (framesOut--) { 4446 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4447 src += 2; 4448 } 4449 } else { 4450 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4451 } 4452 4453 } 4454 mActiveTrack->releaseBuffer(&buffer); 4455 mActiveTrack->overflow(); 4456 } 4457 // client isn't retrieving buffers fast enough 4458 else { 4459 if (!mActiveTrack->setOverflow()) { 4460 nsecs_t now = systemTime(); 4461 if ((now - lastWarning) > kWarningThrottleNs) { 4462 ALOGW("RecordThread: buffer overflow"); 4463 lastWarning = now; 4464 } 4465 } 4466 // Release the processor for a while before asking for a new buffer. 4467 // This will give the application more chance to read from the buffer and 4468 // clear the overflow. 4469 usleep(kRecordThreadSleepUs); 4470 } 4471 } 4472 // enable changes in effect chain 4473 unlockEffectChains(effectChains); 4474 effectChains.clear(); 4475 } 4476 4477 if (!mStandby) { 4478 mInput->stream->common.standby(&mInput->stream->common); 4479 } 4480 mActiveTrack.clear(); 4481 4482 mStartStopCond.broadcast(); 4483 4484 releaseWakeLock(); 4485 4486 ALOGV("RecordThread %p exiting", this); 4487 return false; 4488} 4489 4490 4491sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4492 const sp<AudioFlinger::Client>& client, 4493 uint32_t sampleRate, 4494 audio_format_t format, 4495 int channelMask, 4496 int frameCount, 4497 uint32_t flags, 4498 int sessionId, 4499 status_t *status) 4500{ 4501 sp<RecordTrack> track; 4502 status_t lStatus; 4503 4504 lStatus = initCheck(); 4505 if (lStatus != NO_ERROR) { 4506 ALOGE("Audio driver not initialized."); 4507 goto Exit; 4508 } 4509 4510 { // scope for mLock 4511 Mutex::Autolock _l(mLock); 4512 4513 track = new RecordTrack(this, client, sampleRate, 4514 format, channelMask, frameCount, flags, sessionId); 4515 4516 if (track->getCblk() == NULL) { 4517 lStatus = NO_MEMORY; 4518 goto Exit; 4519 } 4520 4521 mTrack = track.get(); 4522 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4523 bool suspend = audio_is_bluetooth_sco_device( 4524 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4525 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4526 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4527 } 4528 lStatus = NO_ERROR; 4529 4530Exit: 4531 if (status) { 4532 *status = lStatus; 4533 } 4534 return track; 4535} 4536 4537status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4538{ 4539 ALOGV("RecordThread::start"); 4540 sp <ThreadBase> strongMe = this; 4541 status_t status = NO_ERROR; 4542 { 4543 AutoMutex lock(mLock); 4544 if (mActiveTrack != 0) { 4545 if (recordTrack != mActiveTrack.get()) { 4546 status = -EBUSY; 4547 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4548 mActiveTrack->mState = TrackBase::ACTIVE; 4549 } 4550 return status; 4551 } 4552 4553 recordTrack->mState = TrackBase::IDLE; 4554 mActiveTrack = recordTrack; 4555 mLock.unlock(); 4556 status_t status = AudioSystem::startInput(mId); 4557 mLock.lock(); 4558 if (status != NO_ERROR) { 4559 mActiveTrack.clear(); 4560 return status; 4561 } 4562 mRsmpInIndex = mFrameCount; 4563 mBytesRead = 0; 4564 if (mResampler != NULL) { 4565 mResampler->reset(); 4566 } 4567 mActiveTrack->mState = TrackBase::RESUMING; 4568 // signal thread to start 4569 ALOGV("Signal record thread"); 4570 mWaitWorkCV.signal(); 4571 // do not wait for mStartStopCond if exiting 4572 if (mExiting) { 4573 mActiveTrack.clear(); 4574 status = INVALID_OPERATION; 4575 goto startError; 4576 } 4577 mStartStopCond.wait(mLock); 4578 if (mActiveTrack == 0) { 4579 ALOGV("Record failed to start"); 4580 status = BAD_VALUE; 4581 goto startError; 4582 } 4583 ALOGV("Record started OK"); 4584 return status; 4585 } 4586startError: 4587 AudioSystem::stopInput(mId); 4588 return status; 4589} 4590 4591void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4592 ALOGV("RecordThread::stop"); 4593 sp <ThreadBase> strongMe = this; 4594 { 4595 AutoMutex lock(mLock); 4596 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4597 mActiveTrack->mState = TrackBase::PAUSING; 4598 // do not wait for mStartStopCond if exiting 4599 if (mExiting) { 4600 return; 4601 } 4602 mStartStopCond.wait(mLock); 4603 // if we have been restarted, recordTrack == mActiveTrack.get() here 4604 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4605 mLock.unlock(); 4606 AudioSystem::stopInput(mId); 4607 mLock.lock(); 4608 ALOGV("Record stopped OK"); 4609 } 4610 } 4611 } 4612} 4613 4614status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4615{ 4616 const size_t SIZE = 256; 4617 char buffer[SIZE]; 4618 String8 result; 4619 pid_t pid = 0; 4620 4621 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4622 result.append(buffer); 4623 4624 if (mActiveTrack != 0) { 4625 result.append("Active Track:\n"); 4626 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4627 mActiveTrack->dump(buffer, SIZE); 4628 result.append(buffer); 4629 4630 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4631 result.append(buffer); 4632 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4633 result.append(buffer); 4634 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4635 result.append(buffer); 4636 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4637 result.append(buffer); 4638 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4639 result.append(buffer); 4640 4641 4642 } else { 4643 result.append("No record client\n"); 4644 } 4645 write(fd, result.string(), result.size()); 4646 4647 dumpBase(fd, args); 4648 dumpEffectChains(fd, args); 4649 4650 return NO_ERROR; 4651} 4652 4653status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4654{ 4655 size_t framesReq = buffer->frameCount; 4656 size_t framesReady = mFrameCount - mRsmpInIndex; 4657 int channelCount; 4658 4659 if (framesReady == 0) { 4660 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4661 if (mBytesRead < 0) { 4662 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4663 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4664 // Force input into standby so that it tries to 4665 // recover at next read attempt 4666 mInput->stream->common.standby(&mInput->stream->common); 4667 usleep(kRecordThreadSleepUs); 4668 } 4669 buffer->raw = NULL; 4670 buffer->frameCount = 0; 4671 return NOT_ENOUGH_DATA; 4672 } 4673 mRsmpInIndex = 0; 4674 framesReady = mFrameCount; 4675 } 4676 4677 if (framesReq > framesReady) { 4678 framesReq = framesReady; 4679 } 4680 4681 if (mChannelCount == 1 && mReqChannelCount == 2) { 4682 channelCount = 1; 4683 } else { 4684 channelCount = 2; 4685 } 4686 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4687 buffer->frameCount = framesReq; 4688 return NO_ERROR; 4689} 4690 4691void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4692{ 4693 mRsmpInIndex += buffer->frameCount; 4694 buffer->frameCount = 0; 4695} 4696 4697bool AudioFlinger::RecordThread::checkForNewParameters_l() 4698{ 4699 bool reconfig = false; 4700 4701 while (!mNewParameters.isEmpty()) { 4702 status_t status = NO_ERROR; 4703 String8 keyValuePair = mNewParameters[0]; 4704 AudioParameter param = AudioParameter(keyValuePair); 4705 int value; 4706 audio_format_t reqFormat = mFormat; 4707 int reqSamplingRate = mReqSampleRate; 4708 int reqChannelCount = mReqChannelCount; 4709 4710 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4711 reqSamplingRate = value; 4712 reconfig = true; 4713 } 4714 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4715 reqFormat = (audio_format_t) value; 4716 reconfig = true; 4717 } 4718 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4719 reqChannelCount = popcount(value); 4720 reconfig = true; 4721 } 4722 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4723 // do not accept frame count changes if tracks are open as the track buffer 4724 // size depends on frame count and correct behavior would not be garantied 4725 // if frame count is changed after track creation 4726 if (mActiveTrack != 0) { 4727 status = INVALID_OPERATION; 4728 } else { 4729 reconfig = true; 4730 } 4731 } 4732 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4733 // forward device change to effects that have requested to be 4734 // aware of attached audio device. 4735 for (size_t i = 0; i < mEffectChains.size(); i++) { 4736 mEffectChains[i]->setDevice_l(value); 4737 } 4738 // store input device and output device but do not forward output device to audio HAL. 4739 // Note that status is ignored by the caller for output device 4740 // (see AudioFlinger::setParameters() 4741 if (value & AUDIO_DEVICE_OUT_ALL) { 4742 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4743 status = BAD_VALUE; 4744 } else { 4745 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4746 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4747 if (mTrack != NULL) { 4748 bool suspend = audio_is_bluetooth_sco_device( 4749 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4750 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4751 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4752 } 4753 } 4754 mDevice |= (uint32_t)value; 4755 } 4756 if (status == NO_ERROR) { 4757 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4758 if (status == INVALID_OPERATION) { 4759 mInput->stream->common.standby(&mInput->stream->common); 4760 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4761 } 4762 if (reconfig) { 4763 if (status == BAD_VALUE && 4764 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4765 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4766 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4767 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4768 (reqChannelCount < 3)) { 4769 status = NO_ERROR; 4770 } 4771 if (status == NO_ERROR) { 4772 readInputParameters(); 4773 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4774 } 4775 } 4776 } 4777 4778 mNewParameters.removeAt(0); 4779 4780 mParamStatus = status; 4781 mParamCond.signal(); 4782 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4783 // already timed out waiting for the status and will never signal the condition. 4784 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4785 } 4786 return reconfig; 4787} 4788 4789String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4790{ 4791 char *s; 4792 String8 out_s8 = String8(); 4793 4794 Mutex::Autolock _l(mLock); 4795 if (initCheck() != NO_ERROR) { 4796 return out_s8; 4797 } 4798 4799 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4800 out_s8 = String8(s); 4801 free(s); 4802 return out_s8; 4803} 4804 4805void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4806 AudioSystem::OutputDescriptor desc; 4807 void *param2 = 0; 4808 4809 switch (event) { 4810 case AudioSystem::INPUT_OPENED: 4811 case AudioSystem::INPUT_CONFIG_CHANGED: 4812 desc.channels = mChannelMask; 4813 desc.samplingRate = mSampleRate; 4814 desc.format = mFormat; 4815 desc.frameCount = mFrameCount; 4816 desc.latency = 0; 4817 param2 = &desc; 4818 break; 4819 4820 case AudioSystem::INPUT_CLOSED: 4821 default: 4822 break; 4823 } 4824 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4825} 4826 4827void AudioFlinger::RecordThread::readInputParameters() 4828{ 4829 if (mRsmpInBuffer) delete mRsmpInBuffer; 4830 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4831 if (mResampler) delete mResampler; 4832 mResampler = NULL; 4833 4834 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4835 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4836 mChannelCount = (uint16_t)popcount(mChannelMask); 4837 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4838 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4839 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4840 mFrameCount = mInputBytes / mFrameSize; 4841 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4842 4843 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4844 { 4845 int channelCount; 4846 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4847 // stereo to mono post process as the resampler always outputs stereo. 4848 if (mChannelCount == 1 && mReqChannelCount == 2) { 4849 channelCount = 1; 4850 } else { 4851 channelCount = 2; 4852 } 4853 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4854 mResampler->setSampleRate(mSampleRate); 4855 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4856 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4857 4858 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4859 if (mChannelCount == 1 && mReqChannelCount == 1) { 4860 mFrameCount >>= 1; 4861 } 4862 4863 } 4864 mRsmpInIndex = mFrameCount; 4865} 4866 4867unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4868{ 4869 Mutex::Autolock _l(mLock); 4870 if (initCheck() != NO_ERROR) { 4871 return 0; 4872 } 4873 4874 return mInput->stream->get_input_frames_lost(mInput->stream); 4875} 4876 4877uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4878{ 4879 Mutex::Autolock _l(mLock); 4880 uint32_t result = 0; 4881 if (getEffectChain_l(sessionId) != 0) { 4882 result = EFFECT_SESSION; 4883 } 4884 4885 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4886 result |= TRACK_SESSION; 4887 } 4888 4889 return result; 4890} 4891 4892AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4893{ 4894 Mutex::Autolock _l(mLock); 4895 return mTrack; 4896} 4897 4898AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() 4899{ 4900 Mutex::Autolock _l(mLock); 4901 return mInput; 4902} 4903 4904AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4905{ 4906 Mutex::Autolock _l(mLock); 4907 AudioStreamIn *input = mInput; 4908 mInput = NULL; 4909 return input; 4910} 4911 4912// this method must always be called either with ThreadBase mLock held or inside the thread loop 4913audio_stream_t* AudioFlinger::RecordThread::stream() 4914{ 4915 if (mInput == NULL) { 4916 return NULL; 4917 } 4918 return &mInput->stream->common; 4919} 4920 4921 4922// ---------------------------------------------------------------------------- 4923 4924int AudioFlinger::openOutput(uint32_t *pDevices, 4925 uint32_t *pSamplingRate, 4926 audio_format_t *pFormat, 4927 uint32_t *pChannels, 4928 uint32_t *pLatencyMs, 4929 uint32_t flags) 4930{ 4931 status_t status; 4932 PlaybackThread *thread = NULL; 4933 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4934 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4935 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4936 uint32_t channels = pChannels ? *pChannels : 0; 4937 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4938 audio_stream_out_t *outStream; 4939 audio_hw_device_t *outHwDev; 4940 4941 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4942 pDevices ? *pDevices : 0, 4943 samplingRate, 4944 format, 4945 channels, 4946 flags); 4947 4948 if (pDevices == NULL || *pDevices == 0) { 4949 return 0; 4950 } 4951 4952 Mutex::Autolock _l(mLock); 4953 4954 outHwDev = findSuitableHwDev_l(*pDevices); 4955 if (outHwDev == NULL) 4956 return 0; 4957 4958 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4959 &channels, &samplingRate, &outStream); 4960 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4961 outStream, 4962 samplingRate, 4963 format, 4964 channels, 4965 status); 4966 4967 mHardwareStatus = AUDIO_HW_IDLE; 4968 if (outStream != NULL) { 4969 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4970 int id = nextUniqueId(); 4971 4972 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4973 (format != AUDIO_FORMAT_PCM_16_BIT) || 4974 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4975 thread = new DirectOutputThread(this, output, id, *pDevices); 4976 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4977 } else { 4978 thread = new MixerThread(this, output, id, *pDevices); 4979 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4980 } 4981 mPlaybackThreads.add(id, thread); 4982 4983 if (pSamplingRate) *pSamplingRate = samplingRate; 4984 if (pFormat) *pFormat = format; 4985 if (pChannels) *pChannels = channels; 4986 if (pLatencyMs) *pLatencyMs = thread->latency(); 4987 4988 // notify client processes of the new output creation 4989 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4990 return id; 4991 } 4992 4993 return 0; 4994} 4995 4996int AudioFlinger::openDuplicateOutput(int output1, int output2) 4997{ 4998 Mutex::Autolock _l(mLock); 4999 MixerThread *thread1 = checkMixerThread_l(output1); 5000 MixerThread *thread2 = checkMixerThread_l(output2); 5001 5002 if (thread1 == NULL || thread2 == NULL) { 5003 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5004 return 0; 5005 } 5006 5007 int id = nextUniqueId(); 5008 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5009 thread->addOutputTrack(thread2); 5010 mPlaybackThreads.add(id, thread); 5011 // notify client processes of the new output creation 5012 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5013 return id; 5014} 5015 5016status_t AudioFlinger::closeOutput(int output) 5017{ 5018 // keep strong reference on the playback thread so that 5019 // it is not destroyed while exit() is executed 5020 sp <PlaybackThread> thread; 5021 { 5022 Mutex::Autolock _l(mLock); 5023 thread = checkPlaybackThread_l(output); 5024 if (thread == NULL) { 5025 return BAD_VALUE; 5026 } 5027 5028 ALOGV("closeOutput() %d", output); 5029 5030 if (thread->type() == ThreadBase::MIXER) { 5031 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5032 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5033 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5034 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5035 } 5036 } 5037 } 5038 void *param2 = 0; 5039 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5040 mPlaybackThreads.removeItem(output); 5041 } 5042 thread->exit(); 5043 5044 if (thread->type() != ThreadBase::DUPLICATING) { 5045 AudioStreamOut *out = thread->clearOutput(); 5046 // from now on thread->mOutput is NULL 5047 out->hwDev->close_output_stream(out->hwDev, out->stream); 5048 delete out; 5049 } 5050 return NO_ERROR; 5051} 5052 5053status_t AudioFlinger::suspendOutput(int output) 5054{ 5055 Mutex::Autolock _l(mLock); 5056 PlaybackThread *thread = checkPlaybackThread_l(output); 5057 5058 if (thread == NULL) { 5059 return BAD_VALUE; 5060 } 5061 5062 ALOGV("suspendOutput() %d", output); 5063 thread->suspend(); 5064 5065 return NO_ERROR; 5066} 5067 5068status_t AudioFlinger::restoreOutput(int output) 5069{ 5070 Mutex::Autolock _l(mLock); 5071 PlaybackThread *thread = checkPlaybackThread_l(output); 5072 5073 if (thread == NULL) { 5074 return BAD_VALUE; 5075 } 5076 5077 ALOGV("restoreOutput() %d", output); 5078 5079 thread->restore(); 5080 5081 return NO_ERROR; 5082} 5083 5084int AudioFlinger::openInput(uint32_t *pDevices, 5085 uint32_t *pSamplingRate, 5086 audio_format_t *pFormat, 5087 uint32_t *pChannels, 5088 uint32_t acoustics) 5089{ 5090 status_t status; 5091 RecordThread *thread = NULL; 5092 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5093 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5094 uint32_t channels = pChannels ? *pChannels : 0; 5095 uint32_t reqSamplingRate = samplingRate; 5096 audio_format_t reqFormat = format; 5097 uint32_t reqChannels = channels; 5098 audio_stream_in_t *inStream; 5099 audio_hw_device_t *inHwDev; 5100 5101 if (pDevices == NULL || *pDevices == 0) { 5102 return 0; 5103 } 5104 5105 Mutex::Autolock _l(mLock); 5106 5107 inHwDev = findSuitableHwDev_l(*pDevices); 5108 if (inHwDev == NULL) 5109 return 0; 5110 5111 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5112 &channels, &samplingRate, 5113 (audio_in_acoustics_t)acoustics, 5114 &inStream); 5115 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5116 inStream, 5117 samplingRate, 5118 format, 5119 channels, 5120 acoustics, 5121 status); 5122 5123 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5124 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5125 // or stereo to mono conversions on 16 bit PCM inputs. 5126 if (inStream == NULL && status == BAD_VALUE && 5127 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5128 (samplingRate <= 2 * reqSamplingRate) && 5129 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5130 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5131 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5132 &channels, &samplingRate, 5133 (audio_in_acoustics_t)acoustics, 5134 &inStream); 5135 } 5136 5137 if (inStream != NULL) { 5138 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5139 5140 int id = nextUniqueId(); 5141 // Start record thread 5142 // RecorThread require both input and output device indication to forward to audio 5143 // pre processing modules 5144 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5145 thread = new RecordThread(this, 5146 input, 5147 reqSamplingRate, 5148 reqChannels, 5149 id, 5150 device); 5151 mRecordThreads.add(id, thread); 5152 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5153 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5154 if (pFormat) *pFormat = format; 5155 if (pChannels) *pChannels = reqChannels; 5156 5157 input->stream->common.standby(&input->stream->common); 5158 5159 // notify client processes of the new input creation 5160 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5161 return id; 5162 } 5163 5164 return 0; 5165} 5166 5167status_t AudioFlinger::closeInput(int input) 5168{ 5169 // keep strong reference on the record thread so that 5170 // it is not destroyed while exit() is executed 5171 sp <RecordThread> thread; 5172 { 5173 Mutex::Autolock _l(mLock); 5174 thread = checkRecordThread_l(input); 5175 if (thread == NULL) { 5176 return BAD_VALUE; 5177 } 5178 5179 ALOGV("closeInput() %d", input); 5180 void *param2 = 0; 5181 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5182 mRecordThreads.removeItem(input); 5183 } 5184 thread->exit(); 5185 5186 AudioStreamIn *in = thread->clearInput(); 5187 // from now on thread->mInput is NULL 5188 in->hwDev->close_input_stream(in->hwDev, in->stream); 5189 delete in; 5190 5191 return NO_ERROR; 5192} 5193 5194status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5195{ 5196 Mutex::Autolock _l(mLock); 5197 MixerThread *dstThread = checkMixerThread_l(output); 5198 if (dstThread == NULL) { 5199 ALOGW("setStreamOutput() bad output id %d", output); 5200 return BAD_VALUE; 5201 } 5202 5203 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5204 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5205 5206 dstThread->setStreamValid(stream, true); 5207 5208 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5209 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5210 if (thread != dstThread && 5211 thread->type() != ThreadBase::DIRECT) { 5212 MixerThread *srcThread = (MixerThread *)thread; 5213 srcThread->setStreamValid(stream, false); 5214 srcThread->invalidateTracks(stream); 5215 } 5216 } 5217 5218 return NO_ERROR; 5219} 5220 5221 5222int AudioFlinger::newAudioSessionId() 5223{ 5224 return nextUniqueId(); 5225} 5226 5227void AudioFlinger::acquireAudioSessionId(int audioSession) 5228{ 5229 Mutex::Autolock _l(mLock); 5230 int caller = IPCThreadState::self()->getCallingPid(); 5231 ALOGV("acquiring %d from %d", audioSession, caller); 5232 int num = mAudioSessionRefs.size(); 5233 for (int i = 0; i< num; i++) { 5234 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5235 if (ref->sessionid == audioSession && ref->pid == caller) { 5236 ref->cnt++; 5237 ALOGV(" incremented refcount to %d", ref->cnt); 5238 return; 5239 } 5240 } 5241 AudioSessionRef *ref = new AudioSessionRef(); 5242 ref->sessionid = audioSession; 5243 ref->pid = caller; 5244 ref->cnt = 1; 5245 mAudioSessionRefs.push(ref); 5246 ALOGV(" added new entry for %d", ref->sessionid); 5247} 5248 5249void AudioFlinger::releaseAudioSessionId(int audioSession) 5250{ 5251 Mutex::Autolock _l(mLock); 5252 int caller = IPCThreadState::self()->getCallingPid(); 5253 ALOGV("releasing %d from %d", audioSession, caller); 5254 int num = mAudioSessionRefs.size(); 5255 for (int i = 0; i< num; i++) { 5256 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5257 if (ref->sessionid == audioSession && ref->pid == caller) { 5258 ref->cnt--; 5259 ALOGV(" decremented refcount to %d", ref->cnt); 5260 if (ref->cnt == 0) { 5261 mAudioSessionRefs.removeAt(i); 5262 delete ref; 5263 purgeStaleEffects_l(); 5264 } 5265 return; 5266 } 5267 } 5268 ALOGW("session id %d not found for pid %d", audioSession, caller); 5269} 5270 5271void AudioFlinger::purgeStaleEffects_l() { 5272 5273 ALOGV("purging stale effects"); 5274 5275 Vector< sp<EffectChain> > chains; 5276 5277 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5278 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5279 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5280 sp<EffectChain> ec = t->mEffectChains[j]; 5281 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5282 chains.push(ec); 5283 } 5284 } 5285 } 5286 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5287 sp<RecordThread> t = mRecordThreads.valueAt(i); 5288 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5289 sp<EffectChain> ec = t->mEffectChains[j]; 5290 chains.push(ec); 5291 } 5292 } 5293 5294 for (size_t i = 0; i < chains.size(); i++) { 5295 sp<EffectChain> ec = chains[i]; 5296 int sessionid = ec->sessionId(); 5297 sp<ThreadBase> t = ec->mThread.promote(); 5298 if (t == 0) { 5299 continue; 5300 } 5301 size_t numsessionrefs = mAudioSessionRefs.size(); 5302 bool found = false; 5303 for (size_t k = 0; k < numsessionrefs; k++) { 5304 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5305 if (ref->sessionid == sessionid) { 5306 ALOGV(" session %d still exists for %d with %d refs", 5307 sessionid, ref->pid, ref->cnt); 5308 found = true; 5309 break; 5310 } 5311 } 5312 if (!found) { 5313 // remove all effects from the chain 5314 while (ec->mEffects.size()) { 5315 sp<EffectModule> effect = ec->mEffects[0]; 5316 effect->unPin(); 5317 Mutex::Autolock _l (t->mLock); 5318 t->removeEffect_l(effect); 5319 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5320 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5321 if (handle != 0) { 5322 handle->mEffect.clear(); 5323 if (handle->mHasControl && handle->mEnabled) { 5324 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5325 } 5326 } 5327 } 5328 AudioSystem::unregisterEffect(effect->id()); 5329 } 5330 } 5331 } 5332 return; 5333} 5334 5335// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5336AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5337{ 5338 PlaybackThread *thread = NULL; 5339 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5340 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5341 } 5342 return thread; 5343} 5344 5345// checkMixerThread_l() must be called with AudioFlinger::mLock held 5346AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5347{ 5348 PlaybackThread *thread = checkPlaybackThread_l(output); 5349 if (thread != NULL) { 5350 if (thread->type() == ThreadBase::DIRECT) { 5351 thread = NULL; 5352 } 5353 } 5354 return (MixerThread *)thread; 5355} 5356 5357// checkRecordThread_l() must be called with AudioFlinger::mLock held 5358AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5359{ 5360 RecordThread *thread = NULL; 5361 if (mRecordThreads.indexOfKey(input) >= 0) { 5362 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5363 } 5364 return thread; 5365} 5366 5367uint32_t AudioFlinger::nextUniqueId() 5368{ 5369 return android_atomic_inc(&mNextUniqueId); 5370} 5371 5372AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5373{ 5374 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5375 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5376 AudioStreamOut *output = thread->getOutput(); 5377 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5378 return thread; 5379 } 5380 } 5381 return NULL; 5382} 5383 5384uint32_t AudioFlinger::primaryOutputDevice_l() 5385{ 5386 PlaybackThread *thread = primaryPlaybackThread_l(); 5387 5388 if (thread == NULL) { 5389 return 0; 5390 } 5391 5392 return thread->device(); 5393} 5394 5395 5396// ---------------------------------------------------------------------------- 5397// Effect management 5398// ---------------------------------------------------------------------------- 5399 5400 5401status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5402{ 5403 Mutex::Autolock _l(mLock); 5404 return EffectQueryNumberEffects(numEffects); 5405} 5406 5407status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5408{ 5409 Mutex::Autolock _l(mLock); 5410 return EffectQueryEffect(index, descriptor); 5411} 5412 5413status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5414{ 5415 Mutex::Autolock _l(mLock); 5416 return EffectGetDescriptor(pUuid, descriptor); 5417} 5418 5419 5420sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5421 effect_descriptor_t *pDesc, 5422 const sp<IEffectClient>& effectClient, 5423 int32_t priority, 5424 int io, 5425 int sessionId, 5426 status_t *status, 5427 int *id, 5428 int *enabled) 5429{ 5430 status_t lStatus = NO_ERROR; 5431 sp<EffectHandle> handle; 5432 effect_descriptor_t desc; 5433 sp<Client> client; 5434 wp<Client> wclient; 5435 5436 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5437 pid, effectClient.get(), priority, sessionId, io); 5438 5439 if (pDesc == NULL) { 5440 lStatus = BAD_VALUE; 5441 goto Exit; 5442 } 5443 5444 // check audio settings permission for global effects 5445 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5446 lStatus = PERMISSION_DENIED; 5447 goto Exit; 5448 } 5449 5450 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5451 // that can only be created by audio policy manager (running in same process) 5452 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5453 lStatus = PERMISSION_DENIED; 5454 goto Exit; 5455 } 5456 5457 if (io == 0) { 5458 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5459 // output must be specified by AudioPolicyManager when using session 5460 // AUDIO_SESSION_OUTPUT_STAGE 5461 lStatus = BAD_VALUE; 5462 goto Exit; 5463 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5464 // if the output returned by getOutputForEffect() is removed before we lock the 5465 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5466 // and we will exit safely 5467 io = AudioSystem::getOutputForEffect(&desc); 5468 } 5469 } 5470 5471 { 5472 Mutex::Autolock _l(mLock); 5473 5474 5475 if (!EffectIsNullUuid(&pDesc->uuid)) { 5476 // if uuid is specified, request effect descriptor 5477 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5478 if (lStatus < 0) { 5479 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5480 goto Exit; 5481 } 5482 } else { 5483 // if uuid is not specified, look for an available implementation 5484 // of the required type in effect factory 5485 if (EffectIsNullUuid(&pDesc->type)) { 5486 ALOGW("createEffect() no effect type"); 5487 lStatus = BAD_VALUE; 5488 goto Exit; 5489 } 5490 uint32_t numEffects = 0; 5491 effect_descriptor_t d; 5492 d.flags = 0; // prevent compiler warning 5493 bool found = false; 5494 5495 lStatus = EffectQueryNumberEffects(&numEffects); 5496 if (lStatus < 0) { 5497 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5498 goto Exit; 5499 } 5500 for (uint32_t i = 0; i < numEffects; i++) { 5501 lStatus = EffectQueryEffect(i, &desc); 5502 if (lStatus < 0) { 5503 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5504 continue; 5505 } 5506 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5507 // If matching type found save effect descriptor. If the session is 5508 // 0 and the effect is not auxiliary, continue enumeration in case 5509 // an auxiliary version of this effect type is available 5510 found = true; 5511 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5512 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5513 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5514 break; 5515 } 5516 } 5517 } 5518 if (!found) { 5519 lStatus = BAD_VALUE; 5520 ALOGW("createEffect() effect not found"); 5521 goto Exit; 5522 } 5523 // For same effect type, chose auxiliary version over insert version if 5524 // connect to output mix (Compliance to OpenSL ES) 5525 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5526 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5527 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5528 } 5529 } 5530 5531 // Do not allow auxiliary effects on a session different from 0 (output mix) 5532 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5533 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5534 lStatus = INVALID_OPERATION; 5535 goto Exit; 5536 } 5537 5538 // check recording permission for visualizer 5539 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5540 !recordingAllowed()) { 5541 lStatus = PERMISSION_DENIED; 5542 goto Exit; 5543 } 5544 5545 // return effect descriptor 5546 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5547 5548 // If output is not specified try to find a matching audio session ID in one of the 5549 // output threads. 5550 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5551 // because of code checking output when entering the function. 5552 // Note: io is never 0 when creating an effect on an input 5553 if (io == 0) { 5554 // look for the thread where the specified audio session is present 5555 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5556 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5557 io = mPlaybackThreads.keyAt(i); 5558 break; 5559 } 5560 } 5561 if (io == 0) { 5562 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5563 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5564 io = mRecordThreads.keyAt(i); 5565 break; 5566 } 5567 } 5568 } 5569 // If no output thread contains the requested session ID, default to 5570 // first output. The effect chain will be moved to the correct output 5571 // thread when a track with the same session ID is created 5572 if (io == 0 && mPlaybackThreads.size()) { 5573 io = mPlaybackThreads.keyAt(0); 5574 } 5575 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5576 } 5577 ThreadBase *thread = checkRecordThread_l(io); 5578 if (thread == NULL) { 5579 thread = checkPlaybackThread_l(io); 5580 if (thread == NULL) { 5581 ALOGE("createEffect() unknown output thread"); 5582 lStatus = BAD_VALUE; 5583 goto Exit; 5584 } 5585 } 5586 5587 wclient = mClients.valueFor(pid); 5588 5589 if (wclient != NULL) { 5590 client = wclient.promote(); 5591 } else { 5592 client = new Client(this, pid); 5593 mClients.add(pid, client); 5594 } 5595 5596 // create effect on selected output thread 5597 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5598 &desc, enabled, &lStatus); 5599 if (handle != 0 && id != NULL) { 5600 *id = handle->id(); 5601 } 5602 } 5603 5604Exit: 5605 if(status) { 5606 *status = lStatus; 5607 } 5608 return handle; 5609} 5610 5611status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5612{ 5613 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5614 sessionId, srcOutput, dstOutput); 5615 Mutex::Autolock _l(mLock); 5616 if (srcOutput == dstOutput) { 5617 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5618 return NO_ERROR; 5619 } 5620 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5621 if (srcThread == NULL) { 5622 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5623 return BAD_VALUE; 5624 } 5625 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5626 if (dstThread == NULL) { 5627 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5628 return BAD_VALUE; 5629 } 5630 5631 Mutex::Autolock _dl(dstThread->mLock); 5632 Mutex::Autolock _sl(srcThread->mLock); 5633 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5634 5635 return NO_ERROR; 5636} 5637 5638// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5639status_t AudioFlinger::moveEffectChain_l(int sessionId, 5640 AudioFlinger::PlaybackThread *srcThread, 5641 AudioFlinger::PlaybackThread *dstThread, 5642 bool reRegister) 5643{ 5644 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5645 sessionId, srcThread, dstThread); 5646 5647 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5648 if (chain == 0) { 5649 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5650 sessionId, srcThread); 5651 return INVALID_OPERATION; 5652 } 5653 5654 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5655 // so that a new chain is created with correct parameters when first effect is added. This is 5656 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5657 // removed. 5658 srcThread->removeEffectChain_l(chain); 5659 5660 // transfer all effects one by one so that new effect chain is created on new thread with 5661 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5662 int dstOutput = dstThread->id(); 5663 sp<EffectChain> dstChain; 5664 uint32_t strategy = 0; // prevent compiler warning 5665 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5666 while (effect != 0) { 5667 srcThread->removeEffect_l(effect); 5668 dstThread->addEffect_l(effect); 5669 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5670 if (effect->state() == EffectModule::ACTIVE || 5671 effect->state() == EffectModule::STOPPING) { 5672 effect->start(); 5673 } 5674 // if the move request is not received from audio policy manager, the effect must be 5675 // re-registered with the new strategy and output 5676 if (dstChain == 0) { 5677 dstChain = effect->chain().promote(); 5678 if (dstChain == 0) { 5679 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5680 srcThread->addEffect_l(effect); 5681 return NO_INIT; 5682 } 5683 strategy = dstChain->strategy(); 5684 } 5685 if (reRegister) { 5686 AudioSystem::unregisterEffect(effect->id()); 5687 AudioSystem::registerEffect(&effect->desc(), 5688 dstOutput, 5689 strategy, 5690 sessionId, 5691 effect->id()); 5692 } 5693 effect = chain->getEffectFromId_l(0); 5694 } 5695 5696 return NO_ERROR; 5697} 5698 5699 5700// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5701sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5702 const sp<AudioFlinger::Client>& client, 5703 const sp<IEffectClient>& effectClient, 5704 int32_t priority, 5705 int sessionId, 5706 effect_descriptor_t *desc, 5707 int *enabled, 5708 status_t *status 5709 ) 5710{ 5711 sp<EffectModule> effect; 5712 sp<EffectHandle> handle; 5713 status_t lStatus; 5714 sp<EffectChain> chain; 5715 bool chainCreated = false; 5716 bool effectCreated = false; 5717 bool effectRegistered = false; 5718 5719 lStatus = initCheck(); 5720 if (lStatus != NO_ERROR) { 5721 ALOGW("createEffect_l() Audio driver not initialized."); 5722 goto Exit; 5723 } 5724 5725 // Do not allow effects with session ID 0 on direct output or duplicating threads 5726 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5727 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5728 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5729 desc->name, sessionId); 5730 lStatus = BAD_VALUE; 5731 goto Exit; 5732 } 5733 // Only Pre processor effects are allowed on input threads and only on input threads 5734 if ((mType == RECORD && 5735 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5736 (mType != RECORD && 5737 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5738 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5739 desc->name, desc->flags, mType); 5740 lStatus = BAD_VALUE; 5741 goto Exit; 5742 } 5743 5744 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5745 5746 { // scope for mLock 5747 Mutex::Autolock _l(mLock); 5748 5749 // check for existing effect chain with the requested audio session 5750 chain = getEffectChain_l(sessionId); 5751 if (chain == 0) { 5752 // create a new chain for this session 5753 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5754 chain = new EffectChain(this, sessionId); 5755 addEffectChain_l(chain); 5756 chain->setStrategy(getStrategyForSession_l(sessionId)); 5757 chainCreated = true; 5758 } else { 5759 effect = chain->getEffectFromDesc_l(desc); 5760 } 5761 5762 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5763 5764 if (effect == 0) { 5765 int id = mAudioFlinger->nextUniqueId(); 5766 // Check CPU and memory usage 5767 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5768 if (lStatus != NO_ERROR) { 5769 goto Exit; 5770 } 5771 effectRegistered = true; 5772 // create a new effect module if none present in the chain 5773 effect = new EffectModule(this, chain, desc, id, sessionId); 5774 lStatus = effect->status(); 5775 if (lStatus != NO_ERROR) { 5776 goto Exit; 5777 } 5778 lStatus = chain->addEffect_l(effect); 5779 if (lStatus != NO_ERROR) { 5780 goto Exit; 5781 } 5782 effectCreated = true; 5783 5784 effect->setDevice(mDevice); 5785 effect->setMode(mAudioFlinger->getMode()); 5786 } 5787 // create effect handle and connect it to effect module 5788 handle = new EffectHandle(effect, client, effectClient, priority); 5789 lStatus = effect->addHandle(handle); 5790 if (enabled) { 5791 *enabled = (int)effect->isEnabled(); 5792 } 5793 } 5794 5795Exit: 5796 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5797 Mutex::Autolock _l(mLock); 5798 if (effectCreated) { 5799 chain->removeEffect_l(effect); 5800 } 5801 if (effectRegistered) { 5802 AudioSystem::unregisterEffect(effect->id()); 5803 } 5804 if (chainCreated) { 5805 removeEffectChain_l(chain); 5806 } 5807 handle.clear(); 5808 } 5809 5810 if(status) { 5811 *status = lStatus; 5812 } 5813 return handle; 5814} 5815 5816sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5817{ 5818 sp<EffectModule> effect; 5819 5820 sp<EffectChain> chain = getEffectChain_l(sessionId); 5821 if (chain != 0) { 5822 effect = chain->getEffectFromId_l(effectId); 5823 } 5824 return effect; 5825} 5826 5827// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5828// PlaybackThread::mLock held 5829status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5830{ 5831 // check for existing effect chain with the requested audio session 5832 int sessionId = effect->sessionId(); 5833 sp<EffectChain> chain = getEffectChain_l(sessionId); 5834 bool chainCreated = false; 5835 5836 if (chain == 0) { 5837 // create a new chain for this session 5838 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5839 chain = new EffectChain(this, sessionId); 5840 addEffectChain_l(chain); 5841 chain->setStrategy(getStrategyForSession_l(sessionId)); 5842 chainCreated = true; 5843 } 5844 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5845 5846 if (chain->getEffectFromId_l(effect->id()) != 0) { 5847 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5848 this, effect->desc().name, chain.get()); 5849 return BAD_VALUE; 5850 } 5851 5852 status_t status = chain->addEffect_l(effect); 5853 if (status != NO_ERROR) { 5854 if (chainCreated) { 5855 removeEffectChain_l(chain); 5856 } 5857 return status; 5858 } 5859 5860 effect->setDevice(mDevice); 5861 effect->setMode(mAudioFlinger->getMode()); 5862 return NO_ERROR; 5863} 5864 5865void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5866 5867 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5868 effect_descriptor_t desc = effect->desc(); 5869 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5870 detachAuxEffect_l(effect->id()); 5871 } 5872 5873 sp<EffectChain> chain = effect->chain().promote(); 5874 if (chain != 0) { 5875 // remove effect chain if removing last effect 5876 if (chain->removeEffect_l(effect) == 0) { 5877 removeEffectChain_l(chain); 5878 } 5879 } else { 5880 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5881 } 5882} 5883 5884void AudioFlinger::ThreadBase::lockEffectChains_l( 5885 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5886{ 5887 effectChains = mEffectChains; 5888 for (size_t i = 0; i < mEffectChains.size(); i++) { 5889 mEffectChains[i]->lock(); 5890 } 5891} 5892 5893void AudioFlinger::ThreadBase::unlockEffectChains( 5894 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5895{ 5896 for (size_t i = 0; i < effectChains.size(); i++) { 5897 effectChains[i]->unlock(); 5898 } 5899} 5900 5901sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5902{ 5903 Mutex::Autolock _l(mLock); 5904 return getEffectChain_l(sessionId); 5905} 5906 5907sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5908{ 5909 sp<EffectChain> chain; 5910 5911 size_t size = mEffectChains.size(); 5912 for (size_t i = 0; i < size; i++) { 5913 if (mEffectChains[i]->sessionId() == sessionId) { 5914 chain = mEffectChains[i]; 5915 break; 5916 } 5917 } 5918 return chain; 5919} 5920 5921void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5922{ 5923 Mutex::Autolock _l(mLock); 5924 size_t size = mEffectChains.size(); 5925 for (size_t i = 0; i < size; i++) { 5926 mEffectChains[i]->setMode_l(mode); 5927 } 5928} 5929 5930void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5931 const wp<EffectHandle>& handle, 5932 bool unpiniflast) { 5933 5934 Mutex::Autolock _l(mLock); 5935 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5936 // delete the effect module if removing last handle on it 5937 if (effect->removeHandle(handle) == 0) { 5938 if (!effect->isPinned() || unpiniflast) { 5939 removeEffect_l(effect); 5940 AudioSystem::unregisterEffect(effect->id()); 5941 } 5942 } 5943} 5944 5945status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5946{ 5947 int session = chain->sessionId(); 5948 int16_t *buffer = mMixBuffer; 5949 bool ownsBuffer = false; 5950 5951 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5952 if (session > 0) { 5953 // Only one effect chain can be present in direct output thread and it uses 5954 // the mix buffer as input 5955 if (mType != DIRECT) { 5956 size_t numSamples = mFrameCount * mChannelCount; 5957 buffer = new int16_t[numSamples]; 5958 memset(buffer, 0, numSamples * sizeof(int16_t)); 5959 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5960 ownsBuffer = true; 5961 } 5962 5963 // Attach all tracks with same session ID to this chain. 5964 for (size_t i = 0; i < mTracks.size(); ++i) { 5965 sp<Track> track = mTracks[i]; 5966 if (session == track->sessionId()) { 5967 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5968 track->setMainBuffer(buffer); 5969 chain->incTrackCnt(); 5970 } 5971 } 5972 5973 // indicate all active tracks in the chain 5974 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5975 sp<Track> track = mActiveTracks[i].promote(); 5976 if (track == 0) continue; 5977 if (session == track->sessionId()) { 5978 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5979 chain->incActiveTrackCnt(); 5980 } 5981 } 5982 } 5983 5984 chain->setInBuffer(buffer, ownsBuffer); 5985 chain->setOutBuffer(mMixBuffer); 5986 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5987 // chains list in order to be processed last as it contains output stage effects 5988 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5989 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5990 // after track specific effects and before output stage 5991 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5992 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5993 // Effect chain for other sessions are inserted at beginning of effect 5994 // chains list to be processed before output mix effects. Relative order between other 5995 // sessions is not important 5996 size_t size = mEffectChains.size(); 5997 size_t i = 0; 5998 for (i = 0; i < size; i++) { 5999 if (mEffectChains[i]->sessionId() < session) break; 6000 } 6001 mEffectChains.insertAt(chain, i); 6002 checkSuspendOnAddEffectChain_l(chain); 6003 6004 return NO_ERROR; 6005} 6006 6007size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6008{ 6009 int session = chain->sessionId(); 6010 6011 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6012 6013 for (size_t i = 0; i < mEffectChains.size(); i++) { 6014 if (chain == mEffectChains[i]) { 6015 mEffectChains.removeAt(i); 6016 // detach all active tracks from the chain 6017 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6018 sp<Track> track = mActiveTracks[i].promote(); 6019 if (track == 0) continue; 6020 if (session == track->sessionId()) { 6021 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6022 chain.get(), session); 6023 chain->decActiveTrackCnt(); 6024 } 6025 } 6026 6027 // detach all tracks with same session ID from this chain 6028 for (size_t i = 0; i < mTracks.size(); ++i) { 6029 sp<Track> track = mTracks[i]; 6030 if (session == track->sessionId()) { 6031 track->setMainBuffer(mMixBuffer); 6032 chain->decTrackCnt(); 6033 } 6034 } 6035 break; 6036 } 6037 } 6038 return mEffectChains.size(); 6039} 6040 6041status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6042 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6043{ 6044 Mutex::Autolock _l(mLock); 6045 return attachAuxEffect_l(track, EffectId); 6046} 6047 6048status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6049 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6050{ 6051 status_t status = NO_ERROR; 6052 6053 if (EffectId == 0) { 6054 track->setAuxBuffer(0, NULL); 6055 } else { 6056 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6057 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6058 if (effect != 0) { 6059 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6060 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6061 } else { 6062 status = INVALID_OPERATION; 6063 } 6064 } else { 6065 status = BAD_VALUE; 6066 } 6067 } 6068 return status; 6069} 6070 6071void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6072{ 6073 for (size_t i = 0; i < mTracks.size(); ++i) { 6074 sp<Track> track = mTracks[i]; 6075 if (track->auxEffectId() == effectId) { 6076 attachAuxEffect_l(track, 0); 6077 } 6078 } 6079} 6080 6081status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6082{ 6083 // only one chain per input thread 6084 if (mEffectChains.size() != 0) { 6085 return INVALID_OPERATION; 6086 } 6087 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6088 6089 chain->setInBuffer(NULL); 6090 chain->setOutBuffer(NULL); 6091 6092 checkSuspendOnAddEffectChain_l(chain); 6093 6094 mEffectChains.add(chain); 6095 6096 return NO_ERROR; 6097} 6098 6099size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6100{ 6101 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6102 ALOGW_IF(mEffectChains.size() != 1, 6103 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6104 chain.get(), mEffectChains.size(), this); 6105 if (mEffectChains.size() == 1) { 6106 mEffectChains.removeAt(0); 6107 } 6108 return 0; 6109} 6110 6111// ---------------------------------------------------------------------------- 6112// EffectModule implementation 6113// ---------------------------------------------------------------------------- 6114 6115#undef LOG_TAG 6116#define LOG_TAG "AudioFlinger::EffectModule" 6117 6118AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6119 const wp<AudioFlinger::EffectChain>& chain, 6120 effect_descriptor_t *desc, 6121 int id, 6122 int sessionId) 6123 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6124 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6125{ 6126 ALOGV("Constructor %p", this); 6127 int lStatus; 6128 sp<ThreadBase> thread = mThread.promote(); 6129 if (thread == 0) { 6130 return; 6131 } 6132 6133 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6134 6135 // create effect engine from effect factory 6136 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6137 6138 if (mStatus != NO_ERROR) { 6139 return; 6140 } 6141 lStatus = init(); 6142 if (lStatus < 0) { 6143 mStatus = lStatus; 6144 goto Error; 6145 } 6146 6147 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6148 mPinned = true; 6149 } 6150 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6151 return; 6152Error: 6153 EffectRelease(mEffectInterface); 6154 mEffectInterface = NULL; 6155 ALOGV("Constructor Error %d", mStatus); 6156} 6157 6158AudioFlinger::EffectModule::~EffectModule() 6159{ 6160 ALOGV("Destructor %p", this); 6161 if (mEffectInterface != NULL) { 6162 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6163 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6164 sp<ThreadBase> thread = mThread.promote(); 6165 if (thread != 0) { 6166 audio_stream_t *stream = thread->stream(); 6167 if (stream != NULL) { 6168 stream->remove_audio_effect(stream, mEffectInterface); 6169 } 6170 } 6171 } 6172 // release effect engine 6173 EffectRelease(mEffectInterface); 6174 } 6175} 6176 6177status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6178{ 6179 status_t status; 6180 6181 Mutex::Autolock _l(mLock); 6182 // First handle in mHandles has highest priority and controls the effect module 6183 int priority = handle->priority(); 6184 size_t size = mHandles.size(); 6185 sp<EffectHandle> h; 6186 size_t i; 6187 for (i = 0; i < size; i++) { 6188 h = mHandles[i].promote(); 6189 if (h == 0) continue; 6190 if (h->priority() <= priority) break; 6191 } 6192 // if inserted in first place, move effect control from previous owner to this handle 6193 if (i == 0) { 6194 bool enabled = false; 6195 if (h != 0) { 6196 enabled = h->enabled(); 6197 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6198 } 6199 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6200 status = NO_ERROR; 6201 } else { 6202 status = ALREADY_EXISTS; 6203 } 6204 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6205 mHandles.insertAt(handle, i); 6206 return status; 6207} 6208 6209size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6210{ 6211 Mutex::Autolock _l(mLock); 6212 size_t size = mHandles.size(); 6213 size_t i; 6214 for (i = 0; i < size; i++) { 6215 if (mHandles[i] == handle) break; 6216 } 6217 if (i == size) { 6218 return size; 6219 } 6220 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6221 6222 bool enabled = false; 6223 EffectHandle *hdl = handle.unsafe_get(); 6224 if (hdl) { 6225 ALOGV("removeHandle() unsafe_get OK"); 6226 enabled = hdl->enabled(); 6227 } 6228 mHandles.removeAt(i); 6229 size = mHandles.size(); 6230 // if removed from first place, move effect control from this handle to next in line 6231 if (i == 0 && size != 0) { 6232 sp<EffectHandle> h = mHandles[0].promote(); 6233 if (h != 0) { 6234 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6235 } 6236 } 6237 6238 // Prevent calls to process() and other functions on effect interface from now on. 6239 // The effect engine will be released by the destructor when the last strong reference on 6240 // this object is released which can happen after next process is called. 6241 if (size == 0 && !mPinned) { 6242 mState = DESTROYED; 6243 } 6244 6245 return size; 6246} 6247 6248sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6249{ 6250 Mutex::Autolock _l(mLock); 6251 sp<EffectHandle> handle; 6252 if (mHandles.size() != 0) { 6253 handle = mHandles[0].promote(); 6254 } 6255 return handle; 6256} 6257 6258void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6259{ 6260 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6261 // keep a strong reference on this EffectModule to avoid calling the 6262 // destructor before we exit 6263 sp<EffectModule> keep(this); 6264 { 6265 sp<ThreadBase> thread = mThread.promote(); 6266 if (thread != 0) { 6267 thread->disconnectEffect(keep, handle, unpiniflast); 6268 } 6269 } 6270} 6271 6272void AudioFlinger::EffectModule::updateState() { 6273 Mutex::Autolock _l(mLock); 6274 6275 switch (mState) { 6276 case RESTART: 6277 reset_l(); 6278 // FALL THROUGH 6279 6280 case STARTING: 6281 // clear auxiliary effect input buffer for next accumulation 6282 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6283 memset(mConfig.inputCfg.buffer.raw, 6284 0, 6285 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6286 } 6287 start_l(); 6288 mState = ACTIVE; 6289 break; 6290 case STOPPING: 6291 stop_l(); 6292 mDisableWaitCnt = mMaxDisableWaitCnt; 6293 mState = STOPPED; 6294 break; 6295 case STOPPED: 6296 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6297 // turn off sequence. 6298 if (--mDisableWaitCnt == 0) { 6299 reset_l(); 6300 mState = IDLE; 6301 } 6302 break; 6303 default: //IDLE , ACTIVE, DESTROYED 6304 break; 6305 } 6306} 6307 6308void AudioFlinger::EffectModule::process() 6309{ 6310 Mutex::Autolock _l(mLock); 6311 6312 if (mState == DESTROYED || mEffectInterface == NULL || 6313 mConfig.inputCfg.buffer.raw == NULL || 6314 mConfig.outputCfg.buffer.raw == NULL) { 6315 return; 6316 } 6317 6318 if (isProcessEnabled()) { 6319 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6320 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6321 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6322 mConfig.inputCfg.buffer.s32, 6323 mConfig.inputCfg.buffer.frameCount/2); 6324 } 6325 6326 // do the actual processing in the effect engine 6327 int ret = (*mEffectInterface)->process(mEffectInterface, 6328 &mConfig.inputCfg.buffer, 6329 &mConfig.outputCfg.buffer); 6330 6331 // force transition to IDLE state when engine is ready 6332 if (mState == STOPPED && ret == -ENODATA) { 6333 mDisableWaitCnt = 1; 6334 } 6335 6336 // clear auxiliary effect input buffer for next accumulation 6337 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6338 memset(mConfig.inputCfg.buffer.raw, 0, 6339 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6340 } 6341 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6342 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6343 // If an insert effect is idle and input buffer is different from output buffer, 6344 // accumulate input onto output 6345 sp<EffectChain> chain = mChain.promote(); 6346 if (chain != 0 && chain->activeTrackCnt() != 0) { 6347 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6348 int16_t *in = mConfig.inputCfg.buffer.s16; 6349 int16_t *out = mConfig.outputCfg.buffer.s16; 6350 for (size_t i = 0; i < frameCnt; i++) { 6351 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6352 } 6353 } 6354 } 6355} 6356 6357void AudioFlinger::EffectModule::reset_l() 6358{ 6359 if (mEffectInterface == NULL) { 6360 return; 6361 } 6362 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6363} 6364 6365status_t AudioFlinger::EffectModule::configure() 6366{ 6367 uint32_t channels; 6368 if (mEffectInterface == NULL) { 6369 return NO_INIT; 6370 } 6371 6372 sp<ThreadBase> thread = mThread.promote(); 6373 if (thread == 0) { 6374 return DEAD_OBJECT; 6375 } 6376 6377 // TODO: handle configuration of effects replacing track process 6378 if (thread->channelCount() == 1) { 6379 channels = AUDIO_CHANNEL_OUT_MONO; 6380 } else { 6381 channels = AUDIO_CHANNEL_OUT_STEREO; 6382 } 6383 6384 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6385 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6386 } else { 6387 mConfig.inputCfg.channels = channels; 6388 } 6389 mConfig.outputCfg.channels = channels; 6390 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6391 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6392 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6393 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6394 mConfig.inputCfg.bufferProvider.cookie = NULL; 6395 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6396 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6397 mConfig.outputCfg.bufferProvider.cookie = NULL; 6398 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6399 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6400 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6401 // Insert effect: 6402 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6403 // always overwrites output buffer: input buffer == output buffer 6404 // - in other sessions: 6405 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6406 // other effect: overwrites output buffer: input buffer == output buffer 6407 // Auxiliary effect: 6408 // accumulates in output buffer: input buffer != output buffer 6409 // Therefore: accumulate <=> input buffer != output buffer 6410 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6411 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6412 } else { 6413 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6414 } 6415 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6416 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6417 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6418 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6419 6420 ALOGV("configure() %p thread %p buffer %p framecount %d", 6421 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6422 6423 status_t cmdStatus; 6424 uint32_t size = sizeof(int); 6425 status_t status = (*mEffectInterface)->command(mEffectInterface, 6426 EFFECT_CMD_SET_CONFIG, 6427 sizeof(effect_config_t), 6428 &mConfig, 6429 &size, 6430 &cmdStatus); 6431 if (status == 0) { 6432 status = cmdStatus; 6433 } 6434 6435 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6436 (1000 * mConfig.outputCfg.buffer.frameCount); 6437 6438 return status; 6439} 6440 6441status_t AudioFlinger::EffectModule::init() 6442{ 6443 Mutex::Autolock _l(mLock); 6444 if (mEffectInterface == NULL) { 6445 return NO_INIT; 6446 } 6447 status_t cmdStatus; 6448 uint32_t size = sizeof(status_t); 6449 status_t status = (*mEffectInterface)->command(mEffectInterface, 6450 EFFECT_CMD_INIT, 6451 0, 6452 NULL, 6453 &size, 6454 &cmdStatus); 6455 if (status == 0) { 6456 status = cmdStatus; 6457 } 6458 return status; 6459} 6460 6461status_t AudioFlinger::EffectModule::start() 6462{ 6463 Mutex::Autolock _l(mLock); 6464 return start_l(); 6465} 6466 6467status_t AudioFlinger::EffectModule::start_l() 6468{ 6469 if (mEffectInterface == NULL) { 6470 return NO_INIT; 6471 } 6472 status_t cmdStatus; 6473 uint32_t size = sizeof(status_t); 6474 status_t status = (*mEffectInterface)->command(mEffectInterface, 6475 EFFECT_CMD_ENABLE, 6476 0, 6477 NULL, 6478 &size, 6479 &cmdStatus); 6480 if (status == 0) { 6481 status = cmdStatus; 6482 } 6483 if (status == 0 && 6484 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6485 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6486 sp<ThreadBase> thread = mThread.promote(); 6487 if (thread != 0) { 6488 audio_stream_t *stream = thread->stream(); 6489 if (stream != NULL) { 6490 stream->add_audio_effect(stream, mEffectInterface); 6491 } 6492 } 6493 } 6494 return status; 6495} 6496 6497status_t AudioFlinger::EffectModule::stop() 6498{ 6499 Mutex::Autolock _l(mLock); 6500 return stop_l(); 6501} 6502 6503status_t AudioFlinger::EffectModule::stop_l() 6504{ 6505 if (mEffectInterface == NULL) { 6506 return NO_INIT; 6507 } 6508 status_t cmdStatus; 6509 uint32_t size = sizeof(status_t); 6510 status_t status = (*mEffectInterface)->command(mEffectInterface, 6511 EFFECT_CMD_DISABLE, 6512 0, 6513 NULL, 6514 &size, 6515 &cmdStatus); 6516 if (status == 0) { 6517 status = cmdStatus; 6518 } 6519 if (status == 0 && 6520 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6521 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6522 sp<ThreadBase> thread = mThread.promote(); 6523 if (thread != 0) { 6524 audio_stream_t *stream = thread->stream(); 6525 if (stream != NULL) { 6526 stream->remove_audio_effect(stream, mEffectInterface); 6527 } 6528 } 6529 } 6530 return status; 6531} 6532 6533status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6534 uint32_t cmdSize, 6535 void *pCmdData, 6536 uint32_t *replySize, 6537 void *pReplyData) 6538{ 6539 Mutex::Autolock _l(mLock); 6540// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6541 6542 if (mState == DESTROYED || mEffectInterface == NULL) { 6543 return NO_INIT; 6544 } 6545 status_t status = (*mEffectInterface)->command(mEffectInterface, 6546 cmdCode, 6547 cmdSize, 6548 pCmdData, 6549 replySize, 6550 pReplyData); 6551 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6552 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6553 for (size_t i = 1; i < mHandles.size(); i++) { 6554 sp<EffectHandle> h = mHandles[i].promote(); 6555 if (h != 0) { 6556 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6557 } 6558 } 6559 } 6560 return status; 6561} 6562 6563status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6564{ 6565 6566 Mutex::Autolock _l(mLock); 6567 ALOGV("setEnabled %p enabled %d", this, enabled); 6568 6569 if (enabled != isEnabled()) { 6570 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6571 if (enabled && status != NO_ERROR) { 6572 return status; 6573 } 6574 6575 switch (mState) { 6576 // going from disabled to enabled 6577 case IDLE: 6578 mState = STARTING; 6579 break; 6580 case STOPPED: 6581 mState = RESTART; 6582 break; 6583 case STOPPING: 6584 mState = ACTIVE; 6585 break; 6586 6587 // going from enabled to disabled 6588 case RESTART: 6589 mState = STOPPED; 6590 break; 6591 case STARTING: 6592 mState = IDLE; 6593 break; 6594 case ACTIVE: 6595 mState = STOPPING; 6596 break; 6597 case DESTROYED: 6598 return NO_ERROR; // simply ignore as we are being destroyed 6599 } 6600 for (size_t i = 1; i < mHandles.size(); i++) { 6601 sp<EffectHandle> h = mHandles[i].promote(); 6602 if (h != 0) { 6603 h->setEnabled(enabled); 6604 } 6605 } 6606 } 6607 return NO_ERROR; 6608} 6609 6610bool AudioFlinger::EffectModule::isEnabled() 6611{ 6612 switch (mState) { 6613 case RESTART: 6614 case STARTING: 6615 case ACTIVE: 6616 return true; 6617 case IDLE: 6618 case STOPPING: 6619 case STOPPED: 6620 case DESTROYED: 6621 default: 6622 return false; 6623 } 6624} 6625 6626bool AudioFlinger::EffectModule::isProcessEnabled() 6627{ 6628 switch (mState) { 6629 case RESTART: 6630 case ACTIVE: 6631 case STOPPING: 6632 case STOPPED: 6633 return true; 6634 case IDLE: 6635 case STARTING: 6636 case DESTROYED: 6637 default: 6638 return false; 6639 } 6640} 6641 6642status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6643{ 6644 Mutex::Autolock _l(mLock); 6645 status_t status = NO_ERROR; 6646 6647 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6648 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6649 if (isProcessEnabled() && 6650 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6651 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6652 status_t cmdStatus; 6653 uint32_t volume[2]; 6654 uint32_t *pVolume = NULL; 6655 uint32_t size = sizeof(volume); 6656 volume[0] = *left; 6657 volume[1] = *right; 6658 if (controller) { 6659 pVolume = volume; 6660 } 6661 status = (*mEffectInterface)->command(mEffectInterface, 6662 EFFECT_CMD_SET_VOLUME, 6663 size, 6664 volume, 6665 &size, 6666 pVolume); 6667 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6668 *left = volume[0]; 6669 *right = volume[1]; 6670 } 6671 } 6672 return status; 6673} 6674 6675status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6676{ 6677 Mutex::Autolock _l(mLock); 6678 status_t status = NO_ERROR; 6679 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6680 // audio pre processing modules on RecordThread can receive both output and 6681 // input device indication in the same call 6682 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6683 if (dev) { 6684 status_t cmdStatus; 6685 uint32_t size = sizeof(status_t); 6686 6687 status = (*mEffectInterface)->command(mEffectInterface, 6688 EFFECT_CMD_SET_DEVICE, 6689 sizeof(uint32_t), 6690 &dev, 6691 &size, 6692 &cmdStatus); 6693 if (status == NO_ERROR) { 6694 status = cmdStatus; 6695 } 6696 } 6697 dev = device & AUDIO_DEVICE_IN_ALL; 6698 if (dev) { 6699 status_t cmdStatus; 6700 uint32_t size = sizeof(status_t); 6701 6702 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6703 EFFECT_CMD_SET_INPUT_DEVICE, 6704 sizeof(uint32_t), 6705 &dev, 6706 &size, 6707 &cmdStatus); 6708 if (status2 == NO_ERROR) { 6709 status2 = cmdStatus; 6710 } 6711 if (status == NO_ERROR) { 6712 status = status2; 6713 } 6714 } 6715 } 6716 return status; 6717} 6718 6719status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6720{ 6721 Mutex::Autolock _l(mLock); 6722 status_t status = NO_ERROR; 6723 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6724 status_t cmdStatus; 6725 uint32_t size = sizeof(status_t); 6726 status = (*mEffectInterface)->command(mEffectInterface, 6727 EFFECT_CMD_SET_AUDIO_MODE, 6728 sizeof(audio_mode_t), 6729 &mode, 6730 &size, 6731 &cmdStatus); 6732 if (status == NO_ERROR) { 6733 status = cmdStatus; 6734 } 6735 } 6736 return status; 6737} 6738 6739void AudioFlinger::EffectModule::setSuspended(bool suspended) 6740{ 6741 Mutex::Autolock _l(mLock); 6742 mSuspended = suspended; 6743} 6744 6745bool AudioFlinger::EffectModule::suspended() const 6746{ 6747 Mutex::Autolock _l(mLock); 6748 return mSuspended; 6749} 6750 6751status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6752{ 6753 const size_t SIZE = 256; 6754 char buffer[SIZE]; 6755 String8 result; 6756 6757 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6758 result.append(buffer); 6759 6760 bool locked = tryLock(mLock); 6761 // failed to lock - AudioFlinger is probably deadlocked 6762 if (!locked) { 6763 result.append("\t\tCould not lock Fx mutex:\n"); 6764 } 6765 6766 result.append("\t\tSession Status State Engine:\n"); 6767 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6768 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6769 result.append(buffer); 6770 6771 result.append("\t\tDescriptor:\n"); 6772 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6773 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6774 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6775 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6776 result.append(buffer); 6777 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6778 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6779 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6780 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6781 result.append(buffer); 6782 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6783 mDescriptor.apiVersion, 6784 mDescriptor.flags); 6785 result.append(buffer); 6786 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6787 mDescriptor.name); 6788 result.append(buffer); 6789 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6790 mDescriptor.implementor); 6791 result.append(buffer); 6792 6793 result.append("\t\t- Input configuration:\n"); 6794 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6795 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6796 (uint32_t)mConfig.inputCfg.buffer.raw, 6797 mConfig.inputCfg.buffer.frameCount, 6798 mConfig.inputCfg.samplingRate, 6799 mConfig.inputCfg.channels, 6800 mConfig.inputCfg.format); 6801 result.append(buffer); 6802 6803 result.append("\t\t- Output configuration:\n"); 6804 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6805 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6806 (uint32_t)mConfig.outputCfg.buffer.raw, 6807 mConfig.outputCfg.buffer.frameCount, 6808 mConfig.outputCfg.samplingRate, 6809 mConfig.outputCfg.channels, 6810 mConfig.outputCfg.format); 6811 result.append(buffer); 6812 6813 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6814 result.append(buffer); 6815 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6816 for (size_t i = 0; i < mHandles.size(); ++i) { 6817 sp<EffectHandle> handle = mHandles[i].promote(); 6818 if (handle != 0) { 6819 handle->dump(buffer, SIZE); 6820 result.append(buffer); 6821 } 6822 } 6823 6824 result.append("\n"); 6825 6826 write(fd, result.string(), result.length()); 6827 6828 if (locked) { 6829 mLock.unlock(); 6830 } 6831 6832 return NO_ERROR; 6833} 6834 6835// ---------------------------------------------------------------------------- 6836// EffectHandle implementation 6837// ---------------------------------------------------------------------------- 6838 6839#undef LOG_TAG 6840#define LOG_TAG "AudioFlinger::EffectHandle" 6841 6842AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6843 const sp<AudioFlinger::Client>& client, 6844 const sp<IEffectClient>& effectClient, 6845 int32_t priority) 6846 : BnEffect(), 6847 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6848 mPriority(priority), mHasControl(false), mEnabled(false) 6849{ 6850 ALOGV("constructor %p", this); 6851 6852 if (client == 0) { 6853 return; 6854 } 6855 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6856 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6857 if (mCblkMemory != 0) { 6858 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6859 6860 if (mCblk) { 6861 new(mCblk) effect_param_cblk_t(); 6862 mBuffer = (uint8_t *)mCblk + bufOffset; 6863 } 6864 } else { 6865 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6866 return; 6867 } 6868} 6869 6870AudioFlinger::EffectHandle::~EffectHandle() 6871{ 6872 ALOGV("Destructor %p", this); 6873 disconnect(false); 6874 ALOGV("Destructor DONE %p", this); 6875} 6876 6877status_t AudioFlinger::EffectHandle::enable() 6878{ 6879 ALOGV("enable %p", this); 6880 if (!mHasControl) return INVALID_OPERATION; 6881 if (mEffect == 0) return DEAD_OBJECT; 6882 6883 if (mEnabled) { 6884 return NO_ERROR; 6885 } 6886 6887 mEnabled = true; 6888 6889 sp<ThreadBase> thread = mEffect->thread().promote(); 6890 if (thread != 0) { 6891 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6892 } 6893 6894 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6895 if (mEffect->suspended()) { 6896 return NO_ERROR; 6897 } 6898 6899 status_t status = mEffect->setEnabled(true); 6900 if (status != NO_ERROR) { 6901 if (thread != 0) { 6902 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6903 } 6904 mEnabled = false; 6905 } 6906 return status; 6907} 6908 6909status_t AudioFlinger::EffectHandle::disable() 6910{ 6911 ALOGV("disable %p", this); 6912 if (!mHasControl) return INVALID_OPERATION; 6913 if (mEffect == 0) return DEAD_OBJECT; 6914 6915 if (!mEnabled) { 6916 return NO_ERROR; 6917 } 6918 mEnabled = false; 6919 6920 if (mEffect->suspended()) { 6921 return NO_ERROR; 6922 } 6923 6924 status_t status = mEffect->setEnabled(false); 6925 6926 sp<ThreadBase> thread = mEffect->thread().promote(); 6927 if (thread != 0) { 6928 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6929 } 6930 6931 return status; 6932} 6933 6934void AudioFlinger::EffectHandle::disconnect() 6935{ 6936 disconnect(true); 6937} 6938 6939void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6940{ 6941 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6942 if (mEffect == 0) { 6943 return; 6944 } 6945 mEffect->disconnect(this, unpiniflast); 6946 6947 if (mHasControl && mEnabled) { 6948 sp<ThreadBase> thread = mEffect->thread().promote(); 6949 if (thread != 0) { 6950 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6951 } 6952 } 6953 6954 // release sp on module => module destructor can be called now 6955 mEffect.clear(); 6956 if (mClient != 0) { 6957 if (mCblk) { 6958 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6959 } 6960 mCblkMemory.clear(); // and free the shared memory 6961 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6962 mClient.clear(); 6963 } 6964} 6965 6966status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6967 uint32_t cmdSize, 6968 void *pCmdData, 6969 uint32_t *replySize, 6970 void *pReplyData) 6971{ 6972// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6973// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6974 6975 // only get parameter command is permitted for applications not controlling the effect 6976 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6977 return INVALID_OPERATION; 6978 } 6979 if (mEffect == 0) return DEAD_OBJECT; 6980 if (mClient == 0) return INVALID_OPERATION; 6981 6982 // handle commands that are not forwarded transparently to effect engine 6983 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6984 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6985 // no risk to block the whole media server process or mixer threads is we are stuck here 6986 Mutex::Autolock _l(mCblk->lock); 6987 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6988 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6989 mCblk->serverIndex = 0; 6990 mCblk->clientIndex = 0; 6991 return BAD_VALUE; 6992 } 6993 status_t status = NO_ERROR; 6994 while (mCblk->serverIndex < mCblk->clientIndex) { 6995 int reply; 6996 uint32_t rsize = sizeof(int); 6997 int *p = (int *)(mBuffer + mCblk->serverIndex); 6998 int size = *p++; 6999 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7000 ALOGW("command(): invalid parameter block size"); 7001 break; 7002 } 7003 effect_param_t *param = (effect_param_t *)p; 7004 if (param->psize == 0 || param->vsize == 0) { 7005 ALOGW("command(): null parameter or value size"); 7006 mCblk->serverIndex += size; 7007 continue; 7008 } 7009 uint32_t psize = sizeof(effect_param_t) + 7010 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7011 param->vsize; 7012 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7013 psize, 7014 p, 7015 &rsize, 7016 &reply); 7017 // stop at first error encountered 7018 if (ret != NO_ERROR) { 7019 status = ret; 7020 *(int *)pReplyData = reply; 7021 break; 7022 } else if (reply != NO_ERROR) { 7023 *(int *)pReplyData = reply; 7024 break; 7025 } 7026 mCblk->serverIndex += size; 7027 } 7028 mCblk->serverIndex = 0; 7029 mCblk->clientIndex = 0; 7030 return status; 7031 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7032 *(int *)pReplyData = NO_ERROR; 7033 return enable(); 7034 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7035 *(int *)pReplyData = NO_ERROR; 7036 return disable(); 7037 } 7038 7039 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7040} 7041 7042sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7043 return mCblkMemory; 7044} 7045 7046void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7047{ 7048 ALOGV("setControl %p control %d", this, hasControl); 7049 7050 mHasControl = hasControl; 7051 mEnabled = enabled; 7052 7053 if (signal && mEffectClient != 0) { 7054 mEffectClient->controlStatusChanged(hasControl); 7055 } 7056} 7057 7058void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7059 uint32_t cmdSize, 7060 void *pCmdData, 7061 uint32_t replySize, 7062 void *pReplyData) 7063{ 7064 if (mEffectClient != 0) { 7065 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7066 } 7067} 7068 7069 7070 7071void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7072{ 7073 if (mEffectClient != 0) { 7074 mEffectClient->enableStatusChanged(enabled); 7075 } 7076} 7077 7078status_t AudioFlinger::EffectHandle::onTransact( 7079 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7080{ 7081 return BnEffect::onTransact(code, data, reply, flags); 7082} 7083 7084 7085void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7086{ 7087 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7088 7089 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7090 (mClient == NULL) ? getpid() : mClient->pid(), 7091 mPriority, 7092 mHasControl, 7093 !locked, 7094 mCblk ? mCblk->clientIndex : 0, 7095 mCblk ? mCblk->serverIndex : 0 7096 ); 7097 7098 if (locked) { 7099 mCblk->lock.unlock(); 7100 } 7101} 7102 7103#undef LOG_TAG 7104#define LOG_TAG "AudioFlinger::EffectChain" 7105 7106AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7107 int sessionId) 7108 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7109 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7110 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7111{ 7112 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7113 sp<ThreadBase> thread = mThread.promote(); 7114 if (thread == 0) { 7115 return; 7116 } 7117 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7118 thread->frameCount(); 7119} 7120 7121AudioFlinger::EffectChain::~EffectChain() 7122{ 7123 if (mOwnInBuffer) { 7124 delete mInBuffer; 7125 } 7126 7127} 7128 7129// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7130sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7131{ 7132 sp<EffectModule> effect; 7133 size_t size = mEffects.size(); 7134 7135 for (size_t i = 0; i < size; i++) { 7136 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7137 effect = mEffects[i]; 7138 break; 7139 } 7140 } 7141 return effect; 7142} 7143 7144// getEffectFromId_l() must be called with ThreadBase::mLock held 7145sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7146{ 7147 sp<EffectModule> effect; 7148 size_t size = mEffects.size(); 7149 7150 for (size_t i = 0; i < size; i++) { 7151 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7152 if (id == 0 || mEffects[i]->id() == id) { 7153 effect = mEffects[i]; 7154 break; 7155 } 7156 } 7157 return effect; 7158} 7159 7160// getEffectFromType_l() must be called with ThreadBase::mLock held 7161sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7162 const effect_uuid_t *type) 7163{ 7164 sp<EffectModule> effect; 7165 size_t size = mEffects.size(); 7166 7167 for (size_t i = 0; i < size; i++) { 7168 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7169 effect = mEffects[i]; 7170 break; 7171 } 7172 } 7173 return effect; 7174} 7175 7176// Must be called with EffectChain::mLock locked 7177void AudioFlinger::EffectChain::process_l() 7178{ 7179 sp<ThreadBase> thread = mThread.promote(); 7180 if (thread == 0) { 7181 ALOGW("process_l(): cannot promote mixer thread"); 7182 return; 7183 } 7184 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7185 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7186 // always process effects unless no more tracks are on the session and the effect tail 7187 // has been rendered 7188 bool doProcess = true; 7189 if (!isGlobalSession) { 7190 bool tracksOnSession = (trackCnt() != 0); 7191 7192 if (!tracksOnSession && mTailBufferCount == 0) { 7193 doProcess = false; 7194 } 7195 7196 if (activeTrackCnt() == 0) { 7197 // if no track is active and the effect tail has not been rendered, 7198 // the input buffer must be cleared here as the mixer process will not do it 7199 if (tracksOnSession || mTailBufferCount > 0) { 7200 size_t numSamples = thread->frameCount() * thread->channelCount(); 7201 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7202 if (mTailBufferCount > 0) { 7203 mTailBufferCount--; 7204 } 7205 } 7206 } 7207 } 7208 7209 size_t size = mEffects.size(); 7210 if (doProcess) { 7211 for (size_t i = 0; i < size; i++) { 7212 mEffects[i]->process(); 7213 } 7214 } 7215 for (size_t i = 0; i < size; i++) { 7216 mEffects[i]->updateState(); 7217 } 7218} 7219 7220// addEffect_l() must be called with PlaybackThread::mLock held 7221status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7222{ 7223 effect_descriptor_t desc = effect->desc(); 7224 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7225 7226 Mutex::Autolock _l(mLock); 7227 effect->setChain(this); 7228 sp<ThreadBase> thread = mThread.promote(); 7229 if (thread == 0) { 7230 return NO_INIT; 7231 } 7232 effect->setThread(thread); 7233 7234 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7235 // Auxiliary effects are inserted at the beginning of mEffects vector as 7236 // they are processed first and accumulated in chain input buffer 7237 mEffects.insertAt(effect, 0); 7238 7239 // the input buffer for auxiliary effect contains mono samples in 7240 // 32 bit format. This is to avoid saturation in AudoMixer 7241 // accumulation stage. Saturation is done in EffectModule::process() before 7242 // calling the process in effect engine 7243 size_t numSamples = thread->frameCount(); 7244 int32_t *buffer = new int32_t[numSamples]; 7245 memset(buffer, 0, numSamples * sizeof(int32_t)); 7246 effect->setInBuffer((int16_t *)buffer); 7247 // auxiliary effects output samples to chain input buffer for further processing 7248 // by insert effects 7249 effect->setOutBuffer(mInBuffer); 7250 } else { 7251 // Insert effects are inserted at the end of mEffects vector as they are processed 7252 // after track and auxiliary effects. 7253 // Insert effect order as a function of indicated preference: 7254 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7255 // another effect is present 7256 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7257 // last effect claiming first position 7258 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7259 // first effect claiming last position 7260 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7261 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7262 // already present 7263 7264 int size = (int)mEffects.size(); 7265 int idx_insert = size; 7266 int idx_insert_first = -1; 7267 int idx_insert_last = -1; 7268 7269 for (int i = 0; i < size; i++) { 7270 effect_descriptor_t d = mEffects[i]->desc(); 7271 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7272 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7273 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7274 // check invalid effect chaining combinations 7275 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7276 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7277 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7278 return INVALID_OPERATION; 7279 } 7280 // remember position of first insert effect and by default 7281 // select this as insert position for new effect 7282 if (idx_insert == size) { 7283 idx_insert = i; 7284 } 7285 // remember position of last insert effect claiming 7286 // first position 7287 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7288 idx_insert_first = i; 7289 } 7290 // remember position of first insert effect claiming 7291 // last position 7292 if (iPref == EFFECT_FLAG_INSERT_LAST && 7293 idx_insert_last == -1) { 7294 idx_insert_last = i; 7295 } 7296 } 7297 } 7298 7299 // modify idx_insert from first position if needed 7300 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7301 if (idx_insert_last != -1) { 7302 idx_insert = idx_insert_last; 7303 } else { 7304 idx_insert = size; 7305 } 7306 } else { 7307 if (idx_insert_first != -1) { 7308 idx_insert = idx_insert_first + 1; 7309 } 7310 } 7311 7312 // always read samples from chain input buffer 7313 effect->setInBuffer(mInBuffer); 7314 7315 // if last effect in the chain, output samples to chain 7316 // output buffer, otherwise to chain input buffer 7317 if (idx_insert == size) { 7318 if (idx_insert != 0) { 7319 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7320 mEffects[idx_insert-1]->configure(); 7321 } 7322 effect->setOutBuffer(mOutBuffer); 7323 } else { 7324 effect->setOutBuffer(mInBuffer); 7325 } 7326 mEffects.insertAt(effect, idx_insert); 7327 7328 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7329 } 7330 effect->configure(); 7331 return NO_ERROR; 7332} 7333 7334// removeEffect_l() must be called with PlaybackThread::mLock held 7335size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7336{ 7337 Mutex::Autolock _l(mLock); 7338 int size = (int)mEffects.size(); 7339 int i; 7340 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7341 7342 for (i = 0; i < size; i++) { 7343 if (effect == mEffects[i]) { 7344 // calling stop here will remove pre-processing effect from the audio HAL. 7345 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7346 // the middle of a read from audio HAL 7347 if (mEffects[i]->state() == EffectModule::ACTIVE || 7348 mEffects[i]->state() == EffectModule::STOPPING) { 7349 mEffects[i]->stop(); 7350 } 7351 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7352 delete[] effect->inBuffer(); 7353 } else { 7354 if (i == size - 1 && i != 0) { 7355 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7356 mEffects[i - 1]->configure(); 7357 } 7358 } 7359 mEffects.removeAt(i); 7360 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7361 break; 7362 } 7363 } 7364 7365 return mEffects.size(); 7366} 7367 7368// setDevice_l() must be called with PlaybackThread::mLock held 7369void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7370{ 7371 size_t size = mEffects.size(); 7372 for (size_t i = 0; i < size; i++) { 7373 mEffects[i]->setDevice(device); 7374 } 7375} 7376 7377// setMode_l() must be called with PlaybackThread::mLock held 7378void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7379{ 7380 size_t size = mEffects.size(); 7381 for (size_t i = 0; i < size; i++) { 7382 mEffects[i]->setMode(mode); 7383 } 7384} 7385 7386// setVolume_l() must be called with PlaybackThread::mLock held 7387bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7388{ 7389 uint32_t newLeft = *left; 7390 uint32_t newRight = *right; 7391 bool hasControl = false; 7392 int ctrlIdx = -1; 7393 size_t size = mEffects.size(); 7394 7395 // first update volume controller 7396 for (size_t i = size; i > 0; i--) { 7397 if (mEffects[i - 1]->isProcessEnabled() && 7398 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7399 ctrlIdx = i - 1; 7400 hasControl = true; 7401 break; 7402 } 7403 } 7404 7405 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7406 if (hasControl) { 7407 *left = mNewLeftVolume; 7408 *right = mNewRightVolume; 7409 } 7410 return hasControl; 7411 } 7412 7413 mVolumeCtrlIdx = ctrlIdx; 7414 mLeftVolume = newLeft; 7415 mRightVolume = newRight; 7416 7417 // second get volume update from volume controller 7418 if (ctrlIdx >= 0) { 7419 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7420 mNewLeftVolume = newLeft; 7421 mNewRightVolume = newRight; 7422 } 7423 // then indicate volume to all other effects in chain. 7424 // Pass altered volume to effects before volume controller 7425 // and requested volume to effects after controller 7426 uint32_t lVol = newLeft; 7427 uint32_t rVol = newRight; 7428 7429 for (size_t i = 0; i < size; i++) { 7430 if ((int)i == ctrlIdx) continue; 7431 // this also works for ctrlIdx == -1 when there is no volume controller 7432 if ((int)i > ctrlIdx) { 7433 lVol = *left; 7434 rVol = *right; 7435 } 7436 mEffects[i]->setVolume(&lVol, &rVol, false); 7437 } 7438 *left = newLeft; 7439 *right = newRight; 7440 7441 return hasControl; 7442} 7443 7444status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7445{ 7446 const size_t SIZE = 256; 7447 char buffer[SIZE]; 7448 String8 result; 7449 7450 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7451 result.append(buffer); 7452 7453 bool locked = tryLock(mLock); 7454 // failed to lock - AudioFlinger is probably deadlocked 7455 if (!locked) { 7456 result.append("\tCould not lock mutex:\n"); 7457 } 7458 7459 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7460 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7461 mEffects.size(), 7462 (uint32_t)mInBuffer, 7463 (uint32_t)mOutBuffer, 7464 mActiveTrackCnt); 7465 result.append(buffer); 7466 write(fd, result.string(), result.size()); 7467 7468 for (size_t i = 0; i < mEffects.size(); ++i) { 7469 sp<EffectModule> effect = mEffects[i]; 7470 if (effect != 0) { 7471 effect->dump(fd, args); 7472 } 7473 } 7474 7475 if (locked) { 7476 mLock.unlock(); 7477 } 7478 7479 return NO_ERROR; 7480} 7481 7482// must be called with ThreadBase::mLock held 7483void AudioFlinger::EffectChain::setEffectSuspended_l( 7484 const effect_uuid_t *type, bool suspend) 7485{ 7486 sp<SuspendedEffectDesc> desc; 7487 // use effect type UUID timelow as key as there is no real risk of identical 7488 // timeLow fields among effect type UUIDs. 7489 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7490 if (suspend) { 7491 if (index >= 0) { 7492 desc = mSuspendedEffects.valueAt(index); 7493 } else { 7494 desc = new SuspendedEffectDesc(); 7495 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7496 mSuspendedEffects.add(type->timeLow, desc); 7497 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7498 } 7499 if (desc->mRefCount++ == 0) { 7500 sp<EffectModule> effect = getEffectIfEnabled(type); 7501 if (effect != 0) { 7502 desc->mEffect = effect; 7503 effect->setSuspended(true); 7504 effect->setEnabled(false); 7505 } 7506 } 7507 } else { 7508 if (index < 0) { 7509 return; 7510 } 7511 desc = mSuspendedEffects.valueAt(index); 7512 if (desc->mRefCount <= 0) { 7513 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7514 desc->mRefCount = 1; 7515 } 7516 if (--desc->mRefCount == 0) { 7517 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7518 if (desc->mEffect != 0) { 7519 sp<EffectModule> effect = desc->mEffect.promote(); 7520 if (effect != 0) { 7521 effect->setSuspended(false); 7522 sp<EffectHandle> handle = effect->controlHandle(); 7523 if (handle != 0) { 7524 effect->setEnabled(handle->enabled()); 7525 } 7526 } 7527 desc->mEffect.clear(); 7528 } 7529 mSuspendedEffects.removeItemsAt(index); 7530 } 7531 } 7532} 7533 7534// must be called with ThreadBase::mLock held 7535void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7536{ 7537 sp<SuspendedEffectDesc> desc; 7538 7539 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7540 if (suspend) { 7541 if (index >= 0) { 7542 desc = mSuspendedEffects.valueAt(index); 7543 } else { 7544 desc = new SuspendedEffectDesc(); 7545 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7546 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7547 } 7548 if (desc->mRefCount++ == 0) { 7549 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7550 for (size_t i = 0; i < effects.size(); i++) { 7551 setEffectSuspended_l(&effects[i]->desc().type, true); 7552 } 7553 } 7554 } else { 7555 if (index < 0) { 7556 return; 7557 } 7558 desc = mSuspendedEffects.valueAt(index); 7559 if (desc->mRefCount <= 0) { 7560 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7561 desc->mRefCount = 1; 7562 } 7563 if (--desc->mRefCount == 0) { 7564 Vector<const effect_uuid_t *> types; 7565 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7566 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7567 continue; 7568 } 7569 types.add(&mSuspendedEffects.valueAt(i)->mType); 7570 } 7571 for (size_t i = 0; i < types.size(); i++) { 7572 setEffectSuspended_l(types[i], false); 7573 } 7574 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7575 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7576 } 7577 } 7578} 7579 7580 7581// The volume effect is used for automated tests only 7582#ifndef OPENSL_ES_H_ 7583static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7584 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7585const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7586#endif //OPENSL_ES_H_ 7587 7588bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7589{ 7590 // auxiliary effects and visualizer are never suspended on output mix 7591 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7592 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7593 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7594 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7595 return false; 7596 } 7597 return true; 7598} 7599 7600Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7601{ 7602 Vector< sp<EffectModule> > effects; 7603 for (size_t i = 0; i < mEffects.size(); i++) { 7604 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7605 continue; 7606 } 7607 effects.add(mEffects[i]); 7608 } 7609 return effects; 7610} 7611 7612sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7613 const effect_uuid_t *type) 7614{ 7615 sp<EffectModule> effect; 7616 effect = getEffectFromType_l(type); 7617 if (effect != 0 && !effect->isEnabled()) { 7618 effect.clear(); 7619 } 7620 return effect; 7621} 7622 7623void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7624 bool enabled) 7625{ 7626 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7627 if (enabled) { 7628 if (index < 0) { 7629 // if the effect is not suspend check if all effects are suspended 7630 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7631 if (index < 0) { 7632 return; 7633 } 7634 if (!isEffectEligibleForSuspend(effect->desc())) { 7635 return; 7636 } 7637 setEffectSuspended_l(&effect->desc().type, enabled); 7638 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7639 if (index < 0) { 7640 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7641 return; 7642 } 7643 } 7644 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7645 effect->desc().type.timeLow); 7646 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7647 // if effect is requested to suspended but was not yet enabled, supend it now. 7648 if (desc->mEffect == 0) { 7649 desc->mEffect = effect; 7650 effect->setEnabled(false); 7651 effect->setSuspended(true); 7652 } 7653 } else { 7654 if (index < 0) { 7655 return; 7656 } 7657 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7658 effect->desc().type.timeLow); 7659 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7660 desc->mEffect.clear(); 7661 effect->setSuspended(false); 7662 } 7663} 7664 7665#undef LOG_TAG 7666#define LOG_TAG "AudioFlinger" 7667 7668// ---------------------------------------------------------------------------- 7669 7670status_t AudioFlinger::onTransact( 7671 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7672{ 7673 return BnAudioFlinger::onTransact(code, data, reply, flags); 7674} 7675 7676}; // namespace android 7677