AudioFlinger.cpp revision dbfafaffe2e97eaf8d74ec6b6c468418a1ad2443
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <audio_effects/effect_visualizer.h> 54#include <audio_effects/effect_ns.h> 55#include <audio_effects/effect_aec.h> 56 57#include <audio_utils/primitives.h> 58 59#include <cpustats/ThreadCpuUsage.h> 60#include <powermanager/PowerManager.h> 61// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 62 63// ---------------------------------------------------------------------------- 64 65 66namespace android { 67 68static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 69static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 70 71//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 72static const float MAX_GAIN = 4096.0f; 73static const uint32_t MAX_GAIN_INT = 0x1000; 74 75// retry counts for buffer fill timeout 76// 50 * ~20msecs = 1 second 77static const int8_t kMaxTrackRetries = 50; 78static const int8_t kMaxTrackStartupRetries = 50; 79// allow less retry attempts on direct output thread. 80// direct outputs can be a scarce resource in audio hardware and should 81// be released as quickly as possible. 82static const int8_t kMaxTrackRetriesDirect = 2; 83 84static const int kDumpLockRetries = 50; 85static const int kDumpLockSleepUs = 20000; 86 87// don't warn about blocked writes or record buffer overflows more often than this 88static const nsecs_t kWarningThrottleNs = seconds(5); 89 90// RecordThread loop sleep time upon application overrun or audio HAL read error 91static const int kRecordThreadSleepUs = 5000; 92 93// maximum time to wait for setParameters to complete 94static const nsecs_t kSetParametersTimeoutNs = seconds(2); 95 96// minimum sleep time for the mixer thread loop when tracks are active but in underrun 97static const uint32_t kMinThreadSleepTimeUs = 5000; 98// maximum divider applied to the active sleep time in the mixer thread loop 99static const uint32_t kMaxThreadSleepTimeShift = 2; 100 101 102// ---------------------------------------------------------------------------- 103 104static bool recordingAllowed() { 105 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 106 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 107 if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO"); 108 return ok; 109} 110 111static bool settingsAllowed() { 112 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 113 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 114 if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 115 return ok; 116} 117 118// To collect the amplifier usage 119static void addBatteryData(uint32_t params) { 120 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 121 if (service == NULL) { 122 // it already logged 123 return; 124 } 125 126 service->addBatteryData(params); 127} 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 165 mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 166 mMode(AUDIO_MODE_INVALID), 167 mBtNrecIsOff(false) 168{ 169} 170 171void AudioFlinger::onFirstRef() 172{ 173 int rc = 0; 174 175 Mutex::Autolock _l(mLock); 176 177 /* TODO: move all this work into an Init() function */ 178 179 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 180 const hw_module_t *mod; 181 audio_hw_device_t *dev; 182 183 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 184 if (rc) 185 continue; 186 187 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 188 mod->name, mod->id); 189 mAudioHwDevs.push(dev); 190 191 if (!mPrimaryHardwareDev) { 192 mPrimaryHardwareDev = dev; 193 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 194 mod->name, mod->id, audio_interfaces[i]); 195 } 196 } 197 198 mHardwareStatus = AUDIO_HW_INIT; 199 200 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 201 ALOGE("Primary audio interface not found"); 202 return; 203 } 204 205 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 206 audio_hw_device_t *dev = mAudioHwDevs[i]; 207 208 mHardwareStatus = AUDIO_HW_INIT; 209 rc = dev->init_check(dev); 210 if (rc == 0) { 211 AutoMutex lock(mHardwareLock); 212 213 mMode = AUDIO_MODE_NORMAL; 214 mHardwareStatus = AUDIO_HW_SET_MODE; 215 dev->set_mode(dev, mMode); 216 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 217 dev->set_master_volume(dev, 1.0f); 218 mHardwareStatus = AUDIO_HW_IDLE; 219 } 220 } 221} 222 223status_t AudioFlinger::initCheck() const 224{ 225 Mutex::Autolock _l(mLock); 226 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 227 return NO_INIT; 228 return NO_ERROR; 229} 230 231AudioFlinger::~AudioFlinger() 232{ 233 int num_devs = mAudioHwDevs.size(); 234 235 while (!mRecordThreads.isEmpty()) { 236 // closeInput() will remove first entry from mRecordThreads 237 closeInput(mRecordThreads.keyAt(0)); 238 } 239 while (!mPlaybackThreads.isEmpty()) { 240 // closeOutput() will remove first entry from mPlaybackThreads 241 closeOutput(mPlaybackThreads.keyAt(0)); 242 } 243 244 for (int i = 0; i < num_devs; i++) { 245 audio_hw_device_t *dev = mAudioHwDevs[i]; 246 audio_hw_device_close(dev); 247 } 248} 249 250audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 251{ 252 /* first matching HW device is returned */ 253 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 254 audio_hw_device_t *dev = mAudioHwDevs[i]; 255 if ((dev->get_supported_devices(dev) & devices) == devices) 256 return dev; 257 } 258 return NULL; 259} 260 261status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 262{ 263 const size_t SIZE = 256; 264 char buffer[SIZE]; 265 String8 result; 266 267 result.append("Clients:\n"); 268 for (size_t i = 0; i < mClients.size(); ++i) { 269 sp<Client> client = mClients.valueAt(i).promote(); 270 if (client != 0) { 271 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 272 result.append(buffer); 273 } 274 } 275 276 result.append("Global session refs:\n"); 277 result.append(" session pid cnt\n"); 278 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 279 AudioSessionRef *r = mAudioSessionRefs[i]; 280 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 281 result.append(buffer); 282 } 283 write(fd, result.string(), result.size()); 284 return NO_ERROR; 285} 286 287 288status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 289{ 290 const size_t SIZE = 256; 291 char buffer[SIZE]; 292 String8 result; 293 hardware_call_state hardwareStatus = mHardwareStatus; 294 295 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 296 result.append(buffer); 297 write(fd, result.string(), result.size()); 298 return NO_ERROR; 299} 300 301status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 302{ 303 const size_t SIZE = 256; 304 char buffer[SIZE]; 305 String8 result; 306 snprintf(buffer, SIZE, "Permission Denial: " 307 "can't dump AudioFlinger from pid=%d, uid=%d\n", 308 IPCThreadState::self()->getCallingPid(), 309 IPCThreadState::self()->getCallingUid()); 310 result.append(buffer); 311 write(fd, result.string(), result.size()); 312 return NO_ERROR; 313} 314 315static bool tryLock(Mutex& mutex) 316{ 317 bool locked = false; 318 for (int i = 0; i < kDumpLockRetries; ++i) { 319 if (mutex.tryLock() == NO_ERROR) { 320 locked = true; 321 break; 322 } 323 usleep(kDumpLockSleepUs); 324 } 325 return locked; 326} 327 328status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 329{ 330 if (!checkCallingPermission(String16("android.permission.DUMP"))) { 331 dumpPermissionDenial(fd, args); 332 } else { 333 // get state of hardware lock 334 bool hardwareLocked = tryLock(mHardwareLock); 335 if (!hardwareLocked) { 336 String8 result(kHardwareLockedString); 337 write(fd, result.string(), result.size()); 338 } else { 339 mHardwareLock.unlock(); 340 } 341 342 bool locked = tryLock(mLock); 343 344 // failed to lock - AudioFlinger is probably deadlocked 345 if (!locked) { 346 String8 result(kDeadlockedString); 347 write(fd, result.string(), result.size()); 348 } 349 350 dumpClients(fd, args); 351 dumpInternals(fd, args); 352 353 // dump playback threads 354 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 355 mPlaybackThreads.valueAt(i)->dump(fd, args); 356 } 357 358 // dump record threads 359 for (size_t i = 0; i < mRecordThreads.size(); i++) { 360 mRecordThreads.valueAt(i)->dump(fd, args); 361 } 362 363 // dump all hardware devs 364 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 365 audio_hw_device_t *dev = mAudioHwDevs[i]; 366 dev->dump(dev, fd); 367 } 368 if (locked) mLock.unlock(); 369 } 370 return NO_ERROR; 371} 372 373 374// IAudioFlinger interface 375 376 377sp<IAudioTrack> AudioFlinger::createTrack( 378 pid_t pid, 379 audio_stream_type_t streamType, 380 uint32_t sampleRate, 381 audio_format_t format, 382 uint32_t channelMask, 383 int frameCount, 384 uint32_t flags, 385 const sp<IMemory>& sharedBuffer, 386 int output, 387 int *sessionId, 388 status_t *status) 389{ 390 sp<PlaybackThread::Track> track; 391 sp<TrackHandle> trackHandle; 392 sp<Client> client; 393 wp<Client> wclient; 394 status_t lStatus; 395 int lSessionId; 396 397 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 398 // but if someone uses binder directly they could bypass that and cause us to crash 399 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 400 ALOGE("createTrack() invalid stream type %d", streamType); 401 lStatus = BAD_VALUE; 402 goto Exit; 403 } 404 405 { 406 Mutex::Autolock _l(mLock); 407 PlaybackThread *thread = checkPlaybackThread_l(output); 408 PlaybackThread *effectThread = NULL; 409 if (thread == NULL) { 410 ALOGE("unknown output thread"); 411 lStatus = BAD_VALUE; 412 goto Exit; 413 } 414 415 wclient = mClients.valueFor(pid); 416 417 if (wclient != NULL) { 418 client = wclient.promote(); 419 } else { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 425 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 426 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 427 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 428 if (mPlaybackThreads.keyAt(i) != output) { 429 // prevent same audio session on different output threads 430 uint32_t sessions = t->hasAudioSession(*sessionId); 431 if (sessions & PlaybackThread::TRACK_SESSION) { 432 ALOGE("createTrack() session ID %d already in use", *sessionId); 433 lStatus = BAD_VALUE; 434 goto Exit; 435 } 436 // check if an effect with same session ID is waiting for a track to be created 437 if (sessions & PlaybackThread::EFFECT_SESSION) { 438 effectThread = t.get(); 439 } 440 } 441 } 442 lSessionId = *sessionId; 443 } else { 444 // if no audio session id is provided, create one here 445 lSessionId = nextUniqueId(); 446 if (sessionId != NULL) { 447 *sessionId = lSessionId; 448 } 449 } 450 ALOGV("createTrack() lSessionId: %d", lSessionId); 451 452 track = thread->createTrack_l(client, streamType, sampleRate, format, 453 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 454 455 // move effect chain to this output thread if an effect on same session was waiting 456 // for a track to be created 457 if (lStatus == NO_ERROR && effectThread != NULL) { 458 Mutex::Autolock _dl(thread->mLock); 459 Mutex::Autolock _sl(effectThread->mLock); 460 moveEffectChain_l(lSessionId, effectThread, thread, true); 461 } 462 } 463 if (lStatus == NO_ERROR) { 464 trackHandle = new TrackHandle(track); 465 } else { 466 // remove local strong reference to Client before deleting the Track so that the Client 467 // destructor is called by the TrackBase destructor with mLock held 468 client.clear(); 469 track.clear(); 470 } 471 472Exit: 473 if(status) { 474 *status = lStatus; 475 } 476 return trackHandle; 477} 478 479uint32_t AudioFlinger::sampleRate(int output) const 480{ 481 Mutex::Autolock _l(mLock); 482 PlaybackThread *thread = checkPlaybackThread_l(output); 483 if (thread == NULL) { 484 ALOGW("sampleRate() unknown thread %d", output); 485 return 0; 486 } 487 return thread->sampleRate(); 488} 489 490int AudioFlinger::channelCount(int output) const 491{ 492 Mutex::Autolock _l(mLock); 493 PlaybackThread *thread = checkPlaybackThread_l(output); 494 if (thread == NULL) { 495 ALOGW("channelCount() unknown thread %d", output); 496 return 0; 497 } 498 return thread->channelCount(); 499} 500 501audio_format_t AudioFlinger::format(int output) const 502{ 503 Mutex::Autolock _l(mLock); 504 PlaybackThread *thread = checkPlaybackThread_l(output); 505 if (thread == NULL) { 506 ALOGW("format() unknown thread %d", output); 507 return AUDIO_FORMAT_INVALID; 508 } 509 return thread->format(); 510} 511 512size_t AudioFlinger::frameCount(int output) const 513{ 514 Mutex::Autolock _l(mLock); 515 PlaybackThread *thread = checkPlaybackThread_l(output); 516 if (thread == NULL) { 517 ALOGW("frameCount() unknown thread %d", output); 518 return 0; 519 } 520 return thread->frameCount(); 521} 522 523uint32_t AudioFlinger::latency(int output) const 524{ 525 Mutex::Autolock _l(mLock); 526 PlaybackThread *thread = checkPlaybackThread_l(output); 527 if (thread == NULL) { 528 ALOGW("latency() unknown thread %d", output); 529 return 0; 530 } 531 return thread->latency(); 532} 533 534status_t AudioFlinger::setMasterVolume(float value) 535{ 536 status_t ret = initCheck(); 537 if (ret != NO_ERROR) { 538 return ret; 539 } 540 541 // check calling permissions 542 if (!settingsAllowed()) { 543 return PERMISSION_DENIED; 544 } 545 546 // when hw supports master volume, don't scale in sw mixer 547 { // scope for the lock 548 AutoMutex lock(mHardwareLock); 549 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 550 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 551 value = 1.0f; 552 } 553 mHardwareStatus = AUDIO_HW_IDLE; 554 } 555 556 Mutex::Autolock _l(mLock); 557 mMasterVolume = value; 558 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 559 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 560 561 return NO_ERROR; 562} 563 564status_t AudioFlinger::setMode(audio_mode_t mode) 565{ 566 status_t ret = initCheck(); 567 if (ret != NO_ERROR) { 568 return ret; 569 } 570 571 // check calling permissions 572 if (!settingsAllowed()) { 573 return PERMISSION_DENIED; 574 } 575 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 576 ALOGW("Illegal value: setMode(%d)", mode); 577 return BAD_VALUE; 578 } 579 580 { // scope for the lock 581 AutoMutex lock(mHardwareLock); 582 mHardwareStatus = AUDIO_HW_SET_MODE; 583 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 584 mHardwareStatus = AUDIO_HW_IDLE; 585 } 586 587 if (NO_ERROR == ret) { 588 Mutex::Autolock _l(mLock); 589 mMode = mode; 590 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 591 mPlaybackThreads.valueAt(i)->setMode(mode); 592 } 593 594 return ret; 595} 596 597status_t AudioFlinger::setMicMute(bool state) 598{ 599 status_t ret = initCheck(); 600 if (ret != NO_ERROR) { 601 return ret; 602 } 603 604 // check calling permissions 605 if (!settingsAllowed()) { 606 return PERMISSION_DENIED; 607 } 608 609 AutoMutex lock(mHardwareLock); 610 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 611 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 612 mHardwareStatus = AUDIO_HW_IDLE; 613 return ret; 614} 615 616bool AudioFlinger::getMicMute() const 617{ 618 status_t ret = initCheck(); 619 if (ret != NO_ERROR) { 620 return false; 621 } 622 623 bool state = AUDIO_MODE_INVALID; 624 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 625 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 626 mHardwareStatus = AUDIO_HW_IDLE; 627 return state; 628} 629 630status_t AudioFlinger::setMasterMute(bool muted) 631{ 632 // check calling permissions 633 if (!settingsAllowed()) { 634 return PERMISSION_DENIED; 635 } 636 637 Mutex::Autolock _l(mLock); 638 mMasterMute = muted; 639 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 640 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 641 642 return NO_ERROR; 643} 644 645float AudioFlinger::masterVolume() const 646{ 647 Mutex::Autolock _l(mLock); 648 return masterVolume_l(); 649} 650 651bool AudioFlinger::masterMute() const 652{ 653 Mutex::Autolock _l(mLock); 654 return masterMute_l(); 655} 656 657status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output) 658{ 659 // check calling permissions 660 if (!settingsAllowed()) { 661 return PERMISSION_DENIED; 662 } 663 664 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 665 ALOGE("setStreamVolume() invalid stream %d", stream); 666 return BAD_VALUE; 667 } 668 669 AutoMutex lock(mLock); 670 PlaybackThread *thread = NULL; 671 if (output) { 672 thread = checkPlaybackThread_l(output); 673 if (thread == NULL) { 674 return BAD_VALUE; 675 } 676 } 677 678 mStreamTypes[stream].volume = value; 679 680 if (thread == NULL) { 681 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 682 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 683 } 684 } else { 685 thread->setStreamVolume(stream, value); 686 } 687 688 return NO_ERROR; 689} 690 691status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 692{ 693 // check calling permissions 694 if (!settingsAllowed()) { 695 return PERMISSION_DENIED; 696 } 697 698 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 699 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 700 ALOGE("setStreamMute() invalid stream %d", stream); 701 return BAD_VALUE; 702 } 703 704 AutoMutex lock(mLock); 705 mStreamTypes[stream].mute = muted; 706 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 707 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 708 709 return NO_ERROR; 710} 711 712float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const 713{ 714 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 715 return 0.0f; 716 } 717 718 AutoMutex lock(mLock); 719 float volume; 720 if (output) { 721 PlaybackThread *thread = checkPlaybackThread_l(output); 722 if (thread == NULL) { 723 return 0.0f; 724 } 725 volume = thread->streamVolume(stream); 726 } else { 727 volume = mStreamTypes[stream].volume; 728 } 729 730 return volume; 731} 732 733bool AudioFlinger::streamMute(audio_stream_type_t stream) const 734{ 735 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 736 return true; 737 } 738 739 return mStreamTypes[stream].mute; 740} 741 742status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 743{ 744 status_t result; 745 746 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 747 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 748 // check calling permissions 749 if (!settingsAllowed()) { 750 return PERMISSION_DENIED; 751 } 752 753 // ioHandle == 0 means the parameters are global to the audio hardware interface 754 if (ioHandle == 0) { 755 AutoMutex lock(mHardwareLock); 756 mHardwareStatus = AUDIO_SET_PARAMETER; 757 status_t final_result = NO_ERROR; 758 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 759 audio_hw_device_t *dev = mAudioHwDevs[i]; 760 result = dev->set_parameters(dev, keyValuePairs.string()); 761 final_result = result ?: final_result; 762 } 763 mHardwareStatus = AUDIO_HW_IDLE; 764 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 765 AudioParameter param = AudioParameter(keyValuePairs); 766 String8 value; 767 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 768 Mutex::Autolock _l(mLock); 769 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 770 if (mBtNrecIsOff != btNrecIsOff) { 771 for (size_t i = 0; i < mRecordThreads.size(); i++) { 772 sp<RecordThread> thread = mRecordThreads.valueAt(i); 773 RecordThread::RecordTrack *track = thread->track(); 774 if (track != NULL) { 775 audio_devices_t device = (audio_devices_t)( 776 thread->device() & AUDIO_DEVICE_IN_ALL); 777 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 778 thread->setEffectSuspended(FX_IID_AEC, 779 suspend, 780 track->sessionId()); 781 thread->setEffectSuspended(FX_IID_NS, 782 suspend, 783 track->sessionId()); 784 } 785 } 786 mBtNrecIsOff = btNrecIsOff; 787 } 788 } 789 return final_result; 790 } 791 792 // hold a strong ref on thread in case closeOutput() or closeInput() is called 793 // and the thread is exited once the lock is released 794 sp<ThreadBase> thread; 795 { 796 Mutex::Autolock _l(mLock); 797 thread = checkPlaybackThread_l(ioHandle); 798 if (thread == NULL) { 799 thread = checkRecordThread_l(ioHandle); 800 } else if (thread == primaryPlaybackThread_l()) { 801 // indicate output device change to all input threads for pre processing 802 AudioParameter param = AudioParameter(keyValuePairs); 803 int value; 804 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 805 for (size_t i = 0; i < mRecordThreads.size(); i++) { 806 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 807 } 808 } 809 } 810 } 811 if (thread != 0) { 812 return thread->setParameters(keyValuePairs); 813 } 814 return BAD_VALUE; 815} 816 817String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) const 818{ 819// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 820// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 821 822 if (ioHandle == 0) { 823 String8 out_s8; 824 825 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 826 audio_hw_device_t *dev = mAudioHwDevs[i]; 827 char *s = dev->get_parameters(dev, keys.string()); 828 out_s8 += String8(s); 829 free(s); 830 } 831 return out_s8; 832 } 833 834 Mutex::Autolock _l(mLock); 835 836 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 837 if (playbackThread != NULL) { 838 return playbackThread->getParameters(keys); 839 } 840 RecordThread *recordThread = checkRecordThread_l(ioHandle); 841 if (recordThread != NULL) { 842 return recordThread->getParameters(keys); 843 } 844 return String8(""); 845} 846 847size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 848{ 849 status_t ret = initCheck(); 850 if (ret != NO_ERROR) { 851 return 0; 852 } 853 854 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 855} 856 857unsigned int AudioFlinger::getInputFramesLost(int ioHandle) const 858{ 859 if (ioHandle == 0) { 860 return 0; 861 } 862 863 Mutex::Autolock _l(mLock); 864 865 RecordThread *recordThread = checkRecordThread_l(ioHandle); 866 if (recordThread != NULL) { 867 return recordThread->getInputFramesLost(); 868 } 869 return 0; 870} 871 872status_t AudioFlinger::setVoiceVolume(float value) 873{ 874 status_t ret = initCheck(); 875 if (ret != NO_ERROR) { 876 return ret; 877 } 878 879 // check calling permissions 880 if (!settingsAllowed()) { 881 return PERMISSION_DENIED; 882 } 883 884 AutoMutex lock(mHardwareLock); 885 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 886 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 887 mHardwareStatus = AUDIO_HW_IDLE; 888 889 return ret; 890} 891 892status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) const 893{ 894 status_t status; 895 896 Mutex::Autolock _l(mLock); 897 898 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 899 if (playbackThread != NULL) { 900 return playbackThread->getRenderPosition(halFrames, dspFrames); 901 } 902 903 return BAD_VALUE; 904} 905 906void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 907{ 908 909 Mutex::Autolock _l(mLock); 910 911 pid_t pid = IPCThreadState::self()->getCallingPid(); 912 if (mNotificationClients.indexOfKey(pid) < 0) { 913 sp<NotificationClient> notificationClient = new NotificationClient(this, 914 client, 915 pid); 916 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 917 918 mNotificationClients.add(pid, notificationClient); 919 920 sp<IBinder> binder = client->asBinder(); 921 binder->linkToDeath(notificationClient); 922 923 // the config change is always sent from playback or record threads to avoid deadlock 924 // with AudioSystem::gLock 925 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 926 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 927 } 928 929 for (size_t i = 0; i < mRecordThreads.size(); i++) { 930 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 931 } 932 } 933} 934 935void AudioFlinger::removeNotificationClient(pid_t pid) 936{ 937 Mutex::Autolock _l(mLock); 938 939 int index = mNotificationClients.indexOfKey(pid); 940 if (index >= 0) { 941 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 942 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 943 mNotificationClients.removeItem(pid); 944 } 945 946 ALOGV("%d died, releasing its sessions", pid); 947 int num = mAudioSessionRefs.size(); 948 bool removed = false; 949 for (int i = 0; i< num; i++) { 950 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 951 ALOGV(" pid %d @ %d", ref->pid, i); 952 if (ref->pid == pid) { 953 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 954 mAudioSessionRefs.removeAt(i); 955 delete ref; 956 removed = true; 957 i--; 958 num--; 959 } 960 } 961 if (removed) { 962 purgeStaleEffects_l(); 963 } 964} 965 966// audioConfigChanged_l() must be called with AudioFlinger::mLock held 967void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 968{ 969 size_t size = mNotificationClients.size(); 970 for (size_t i = 0; i < size; i++) { 971 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 972 param2); 973 } 974} 975 976// removeClient_l() must be called with AudioFlinger::mLock held 977void AudioFlinger::removeClient_l(pid_t pid) 978{ 979 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 980 mClients.removeItem(pid); 981} 982 983 984// ---------------------------------------------------------------------------- 985 986AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device, 987 type_t type) 988 : Thread(false), 989 mType(type), 990 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 991 // mChannelMask 992 mChannelCount(0), 993 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 994 mParamStatus(NO_ERROR), 995 mStandby(false), mId(id), mExiting(false), 996 mDevice(device), 997 mDeathRecipient(new PMDeathRecipient(this)) 998{ 999} 1000 1001AudioFlinger::ThreadBase::~ThreadBase() 1002{ 1003 mParamCond.broadcast(); 1004 // do not lock the mutex in destructor 1005 releaseWakeLock_l(); 1006 if (mPowerManager != 0) { 1007 sp<IBinder> binder = mPowerManager->asBinder(); 1008 binder->unlinkToDeath(mDeathRecipient); 1009 } 1010} 1011 1012void AudioFlinger::ThreadBase::exit() 1013{ 1014 // keep a strong ref on ourself so that we won't get 1015 // destroyed in the middle of requestExitAndWait() 1016 sp <ThreadBase> strongMe = this; 1017 1018 ALOGV("ThreadBase::exit"); 1019 { 1020 AutoMutex lock(mLock); 1021 mExiting = true; 1022 requestExit(); 1023 mWaitWorkCV.signal(); 1024 } 1025 requestExitAndWait(); 1026} 1027 1028status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1029{ 1030 status_t status; 1031 1032 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1033 Mutex::Autolock _l(mLock); 1034 1035 mNewParameters.add(keyValuePairs); 1036 mWaitWorkCV.signal(); 1037 // wait condition with timeout in case the thread loop has exited 1038 // before the request could be processed 1039 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1040 status = mParamStatus; 1041 mWaitWorkCV.signal(); 1042 } else { 1043 status = TIMED_OUT; 1044 } 1045 return status; 1046} 1047 1048void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1049{ 1050 Mutex::Autolock _l(mLock); 1051 sendConfigEvent_l(event, param); 1052} 1053 1054// sendConfigEvent_l() must be called with ThreadBase::mLock held 1055void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1056{ 1057 ConfigEvent configEvent; 1058 configEvent.mEvent = event; 1059 configEvent.mParam = param; 1060 mConfigEvents.add(configEvent); 1061 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1062 mWaitWorkCV.signal(); 1063} 1064 1065void AudioFlinger::ThreadBase::processConfigEvents() 1066{ 1067 mLock.lock(); 1068 while(!mConfigEvents.isEmpty()) { 1069 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1070 ConfigEvent configEvent = mConfigEvents[0]; 1071 mConfigEvents.removeAt(0); 1072 // release mLock before locking AudioFlinger mLock: lock order is always 1073 // AudioFlinger then ThreadBase to avoid cross deadlock 1074 mLock.unlock(); 1075 mAudioFlinger->mLock.lock(); 1076 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1077 mAudioFlinger->mLock.unlock(); 1078 mLock.lock(); 1079 } 1080 mLock.unlock(); 1081} 1082 1083status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1084{ 1085 const size_t SIZE = 256; 1086 char buffer[SIZE]; 1087 String8 result; 1088 1089 bool locked = tryLock(mLock); 1090 if (!locked) { 1091 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1092 write(fd, buffer, strlen(buffer)); 1093 } 1094 1095 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1096 result.append(buffer); 1097 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1098 result.append(buffer); 1099 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1100 result.append(buffer); 1101 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1102 result.append(buffer); 1103 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1104 result.append(buffer); 1105 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1106 result.append(buffer); 1107 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1108 result.append(buffer); 1109 1110 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1111 result.append(buffer); 1112 result.append(" Index Command"); 1113 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1114 snprintf(buffer, SIZE, "\n %02d ", i); 1115 result.append(buffer); 1116 result.append(mNewParameters[i]); 1117 } 1118 1119 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, " Index event param\n"); 1122 result.append(buffer); 1123 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1124 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1125 result.append(buffer); 1126 } 1127 result.append("\n"); 1128 1129 write(fd, result.string(), result.size()); 1130 1131 if (locked) { 1132 mLock.unlock(); 1133 } 1134 return NO_ERROR; 1135} 1136 1137status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1138{ 1139 const size_t SIZE = 256; 1140 char buffer[SIZE]; 1141 String8 result; 1142 1143 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1144 write(fd, buffer, strlen(buffer)); 1145 1146 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1147 sp<EffectChain> chain = mEffectChains[i]; 1148 if (chain != 0) { 1149 chain->dump(fd, args); 1150 } 1151 } 1152 return NO_ERROR; 1153} 1154 1155void AudioFlinger::ThreadBase::acquireWakeLock() 1156{ 1157 Mutex::Autolock _l(mLock); 1158 acquireWakeLock_l(); 1159} 1160 1161void AudioFlinger::ThreadBase::acquireWakeLock_l() 1162{ 1163 if (mPowerManager == 0) { 1164 // use checkService() to avoid blocking if power service is not up yet 1165 sp<IBinder> binder = 1166 defaultServiceManager()->checkService(String16("power")); 1167 if (binder == 0) { 1168 ALOGW("Thread %s cannot connect to the power manager service", mName); 1169 } else { 1170 mPowerManager = interface_cast<IPowerManager>(binder); 1171 binder->linkToDeath(mDeathRecipient); 1172 } 1173 } 1174 if (mPowerManager != 0) { 1175 sp<IBinder> binder = new BBinder(); 1176 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1177 binder, 1178 String16(mName)); 1179 if (status == NO_ERROR) { 1180 mWakeLockToken = binder; 1181 } 1182 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1183 } 1184} 1185 1186void AudioFlinger::ThreadBase::releaseWakeLock() 1187{ 1188 Mutex::Autolock _l(mLock); 1189 releaseWakeLock_l(); 1190} 1191 1192void AudioFlinger::ThreadBase::releaseWakeLock_l() 1193{ 1194 if (mWakeLockToken != 0) { 1195 ALOGV("releaseWakeLock_l() %s", mName); 1196 if (mPowerManager != 0) { 1197 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1198 } 1199 mWakeLockToken.clear(); 1200 } 1201} 1202 1203void AudioFlinger::ThreadBase::clearPowerManager() 1204{ 1205 Mutex::Autolock _l(mLock); 1206 releaseWakeLock_l(); 1207 mPowerManager.clear(); 1208} 1209 1210void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1211{ 1212 sp<ThreadBase> thread = mThread.promote(); 1213 if (thread != 0) { 1214 thread->clearPowerManager(); 1215 } 1216 ALOGW("power manager service died !!!"); 1217} 1218 1219void AudioFlinger::ThreadBase::setEffectSuspended( 1220 const effect_uuid_t *type, bool suspend, int sessionId) 1221{ 1222 Mutex::Autolock _l(mLock); 1223 setEffectSuspended_l(type, suspend, sessionId); 1224} 1225 1226void AudioFlinger::ThreadBase::setEffectSuspended_l( 1227 const effect_uuid_t *type, bool suspend, int sessionId) 1228{ 1229 sp<EffectChain> chain = getEffectChain_l(sessionId); 1230 if (chain != 0) { 1231 if (type != NULL) { 1232 chain->setEffectSuspended_l(type, suspend); 1233 } else { 1234 chain->setEffectSuspendedAll_l(suspend); 1235 } 1236 } 1237 1238 updateSuspendedSessions_l(type, suspend, sessionId); 1239} 1240 1241void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1242{ 1243 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1244 if (index < 0) { 1245 return; 1246 } 1247 1248 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1249 mSuspendedSessions.editValueAt(index); 1250 1251 for (size_t i = 0; i < sessionEffects.size(); i++) { 1252 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1253 for (int j = 0; j < desc->mRefCount; j++) { 1254 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1255 chain->setEffectSuspendedAll_l(true); 1256 } else { 1257 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1258 desc->mType.timeLow); 1259 chain->setEffectSuspended_l(&desc->mType, true); 1260 } 1261 } 1262 } 1263} 1264 1265void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1266 bool suspend, 1267 int sessionId) 1268{ 1269 int index = mSuspendedSessions.indexOfKey(sessionId); 1270 1271 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1272 1273 if (suspend) { 1274 if (index >= 0) { 1275 sessionEffects = mSuspendedSessions.editValueAt(index); 1276 } else { 1277 mSuspendedSessions.add(sessionId, sessionEffects); 1278 } 1279 } else { 1280 if (index < 0) { 1281 return; 1282 } 1283 sessionEffects = mSuspendedSessions.editValueAt(index); 1284 } 1285 1286 1287 int key = EffectChain::kKeyForSuspendAll; 1288 if (type != NULL) { 1289 key = type->timeLow; 1290 } 1291 index = sessionEffects.indexOfKey(key); 1292 1293 sp <SuspendedSessionDesc> desc; 1294 if (suspend) { 1295 if (index >= 0) { 1296 desc = sessionEffects.valueAt(index); 1297 } else { 1298 desc = new SuspendedSessionDesc(); 1299 if (type != NULL) { 1300 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1301 } 1302 sessionEffects.add(key, desc); 1303 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1304 } 1305 desc->mRefCount++; 1306 } else { 1307 if (index < 0) { 1308 return; 1309 } 1310 desc = sessionEffects.valueAt(index); 1311 if (--desc->mRefCount == 0) { 1312 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1313 sessionEffects.removeItemsAt(index); 1314 if (sessionEffects.isEmpty()) { 1315 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1316 sessionId); 1317 mSuspendedSessions.removeItem(sessionId); 1318 } 1319 } 1320 } 1321 if (!sessionEffects.isEmpty()) { 1322 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1323 } 1324} 1325 1326void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1327 bool enabled, 1328 int sessionId) 1329{ 1330 Mutex::Autolock _l(mLock); 1331 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1332} 1333 1334void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1335 bool enabled, 1336 int sessionId) 1337{ 1338 if (mType != RECORD) { 1339 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1340 // another session. This gives the priority to well behaved effect control panels 1341 // and applications not using global effects. 1342 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1343 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1344 } 1345 } 1346 1347 sp<EffectChain> chain = getEffectChain_l(sessionId); 1348 if (chain != 0) { 1349 chain->checkSuspendOnEffectEnabled(effect, enabled); 1350 } 1351} 1352 1353// ---------------------------------------------------------------------------- 1354 1355AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1356 AudioStreamOut* output, 1357 int id, 1358 uint32_t device, 1359 type_t type) 1360 : ThreadBase(audioFlinger, id, device, type), 1361 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1362 // Assumes constructor is called by AudioFlinger with it's mLock held, 1363 // but it would be safer to explicitly pass initial masterMute as parameter 1364 mMasterMute(audioFlinger->masterMute_l()), 1365 // mStreamTypes[] initialized in constructor body 1366 mOutput(output), 1367 // Assumes constructor is called by AudioFlinger with it's mLock held, 1368 // but it would be safer to explicitly pass initial masterVolume as parameter 1369 mMasterVolume(audioFlinger->masterVolume_l()), 1370 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1371{ 1372 snprintf(mName, kNameLength, "AudioOut_%d", id); 1373 1374 readOutputParameters(); 1375 1376 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1377 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1378 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1379 stream = (audio_stream_type_t) (stream + 1)) { 1380 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1381 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1382 // initialized by stream_type_t default constructor 1383 // mStreamTypes[stream].valid = true; 1384 } 1385} 1386 1387AudioFlinger::PlaybackThread::~PlaybackThread() 1388{ 1389 delete [] mMixBuffer; 1390} 1391 1392status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1393{ 1394 dumpInternals(fd, args); 1395 dumpTracks(fd, args); 1396 dumpEffectChains(fd, args); 1397 return NO_ERROR; 1398} 1399 1400status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1401{ 1402 const size_t SIZE = 256; 1403 char buffer[SIZE]; 1404 String8 result; 1405 1406 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1407 result.append(buffer); 1408 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1409 for (size_t i = 0; i < mTracks.size(); ++i) { 1410 sp<Track> track = mTracks[i]; 1411 if (track != 0) { 1412 track->dump(buffer, SIZE); 1413 result.append(buffer); 1414 } 1415 } 1416 1417 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1418 result.append(buffer); 1419 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1420 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1421 sp<Track> track = mActiveTracks[i].promote(); 1422 if (track != 0) { 1423 track->dump(buffer, SIZE); 1424 result.append(buffer); 1425 } 1426 } 1427 write(fd, result.string(), result.size()); 1428 return NO_ERROR; 1429} 1430 1431status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1432{ 1433 const size_t SIZE = 256; 1434 char buffer[SIZE]; 1435 String8 result; 1436 1437 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1438 result.append(buffer); 1439 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1440 result.append(buffer); 1441 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1442 result.append(buffer); 1443 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1444 result.append(buffer); 1445 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1446 result.append(buffer); 1447 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1448 result.append(buffer); 1449 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1450 result.append(buffer); 1451 write(fd, result.string(), result.size()); 1452 1453 dumpBase(fd, args); 1454 1455 return NO_ERROR; 1456} 1457 1458// Thread virtuals 1459status_t AudioFlinger::PlaybackThread::readyToRun() 1460{ 1461 status_t status = initCheck(); 1462 if (status == NO_ERROR) { 1463 ALOGI("AudioFlinger's thread %p ready to run", this); 1464 } else { 1465 ALOGE("No working audio driver found."); 1466 } 1467 return status; 1468} 1469 1470void AudioFlinger::PlaybackThread::onFirstRef() 1471{ 1472 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1473} 1474 1475// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1476sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1477 const sp<AudioFlinger::Client>& client, 1478 audio_stream_type_t streamType, 1479 uint32_t sampleRate, 1480 audio_format_t format, 1481 uint32_t channelMask, 1482 int frameCount, 1483 const sp<IMemory>& sharedBuffer, 1484 int sessionId, 1485 status_t *status) 1486{ 1487 sp<Track> track; 1488 status_t lStatus; 1489 1490 if (mType == DIRECT) { 1491 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1492 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1493 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1494 "for output %p with format %d", 1495 sampleRate, format, channelMask, mOutput, mFormat); 1496 lStatus = BAD_VALUE; 1497 goto Exit; 1498 } 1499 } 1500 } else { 1501 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1502 if (sampleRate > mSampleRate*2) { 1503 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1504 lStatus = BAD_VALUE; 1505 goto Exit; 1506 } 1507 } 1508 1509 lStatus = initCheck(); 1510 if (lStatus != NO_ERROR) { 1511 ALOGE("Audio driver not initialized."); 1512 goto Exit; 1513 } 1514 1515 { // scope for mLock 1516 Mutex::Autolock _l(mLock); 1517 1518 // all tracks in same audio session must share the same routing strategy otherwise 1519 // conflicts will happen when tracks are moved from one output to another by audio policy 1520 // manager 1521 uint32_t strategy = 1522 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1523 for (size_t i = 0; i < mTracks.size(); ++i) { 1524 sp<Track> t = mTracks[i]; 1525 if (t != 0) { 1526 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1527 if (sessionId == t->sessionId() && strategy != actual) { 1528 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1529 strategy, actual); 1530 lStatus = BAD_VALUE; 1531 goto Exit; 1532 } 1533 } 1534 } 1535 1536 track = new Track(this, client, streamType, sampleRate, format, 1537 channelMask, frameCount, sharedBuffer, sessionId); 1538 if (track->getCblk() == NULL || track->name() < 0) { 1539 lStatus = NO_MEMORY; 1540 goto Exit; 1541 } 1542 mTracks.add(track); 1543 1544 sp<EffectChain> chain = getEffectChain_l(sessionId); 1545 if (chain != 0) { 1546 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1547 track->setMainBuffer(chain->inBuffer()); 1548 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1549 chain->incTrackCnt(); 1550 } 1551 1552 // invalidate track immediately if the stream type was moved to another thread since 1553 // createTrack() was called by the client process. 1554 if (!mStreamTypes[streamType].valid) { 1555 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1556 this, streamType); 1557 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1558 } 1559 } 1560 lStatus = NO_ERROR; 1561 1562Exit: 1563 if(status) { 1564 *status = lStatus; 1565 } 1566 return track; 1567} 1568 1569uint32_t AudioFlinger::PlaybackThread::latency() const 1570{ 1571 Mutex::Autolock _l(mLock); 1572 if (initCheck() == NO_ERROR) { 1573 return mOutput->stream->get_latency(mOutput->stream); 1574 } else { 1575 return 0; 1576 } 1577} 1578 1579status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1580{ 1581 mMasterVolume = value; 1582 return NO_ERROR; 1583} 1584 1585status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1586{ 1587 mMasterMute = muted; 1588 return NO_ERROR; 1589} 1590 1591status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1592{ 1593 mStreamTypes[stream].volume = value; 1594 return NO_ERROR; 1595} 1596 1597status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1598{ 1599 mStreamTypes[stream].mute = muted; 1600 return NO_ERROR; 1601} 1602 1603float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1604{ 1605 return mStreamTypes[stream].volume; 1606} 1607 1608bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1609{ 1610 return mStreamTypes[stream].mute; 1611} 1612 1613// addTrack_l() must be called with ThreadBase::mLock held 1614status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1615{ 1616 status_t status = ALREADY_EXISTS; 1617 1618 // set retry count for buffer fill 1619 track->mRetryCount = kMaxTrackStartupRetries; 1620 if (mActiveTracks.indexOf(track) < 0) { 1621 // the track is newly added, make sure it fills up all its 1622 // buffers before playing. This is to ensure the client will 1623 // effectively get the latency it requested. 1624 track->mFillingUpStatus = Track::FS_FILLING; 1625 track->mResetDone = false; 1626 mActiveTracks.add(track); 1627 if (track->mainBuffer() != mMixBuffer) { 1628 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1629 if (chain != 0) { 1630 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1631 chain->incActiveTrackCnt(); 1632 } 1633 } 1634 1635 status = NO_ERROR; 1636 } 1637 1638 ALOGV("mWaitWorkCV.broadcast"); 1639 mWaitWorkCV.broadcast(); 1640 1641 return status; 1642} 1643 1644// destroyTrack_l() must be called with ThreadBase::mLock held 1645void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1646{ 1647 track->mState = TrackBase::TERMINATED; 1648 if (mActiveTracks.indexOf(track) < 0) { 1649 removeTrack_l(track); 1650 } 1651} 1652 1653void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1654{ 1655 mTracks.remove(track); 1656 deleteTrackName_l(track->name()); 1657 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1658 if (chain != 0) { 1659 chain->decTrackCnt(); 1660 } 1661} 1662 1663String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1664{ 1665 String8 out_s8 = String8(""); 1666 char *s; 1667 1668 Mutex::Autolock _l(mLock); 1669 if (initCheck() != NO_ERROR) { 1670 return out_s8; 1671 } 1672 1673 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1674 out_s8 = String8(s); 1675 free(s); 1676 return out_s8; 1677} 1678 1679// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1680void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1681 AudioSystem::OutputDescriptor desc; 1682 void *param2 = NULL; 1683 1684 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1685 1686 switch (event) { 1687 case AudioSystem::OUTPUT_OPENED: 1688 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1689 desc.channels = mChannelMask; 1690 desc.samplingRate = mSampleRate; 1691 desc.format = mFormat; 1692 desc.frameCount = mFrameCount; 1693 desc.latency = latency(); 1694 param2 = &desc; 1695 break; 1696 1697 case AudioSystem::STREAM_CONFIG_CHANGED: 1698 param2 = ¶m; 1699 case AudioSystem::OUTPUT_CLOSED: 1700 default: 1701 break; 1702 } 1703 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1704} 1705 1706void AudioFlinger::PlaybackThread::readOutputParameters() 1707{ 1708 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1709 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1710 mChannelCount = (uint16_t)popcount(mChannelMask); 1711 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1712 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1713 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1714 1715 // FIXME - Current mixer implementation only supports stereo output: Always 1716 // Allocate a stereo buffer even if HW output is mono. 1717 delete[] mMixBuffer; 1718 mMixBuffer = new int16_t[mFrameCount * 2]; 1719 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1720 1721 // force reconfiguration of effect chains and engines to take new buffer size and audio 1722 // parameters into account 1723 // Note that mLock is not held when readOutputParameters() is called from the constructor 1724 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1725 // matter. 1726 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1727 Vector< sp<EffectChain> > effectChains = mEffectChains; 1728 for (size_t i = 0; i < effectChains.size(); i ++) { 1729 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1730 } 1731} 1732 1733status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1734{ 1735 if (halFrames == NULL || dspFrames == NULL) { 1736 return BAD_VALUE; 1737 } 1738 Mutex::Autolock _l(mLock); 1739 if (initCheck() != NO_ERROR) { 1740 return INVALID_OPERATION; 1741 } 1742 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1743 1744 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1745} 1746 1747uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1748{ 1749 Mutex::Autolock _l(mLock); 1750 uint32_t result = 0; 1751 if (getEffectChain_l(sessionId) != 0) { 1752 result = EFFECT_SESSION; 1753 } 1754 1755 for (size_t i = 0; i < mTracks.size(); ++i) { 1756 sp<Track> track = mTracks[i]; 1757 if (sessionId == track->sessionId() && 1758 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1759 result |= TRACK_SESSION; 1760 break; 1761 } 1762 } 1763 1764 return result; 1765} 1766 1767uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1768{ 1769 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1770 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1771 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1772 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1773 } 1774 for (size_t i = 0; i < mTracks.size(); i++) { 1775 sp<Track> track = mTracks[i]; 1776 if (sessionId == track->sessionId() && 1777 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1778 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1779 } 1780 } 1781 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1782} 1783 1784 1785AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1786{ 1787 Mutex::Autolock _l(mLock); 1788 return mOutput; 1789} 1790 1791AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1792{ 1793 Mutex::Autolock _l(mLock); 1794 AudioStreamOut *output = mOutput; 1795 mOutput = NULL; 1796 return output; 1797} 1798 1799// this method must always be called either with ThreadBase mLock held or inside the thread loop 1800audio_stream_t* AudioFlinger::PlaybackThread::stream() 1801{ 1802 if (mOutput == NULL) { 1803 return NULL; 1804 } 1805 return &mOutput->stream->common; 1806} 1807 1808uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1809{ 1810 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1811 // decoding and transfer time. So sleeping for half of the latency would likely cause 1812 // underruns 1813 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1814 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1815 } else { 1816 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1817 } 1818} 1819 1820// ---------------------------------------------------------------------------- 1821 1822AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1823 int id, uint32_t device, type_t type) 1824 : PlaybackThread(audioFlinger, output, id, device, type), 1825 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1826 mPrevMixerStatus(MIXER_IDLE) 1827{ 1828 // FIXME - Current mixer implementation only supports stereo output 1829 if (mChannelCount == 1) { 1830 ALOGE("Invalid audio hardware channel count"); 1831 } 1832} 1833 1834AudioFlinger::MixerThread::~MixerThread() 1835{ 1836 delete mAudioMixer; 1837} 1838 1839bool AudioFlinger::MixerThread::threadLoop() 1840{ 1841 Vector< sp<Track> > tracksToRemove; 1842 mixer_state mixerStatus = MIXER_IDLE; 1843 nsecs_t standbyTime = systemTime(); 1844 size_t mixBufferSize = mFrameCount * mFrameSize; 1845 // FIXME: Relaxed timing because of a certain device that can't meet latency 1846 // Should be reduced to 2x after the vendor fixes the driver issue 1847 // increase threshold again due to low power audio mode. The way this warning threshold is 1848 // calculated and its usefulness should be reconsidered anyway. 1849 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1850 nsecs_t lastWarning = 0; 1851 bool longStandbyExit = false; 1852 uint32_t activeSleepTime = activeSleepTimeUs(); 1853 uint32_t idleSleepTime = idleSleepTimeUs(); 1854 uint32_t sleepTime = idleSleepTime; 1855 uint32_t sleepTimeShift = 0; 1856 Vector< sp<EffectChain> > effectChains; 1857#ifdef DEBUG_CPU_USAGE 1858 ThreadCpuUsage cpu; 1859 const CentralTendencyStatistics& stats = cpu.statistics(); 1860#endif 1861 1862 acquireWakeLock(); 1863 1864 while (!exitPending()) 1865 { 1866#ifdef DEBUG_CPU_USAGE 1867 cpu.sampleAndEnable(); 1868 unsigned n = stats.n(); 1869 // cpu.elapsed() is expensive, so don't call it every loop 1870 if ((n & 127) == 1) { 1871 long long elapsed = cpu.elapsed(); 1872 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1873 double perLoop = elapsed / (double) n; 1874 double perLoop100 = perLoop * 0.01; 1875 double mean = stats.mean(); 1876 double stddev = stats.stddev(); 1877 double minimum = stats.minimum(); 1878 double maximum = stats.maximum(); 1879 cpu.resetStatistics(); 1880 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", 1881 elapsed * .000000001, n, perLoop * .000001, 1882 mean * .001, 1883 stddev * .001, 1884 minimum * .001, 1885 maximum * .001, 1886 mean / perLoop100, 1887 stddev / perLoop100, 1888 minimum / perLoop100, 1889 maximum / perLoop100); 1890 } 1891 } 1892#endif 1893 processConfigEvents(); 1894 1895 mixerStatus = MIXER_IDLE; 1896 { // scope for mLock 1897 1898 Mutex::Autolock _l(mLock); 1899 1900 if (checkForNewParameters_l()) { 1901 mixBufferSize = mFrameCount * mFrameSize; 1902 // FIXME: Relaxed timing because of a certain device that can't meet latency 1903 // Should be reduced to 2x after the vendor fixes the driver issue 1904 // increase threshold again due to low power audio mode. The way this warning 1905 // threshold is calculated and its usefulness should be reconsidered anyway. 1906 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1907 activeSleepTime = activeSleepTimeUs(); 1908 idleSleepTime = idleSleepTimeUs(); 1909 } 1910 1911 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1912 1913 // put audio hardware into standby after short delay 1914 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1915 mSuspended)) { 1916 if (!mStandby) { 1917 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1918 mOutput->stream->common.standby(&mOutput->stream->common); 1919 mStandby = true; 1920 mBytesWritten = 0; 1921 } 1922 1923 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1924 // we're about to wait, flush the binder command buffer 1925 IPCThreadState::self()->flushCommands(); 1926 1927 if (exitPending()) break; 1928 1929 releaseWakeLock_l(); 1930 // wait until we have something to do... 1931 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1932 mWaitWorkCV.wait(mLock); 1933 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1934 acquireWakeLock_l(); 1935 1936 mPrevMixerStatus = MIXER_IDLE; 1937 if (!mMasterMute) { 1938 char value[PROPERTY_VALUE_MAX]; 1939 property_get("ro.audio.silent", value, "0"); 1940 if (atoi(value)) { 1941 ALOGD("Silence is golden"); 1942 setMasterMute(true); 1943 } 1944 } 1945 1946 standbyTime = systemTime() + kStandbyTimeInNsecs; 1947 sleepTime = idleSleepTime; 1948 sleepTimeShift = 0; 1949 continue; 1950 } 1951 } 1952 1953 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1954 1955 // prevent any changes in effect chain list and in each effect chain 1956 // during mixing and effect process as the audio buffers could be deleted 1957 // or modified if an effect is created or deleted 1958 lockEffectChains_l(effectChains); 1959 } 1960 1961 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1962 // mix buffers... 1963 mAudioMixer->process(); 1964 // increase sleep time progressively when application underrun condition clears. 1965 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1966 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1967 // such that we would underrun the audio HAL. 1968 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 1969 sleepTimeShift--; 1970 } 1971 sleepTime = 0; 1972 standbyTime = systemTime() + kStandbyTimeInNsecs; 1973 //TODO: delay standby when effects have a tail 1974 } else { 1975 // If no tracks are ready, sleep once for the duration of an output 1976 // buffer size, then write 0s to the output 1977 if (sleepTime == 0) { 1978 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1979 sleepTime = activeSleepTime >> sleepTimeShift; 1980 if (sleepTime < kMinThreadSleepTimeUs) { 1981 sleepTime = kMinThreadSleepTimeUs; 1982 } 1983 // reduce sleep time in case of consecutive application underruns to avoid 1984 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 1985 // duration we would end up writing less data than needed by the audio HAL if 1986 // the condition persists. 1987 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 1988 sleepTimeShift++; 1989 } 1990 } else { 1991 sleepTime = idleSleepTime; 1992 } 1993 } else if (mBytesWritten != 0 || 1994 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 1995 memset (mMixBuffer, 0, mixBufferSize); 1996 sleepTime = 0; 1997 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 1998 } 1999 // TODO add standby time extension fct of effect tail 2000 } 2001 2002 if (mSuspended) { 2003 sleepTime = suspendSleepTimeUs(); 2004 } 2005 // sleepTime == 0 means we must write to audio hardware 2006 if (sleepTime == 0) { 2007 for (size_t i = 0; i < effectChains.size(); i ++) { 2008 effectChains[i]->process_l(); 2009 } 2010 // enable changes in effect chain 2011 unlockEffectChains(effectChains); 2012 mLastWriteTime = systemTime(); 2013 mInWrite = true; 2014 mBytesWritten += mixBufferSize; 2015 2016 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2017 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2018 mNumWrites++; 2019 mInWrite = false; 2020 nsecs_t now = systemTime(); 2021 nsecs_t delta = now - mLastWriteTime; 2022 if (!mStandby && delta > maxPeriod) { 2023 mNumDelayedWrites++; 2024 if ((now - lastWarning) > kWarningThrottleNs) { 2025 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2026 ns2ms(delta), mNumDelayedWrites, this); 2027 lastWarning = now; 2028 } 2029 if (mStandby) { 2030 longStandbyExit = true; 2031 } 2032 } 2033 mStandby = false; 2034 } else { 2035 // enable changes in effect chain 2036 unlockEffectChains(effectChains); 2037 usleep(sleepTime); 2038 } 2039 2040 // finally let go of all our tracks, without the lock held 2041 // since we can't guarantee the destructors won't acquire that 2042 // same lock. 2043 tracksToRemove.clear(); 2044 2045 // Effect chains will be actually deleted here if they were removed from 2046 // mEffectChains list during mixing or effects processing 2047 effectChains.clear(); 2048 } 2049 2050 if (!mStandby) { 2051 mOutput->stream->common.standby(&mOutput->stream->common); 2052 } 2053 2054 releaseWakeLock(); 2055 2056 ALOGV("MixerThread %p exiting", this); 2057 return false; 2058} 2059 2060// prepareTracks_l() must be called with ThreadBase::mLock held 2061AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2062 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2063{ 2064 2065 mixer_state mixerStatus = MIXER_IDLE; 2066 // find out which tracks need to be processed 2067 size_t count = activeTracks.size(); 2068 size_t mixedTracks = 0; 2069 size_t tracksWithEffect = 0; 2070 2071 float masterVolume = mMasterVolume; 2072 bool masterMute = mMasterMute; 2073 2074 if (masterMute) { 2075 masterVolume = 0; 2076 } 2077 // Delegate master volume control to effect in output mix effect chain if needed 2078 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2079 if (chain != 0) { 2080 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2081 chain->setVolume_l(&v, &v); 2082 masterVolume = (float)((v + (1 << 23)) >> 24); 2083 chain.clear(); 2084 } 2085 2086 for (size_t i=0 ; i<count ; i++) { 2087 sp<Track> t = activeTracks[i].promote(); 2088 if (t == 0) continue; 2089 2090 // this const just means the local variable doesn't change 2091 Track* const track = t.get(); 2092 audio_track_cblk_t* cblk = track->cblk(); 2093 2094 // The first time a track is added we wait 2095 // for all its buffers to be filled before processing it 2096 int name = track->name(); 2097 // make sure that we have enough frames to mix one full buffer. 2098 // enforce this condition only once to enable draining the buffer in case the client 2099 // app does not call stop() and relies on underrun to stop: 2100 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2101 // during last round 2102 uint32_t minFrames = 1; 2103 if (!track->isStopped() && !track->isPausing() && 2104 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2105 if (t->sampleRate() == (int)mSampleRate) { 2106 minFrames = mFrameCount; 2107 } else { 2108 // +1 for rounding and +1 for additional sample needed for interpolation 2109 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2110 // add frames already consumed but not yet released by the resampler 2111 // because cblk->framesReady() will include these frames 2112 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2113 // the minimum track buffer size is normally twice the number of frames necessary 2114 // to fill one buffer and the resampler should not leave more than one buffer worth 2115 // of unreleased frames after each pass, but just in case... 2116 ALOG_ASSERT(minFrames <= cblk->frameCount); 2117 } 2118 } 2119 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2120 !track->isPaused() && !track->isTerminated()) 2121 { 2122 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2123 2124 mixedTracks++; 2125 2126 // track->mainBuffer() != mMixBuffer means there is an effect chain 2127 // connected to the track 2128 chain.clear(); 2129 if (track->mainBuffer() != mMixBuffer) { 2130 chain = getEffectChain_l(track->sessionId()); 2131 // Delegate volume control to effect in track effect chain if needed 2132 if (chain != 0) { 2133 tracksWithEffect++; 2134 } else { 2135 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2136 name, track->sessionId()); 2137 } 2138 } 2139 2140 2141 int param = AudioMixer::VOLUME; 2142 if (track->mFillingUpStatus == Track::FS_FILLED) { 2143 // no ramp for the first volume setting 2144 track->mFillingUpStatus = Track::FS_ACTIVE; 2145 if (track->mState == TrackBase::RESUMING) { 2146 track->mState = TrackBase::ACTIVE; 2147 param = AudioMixer::RAMP_VOLUME; 2148 } 2149 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2150 } else if (cblk->server != 0) { 2151 // If the track is stopped before the first frame was mixed, 2152 // do not apply ramp 2153 param = AudioMixer::RAMP_VOLUME; 2154 } 2155 2156 // compute volume for this track 2157 uint32_t vl, vr, va; 2158 if (track->isMuted() || track->isPausing() || 2159 mStreamTypes[track->type()].mute) { 2160 vl = vr = va = 0; 2161 if (track->isPausing()) { 2162 track->setPaused(); 2163 } 2164 } else { 2165 2166 // read original volumes with volume control 2167 float typeVolume = mStreamTypes[track->type()].volume; 2168 float v = masterVolume * typeVolume; 2169 uint32_t vlr = cblk->getVolumeLR(); 2170 vl = vlr & 0xFFFF; 2171 vr = vlr >> 16; 2172 // track volumes come from shared memory, so can't be trusted and must be clamped 2173 if (vl > MAX_GAIN_INT) { 2174 ALOGV("Track left volume out of range: %04X", vl); 2175 vl = MAX_GAIN_INT; 2176 } 2177 if (vr > MAX_GAIN_INT) { 2178 ALOGV("Track right volume out of range: %04X", vr); 2179 vr = MAX_GAIN_INT; 2180 } 2181 // now apply the master volume and stream type volume 2182 vl = (uint32_t)(v * vl) << 12; 2183 vr = (uint32_t)(v * vr) << 12; 2184 // assuming master volume and stream type volume each go up to 1.0, 2185 // vl and vr are now in 8.24 format 2186 2187 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2188 // send level comes from shared memory and so may be corrupt 2189 if (sendLevel >= MAX_GAIN_INT) { 2190 ALOGV("Track send level out of range: %04X", sendLevel); 2191 sendLevel = MAX_GAIN_INT; 2192 } 2193 va = (uint32_t)(v * sendLevel); 2194 } 2195 // Delegate volume control to effect in track effect chain if needed 2196 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2197 // Do not ramp volume if volume is controlled by effect 2198 param = AudioMixer::VOLUME; 2199 track->mHasVolumeController = true; 2200 } else { 2201 // force no volume ramp when volume controller was just disabled or removed 2202 // from effect chain to avoid volume spike 2203 if (track->mHasVolumeController) { 2204 param = AudioMixer::VOLUME; 2205 } 2206 track->mHasVolumeController = false; 2207 } 2208 2209 // Convert volumes from 8.24 to 4.12 format 2210 int16_t left, right, aux; 2211 // This additional clamping is needed in case chain->setVolume_l() overshot 2212 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2213 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2214 left = int16_t(v_clamped); 2215 v_clamped = (vr + (1 << 11)) >> 12; 2216 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2217 right = int16_t(v_clamped); 2218 2219 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2220 aux = int16_t(va); 2221 2222 // XXX: these things DON'T need to be done each time 2223 mAudioMixer->setBufferProvider(name, track); 2224 mAudioMixer->enable(name); 2225 2226 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2227 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2228 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2229 mAudioMixer->setParameter( 2230 name, 2231 AudioMixer::TRACK, 2232 AudioMixer::FORMAT, (void *)track->format()); 2233 mAudioMixer->setParameter( 2234 name, 2235 AudioMixer::TRACK, 2236 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2237 mAudioMixer->setParameter( 2238 name, 2239 AudioMixer::RESAMPLE, 2240 AudioMixer::SAMPLE_RATE, 2241 (void *)(cblk->sampleRate)); 2242 mAudioMixer->setParameter( 2243 name, 2244 AudioMixer::TRACK, 2245 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2246 mAudioMixer->setParameter( 2247 name, 2248 AudioMixer::TRACK, 2249 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2250 2251 // reset retry count 2252 track->mRetryCount = kMaxTrackRetries; 2253 // If one track is ready, set the mixer ready if: 2254 // - the mixer was not ready during previous round OR 2255 // - no other track is not ready 2256 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2257 mixerStatus != MIXER_TRACKS_ENABLED) { 2258 mixerStatus = MIXER_TRACKS_READY; 2259 } 2260 } else { 2261 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2262 if (track->isStopped()) { 2263 track->reset(); 2264 } 2265 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2266 // We have consumed all the buffers of this track. 2267 // Remove it from the list of active tracks. 2268 tracksToRemove->add(track); 2269 } else { 2270 // No buffers for this track. Give it a few chances to 2271 // fill a buffer, then remove it from active list. 2272 if (--(track->mRetryCount) <= 0) { 2273 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2274 tracksToRemove->add(track); 2275 // indicate to client process that the track was disabled because of underrun 2276 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2277 // If one track is not ready, mark the mixer also not ready if: 2278 // - the mixer was ready during previous round OR 2279 // - no other track is ready 2280 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2281 mixerStatus != MIXER_TRACKS_READY) { 2282 mixerStatus = MIXER_TRACKS_ENABLED; 2283 } 2284 } 2285 mAudioMixer->disable(name); 2286 } 2287 } 2288 2289 // remove all the tracks that need to be... 2290 count = tracksToRemove->size(); 2291 if (CC_UNLIKELY(count)) { 2292 for (size_t i=0 ; i<count ; i++) { 2293 const sp<Track>& track = tracksToRemove->itemAt(i); 2294 mActiveTracks.remove(track); 2295 if (track->mainBuffer() != mMixBuffer) { 2296 chain = getEffectChain_l(track->sessionId()); 2297 if (chain != 0) { 2298 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2299 chain->decActiveTrackCnt(); 2300 } 2301 } 2302 if (track->isTerminated()) { 2303 removeTrack_l(track); 2304 } 2305 } 2306 } 2307 2308 // mix buffer must be cleared if all tracks are connected to an 2309 // effect chain as in this case the mixer will not write to 2310 // mix buffer and track effects will accumulate into it 2311 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2312 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2313 } 2314 2315 mPrevMixerStatus = mixerStatus; 2316 return mixerStatus; 2317} 2318 2319void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2320{ 2321 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2322 this, streamType, mTracks.size()); 2323 Mutex::Autolock _l(mLock); 2324 2325 size_t size = mTracks.size(); 2326 for (size_t i = 0; i < size; i++) { 2327 sp<Track> t = mTracks[i]; 2328 if (t->type() == streamType) { 2329 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2330 t->mCblk->cv.signal(); 2331 } 2332 } 2333} 2334 2335void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2336{ 2337 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2338 this, streamType, valid); 2339 Mutex::Autolock _l(mLock); 2340 2341 mStreamTypes[streamType].valid = valid; 2342} 2343 2344// getTrackName_l() must be called with ThreadBase::mLock held 2345int AudioFlinger::MixerThread::getTrackName_l() 2346{ 2347 return mAudioMixer->getTrackName(); 2348} 2349 2350// deleteTrackName_l() must be called with ThreadBase::mLock held 2351void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2352{ 2353 ALOGV("remove track (%d) and delete from mixer", name); 2354 mAudioMixer->deleteTrackName(name); 2355} 2356 2357// checkForNewParameters_l() must be called with ThreadBase::mLock held 2358bool AudioFlinger::MixerThread::checkForNewParameters_l() 2359{ 2360 bool reconfig = false; 2361 2362 while (!mNewParameters.isEmpty()) { 2363 status_t status = NO_ERROR; 2364 String8 keyValuePair = mNewParameters[0]; 2365 AudioParameter param = AudioParameter(keyValuePair); 2366 int value; 2367 2368 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2369 reconfig = true; 2370 } 2371 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2372 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2373 status = BAD_VALUE; 2374 } else { 2375 reconfig = true; 2376 } 2377 } 2378 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2379 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2380 status = BAD_VALUE; 2381 } else { 2382 reconfig = true; 2383 } 2384 } 2385 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2386 // do not accept frame count changes if tracks are open as the track buffer 2387 // size depends on frame count and correct behavior would not be guaranteed 2388 // if frame count is changed after track creation 2389 if (!mTracks.isEmpty()) { 2390 status = INVALID_OPERATION; 2391 } else { 2392 reconfig = true; 2393 } 2394 } 2395 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2396 // when changing the audio output device, call addBatteryData to notify 2397 // the change 2398 if ((int)mDevice != value) { 2399 uint32_t params = 0; 2400 // check whether speaker is on 2401 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2402 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2403 } 2404 2405 int deviceWithoutSpeaker 2406 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2407 // check if any other device (except speaker) is on 2408 if (value & deviceWithoutSpeaker ) { 2409 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2410 } 2411 2412 if (params != 0) { 2413 addBatteryData(params); 2414 } 2415 } 2416 2417 // forward device change to effects that have requested to be 2418 // aware of attached audio device. 2419 mDevice = (uint32_t)value; 2420 for (size_t i = 0; i < mEffectChains.size(); i++) { 2421 mEffectChains[i]->setDevice_l(mDevice); 2422 } 2423 } 2424 2425 if (status == NO_ERROR) { 2426 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2427 keyValuePair.string()); 2428 if (!mStandby && status == INVALID_OPERATION) { 2429 mOutput->stream->common.standby(&mOutput->stream->common); 2430 mStandby = true; 2431 mBytesWritten = 0; 2432 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2433 keyValuePair.string()); 2434 } 2435 if (status == NO_ERROR && reconfig) { 2436 delete mAudioMixer; 2437 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2438 mAudioMixer = NULL; 2439 readOutputParameters(); 2440 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2441 for (size_t i = 0; i < mTracks.size() ; i++) { 2442 int name = getTrackName_l(); 2443 if (name < 0) break; 2444 mTracks[i]->mName = name; 2445 // limit track sample rate to 2 x new output sample rate 2446 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2447 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2448 } 2449 } 2450 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2451 } 2452 } 2453 2454 mNewParameters.removeAt(0); 2455 2456 mParamStatus = status; 2457 mParamCond.signal(); 2458 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2459 // already timed out waiting for the status and will never signal the condition. 2460 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2461 } 2462 return reconfig; 2463} 2464 2465status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2466{ 2467 const size_t SIZE = 256; 2468 char buffer[SIZE]; 2469 String8 result; 2470 2471 PlaybackThread::dumpInternals(fd, args); 2472 2473 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2474 result.append(buffer); 2475 write(fd, result.string(), result.size()); 2476 return NO_ERROR; 2477} 2478 2479uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2480{ 2481 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2482} 2483 2484uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2485{ 2486 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2487} 2488 2489// ---------------------------------------------------------------------------- 2490AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2491 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2492 // mLeftVolFloat, mRightVolFloat 2493 // mLeftVolShort, mRightVolShort 2494{ 2495} 2496 2497AudioFlinger::DirectOutputThread::~DirectOutputThread() 2498{ 2499} 2500 2501static inline 2502int32_t mul(int16_t in, int16_t v) 2503{ 2504#if defined(__arm__) && !defined(__thumb__) 2505 int32_t out; 2506 asm( "smulbb %[out], %[in], %[v] \n" 2507 : [out]"=r"(out) 2508 : [in]"%r"(in), [v]"r"(v) 2509 : ); 2510 return out; 2511#else 2512 return in * int32_t(v); 2513#endif 2514} 2515 2516void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2517{ 2518 // Do not apply volume on compressed audio 2519 if (!audio_is_linear_pcm(mFormat)) { 2520 return; 2521 } 2522 2523 // convert to signed 16 bit before volume calculation 2524 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2525 size_t count = mFrameCount * mChannelCount; 2526 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2527 int16_t *dst = mMixBuffer + count-1; 2528 while(count--) { 2529 *dst-- = (int16_t)(*src--^0x80) << 8; 2530 } 2531 } 2532 2533 size_t frameCount = mFrameCount; 2534 int16_t *out = mMixBuffer; 2535 if (ramp) { 2536 if (mChannelCount == 1) { 2537 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2538 int32_t vlInc = d / (int32_t)frameCount; 2539 int32_t vl = ((int32_t)mLeftVolShort << 16); 2540 do { 2541 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2542 out++; 2543 vl += vlInc; 2544 } while (--frameCount); 2545 2546 } else { 2547 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2548 int32_t vlInc = d / (int32_t)frameCount; 2549 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2550 int32_t vrInc = d / (int32_t)frameCount; 2551 int32_t vl = ((int32_t)mLeftVolShort << 16); 2552 int32_t vr = ((int32_t)mRightVolShort << 16); 2553 do { 2554 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2555 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2556 out += 2; 2557 vl += vlInc; 2558 vr += vrInc; 2559 } while (--frameCount); 2560 } 2561 } else { 2562 if (mChannelCount == 1) { 2563 do { 2564 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2565 out++; 2566 } while (--frameCount); 2567 } else { 2568 do { 2569 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2570 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2571 out += 2; 2572 } while (--frameCount); 2573 } 2574 } 2575 2576 // convert back to unsigned 8 bit after volume calculation 2577 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2578 size_t count = mFrameCount * mChannelCount; 2579 int16_t *src = mMixBuffer; 2580 uint8_t *dst = (uint8_t *)mMixBuffer; 2581 while(count--) { 2582 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2583 } 2584 } 2585 2586 mLeftVolShort = leftVol; 2587 mRightVolShort = rightVol; 2588} 2589 2590bool AudioFlinger::DirectOutputThread::threadLoop() 2591{ 2592 mixer_state mixerStatus = MIXER_IDLE; 2593 sp<Track> trackToRemove; 2594 sp<Track> activeTrack; 2595 nsecs_t standbyTime = systemTime(); 2596 int8_t *curBuf; 2597 size_t mixBufferSize = mFrameCount*mFrameSize; 2598 uint32_t activeSleepTime = activeSleepTimeUs(); 2599 uint32_t idleSleepTime = idleSleepTimeUs(); 2600 uint32_t sleepTime = idleSleepTime; 2601 // use shorter standby delay as on normal output to release 2602 // hardware resources as soon as possible 2603 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2604 2605 acquireWakeLock(); 2606 2607 while (!exitPending()) 2608 { 2609 bool rampVolume; 2610 uint16_t leftVol; 2611 uint16_t rightVol; 2612 Vector< sp<EffectChain> > effectChains; 2613 2614 processConfigEvents(); 2615 2616 mixerStatus = MIXER_IDLE; 2617 2618 { // scope for the mLock 2619 2620 Mutex::Autolock _l(mLock); 2621 2622 if (checkForNewParameters_l()) { 2623 mixBufferSize = mFrameCount*mFrameSize; 2624 activeSleepTime = activeSleepTimeUs(); 2625 idleSleepTime = idleSleepTimeUs(); 2626 standbyDelay = microseconds(activeSleepTime*2); 2627 } 2628 2629 // put audio hardware into standby after short delay 2630 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2631 mSuspended)) { 2632 // wait until we have something to do... 2633 if (!mStandby) { 2634 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2635 mOutput->stream->common.standby(&mOutput->stream->common); 2636 mStandby = true; 2637 mBytesWritten = 0; 2638 } 2639 2640 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2641 // we're about to wait, flush the binder command buffer 2642 IPCThreadState::self()->flushCommands(); 2643 2644 if (exitPending()) break; 2645 2646 releaseWakeLock_l(); 2647 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2648 mWaitWorkCV.wait(mLock); 2649 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2650 acquireWakeLock_l(); 2651 2652 if (!mMasterMute) { 2653 char value[PROPERTY_VALUE_MAX]; 2654 property_get("ro.audio.silent", value, "0"); 2655 if (atoi(value)) { 2656 ALOGD("Silence is golden"); 2657 setMasterMute(true); 2658 } 2659 } 2660 2661 standbyTime = systemTime() + standbyDelay; 2662 sleepTime = idleSleepTime; 2663 continue; 2664 } 2665 } 2666 2667 effectChains = mEffectChains; 2668 2669 // find out which tracks need to be processed 2670 if (mActiveTracks.size() != 0) { 2671 sp<Track> t = mActiveTracks[0].promote(); 2672 if (t == 0) continue; 2673 2674 Track* const track = t.get(); 2675 audio_track_cblk_t* cblk = track->cblk(); 2676 2677 // The first time a track is added we wait 2678 // for all its buffers to be filled before processing it 2679 if (cblk->framesReady() && track->isReady() && 2680 !track->isPaused() && !track->isTerminated()) 2681 { 2682 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2683 2684 if (track->mFillingUpStatus == Track::FS_FILLED) { 2685 track->mFillingUpStatus = Track::FS_ACTIVE; 2686 mLeftVolFloat = mRightVolFloat = 0; 2687 mLeftVolShort = mRightVolShort = 0; 2688 if (track->mState == TrackBase::RESUMING) { 2689 track->mState = TrackBase::ACTIVE; 2690 rampVolume = true; 2691 } 2692 } else if (cblk->server != 0) { 2693 // If the track is stopped before the first frame was mixed, 2694 // do not apply ramp 2695 rampVolume = true; 2696 } 2697 // compute volume for this track 2698 float left, right; 2699 if (track->isMuted() || mMasterMute || track->isPausing() || 2700 mStreamTypes[track->type()].mute) { 2701 left = right = 0; 2702 if (track->isPausing()) { 2703 track->setPaused(); 2704 } 2705 } else { 2706 float typeVolume = mStreamTypes[track->type()].volume; 2707 float v = mMasterVolume * typeVolume; 2708 uint32_t vlr = cblk->getVolumeLR(); 2709 float v_clamped = v * (vlr & 0xFFFF); 2710 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2711 left = v_clamped/MAX_GAIN; 2712 v_clamped = v * (vlr >> 16); 2713 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2714 right = v_clamped/MAX_GAIN; 2715 } 2716 2717 if (left != mLeftVolFloat || right != mRightVolFloat) { 2718 mLeftVolFloat = left; 2719 mRightVolFloat = right; 2720 2721 // If audio HAL implements volume control, 2722 // force software volume to nominal value 2723 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2724 left = 1.0f; 2725 right = 1.0f; 2726 } 2727 2728 // Convert volumes from float to 8.24 2729 uint32_t vl = (uint32_t)(left * (1 << 24)); 2730 uint32_t vr = (uint32_t)(right * (1 << 24)); 2731 2732 // Delegate volume control to effect in track effect chain if needed 2733 // only one effect chain can be present on DirectOutputThread, so if 2734 // there is one, the track is connected to it 2735 if (!effectChains.isEmpty()) { 2736 // Do not ramp volume if volume is controlled by effect 2737 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2738 rampVolume = false; 2739 } 2740 } 2741 2742 // Convert volumes from 8.24 to 4.12 format 2743 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2744 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2745 leftVol = (uint16_t)v_clamped; 2746 v_clamped = (vr + (1 << 11)) >> 12; 2747 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2748 rightVol = (uint16_t)v_clamped; 2749 } else { 2750 leftVol = mLeftVolShort; 2751 rightVol = mRightVolShort; 2752 rampVolume = false; 2753 } 2754 2755 // reset retry count 2756 track->mRetryCount = kMaxTrackRetriesDirect; 2757 activeTrack = t; 2758 mixerStatus = MIXER_TRACKS_READY; 2759 } else { 2760 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2761 if (track->isStopped()) { 2762 track->reset(); 2763 } 2764 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2765 // We have consumed all the buffers of this track. 2766 // Remove it from the list of active tracks. 2767 trackToRemove = track; 2768 } else { 2769 // No buffers for this track. Give it a few chances to 2770 // fill a buffer, then remove it from active list. 2771 if (--(track->mRetryCount) <= 0) { 2772 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2773 trackToRemove = track; 2774 } else { 2775 mixerStatus = MIXER_TRACKS_ENABLED; 2776 } 2777 } 2778 } 2779 } 2780 2781 // remove all the tracks that need to be... 2782 if (CC_UNLIKELY(trackToRemove != 0)) { 2783 mActiveTracks.remove(trackToRemove); 2784 if (!effectChains.isEmpty()) { 2785 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2786 trackToRemove->sessionId()); 2787 effectChains[0]->decActiveTrackCnt(); 2788 } 2789 if (trackToRemove->isTerminated()) { 2790 removeTrack_l(trackToRemove); 2791 } 2792 } 2793 2794 lockEffectChains_l(effectChains); 2795 } 2796 2797 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2798 AudioBufferProvider::Buffer buffer; 2799 size_t frameCount = mFrameCount; 2800 curBuf = (int8_t *)mMixBuffer; 2801 // output audio to hardware 2802 while (frameCount) { 2803 buffer.frameCount = frameCount; 2804 activeTrack->getNextBuffer(&buffer); 2805 if (CC_UNLIKELY(buffer.raw == NULL)) { 2806 memset(curBuf, 0, frameCount * mFrameSize); 2807 break; 2808 } 2809 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2810 frameCount -= buffer.frameCount; 2811 curBuf += buffer.frameCount * mFrameSize; 2812 activeTrack->releaseBuffer(&buffer); 2813 } 2814 sleepTime = 0; 2815 standbyTime = systemTime() + standbyDelay; 2816 } else { 2817 if (sleepTime == 0) { 2818 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2819 sleepTime = activeSleepTime; 2820 } else { 2821 sleepTime = idleSleepTime; 2822 } 2823 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2824 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2825 sleepTime = 0; 2826 } 2827 } 2828 2829 if (mSuspended) { 2830 sleepTime = suspendSleepTimeUs(); 2831 } 2832 // sleepTime == 0 means we must write to audio hardware 2833 if (sleepTime == 0) { 2834 if (mixerStatus == MIXER_TRACKS_READY) { 2835 applyVolume(leftVol, rightVol, rampVolume); 2836 } 2837 for (size_t i = 0; i < effectChains.size(); i ++) { 2838 effectChains[i]->process_l(); 2839 } 2840 unlockEffectChains(effectChains); 2841 2842 mLastWriteTime = systemTime(); 2843 mInWrite = true; 2844 mBytesWritten += mixBufferSize; 2845 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2846 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2847 mNumWrites++; 2848 mInWrite = false; 2849 mStandby = false; 2850 } else { 2851 unlockEffectChains(effectChains); 2852 usleep(sleepTime); 2853 } 2854 2855 // finally let go of removed track, without the lock held 2856 // since we can't guarantee the destructors won't acquire that 2857 // same lock. 2858 trackToRemove.clear(); 2859 activeTrack.clear(); 2860 2861 // Effect chains will be actually deleted here if they were removed from 2862 // mEffectChains list during mixing or effects processing 2863 effectChains.clear(); 2864 } 2865 2866 if (!mStandby) { 2867 mOutput->stream->common.standby(&mOutput->stream->common); 2868 } 2869 2870 releaseWakeLock(); 2871 2872 ALOGV("DirectOutputThread %p exiting", this); 2873 return false; 2874} 2875 2876// getTrackName_l() must be called with ThreadBase::mLock held 2877int AudioFlinger::DirectOutputThread::getTrackName_l() 2878{ 2879 return 0; 2880} 2881 2882// deleteTrackName_l() must be called with ThreadBase::mLock held 2883void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2884{ 2885} 2886 2887// checkForNewParameters_l() must be called with ThreadBase::mLock held 2888bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2889{ 2890 bool reconfig = false; 2891 2892 while (!mNewParameters.isEmpty()) { 2893 status_t status = NO_ERROR; 2894 String8 keyValuePair = mNewParameters[0]; 2895 AudioParameter param = AudioParameter(keyValuePair); 2896 int value; 2897 2898 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2899 // do not accept frame count changes if tracks are open as the track buffer 2900 // size depends on frame count and correct behavior would not be garantied 2901 // if frame count is changed after track creation 2902 if (!mTracks.isEmpty()) { 2903 status = INVALID_OPERATION; 2904 } else { 2905 reconfig = true; 2906 } 2907 } 2908 if (status == NO_ERROR) { 2909 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2910 keyValuePair.string()); 2911 if (!mStandby && status == INVALID_OPERATION) { 2912 mOutput->stream->common.standby(&mOutput->stream->common); 2913 mStandby = true; 2914 mBytesWritten = 0; 2915 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2916 keyValuePair.string()); 2917 } 2918 if (status == NO_ERROR && reconfig) { 2919 readOutputParameters(); 2920 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2921 } 2922 } 2923 2924 mNewParameters.removeAt(0); 2925 2926 mParamStatus = status; 2927 mParamCond.signal(); 2928 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2929 // already timed out waiting for the status and will never signal the condition. 2930 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2931 } 2932 return reconfig; 2933} 2934 2935uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2936{ 2937 uint32_t time; 2938 if (audio_is_linear_pcm(mFormat)) { 2939 time = PlaybackThread::activeSleepTimeUs(); 2940 } else { 2941 time = 10000; 2942 } 2943 return time; 2944} 2945 2946uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2947{ 2948 uint32_t time; 2949 if (audio_is_linear_pcm(mFormat)) { 2950 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2951 } else { 2952 time = 10000; 2953 } 2954 return time; 2955} 2956 2957uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2958{ 2959 uint32_t time; 2960 if (audio_is_linear_pcm(mFormat)) { 2961 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2962 } else { 2963 time = 10000; 2964 } 2965 return time; 2966} 2967 2968 2969// ---------------------------------------------------------------------------- 2970 2971AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 2972 AudioFlinger::MixerThread* mainThread, int id) 2973 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 2974 mWaitTimeMs(UINT_MAX) 2975{ 2976 addOutputTrack(mainThread); 2977} 2978 2979AudioFlinger::DuplicatingThread::~DuplicatingThread() 2980{ 2981 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2982 mOutputTracks[i]->destroy(); 2983 } 2984} 2985 2986bool AudioFlinger::DuplicatingThread::threadLoop() 2987{ 2988 Vector< sp<Track> > tracksToRemove; 2989 mixer_state mixerStatus = MIXER_IDLE; 2990 nsecs_t standbyTime = systemTime(); 2991 size_t mixBufferSize = mFrameCount*mFrameSize; 2992 SortedVector< sp<OutputTrack> > outputTracks; 2993 uint32_t writeFrames = 0; 2994 uint32_t activeSleepTime = activeSleepTimeUs(); 2995 uint32_t idleSleepTime = idleSleepTimeUs(); 2996 uint32_t sleepTime = idleSleepTime; 2997 Vector< sp<EffectChain> > effectChains; 2998 2999 acquireWakeLock(); 3000 3001 while (!exitPending()) 3002 { 3003 processConfigEvents(); 3004 3005 mixerStatus = MIXER_IDLE; 3006 { // scope for the mLock 3007 3008 Mutex::Autolock _l(mLock); 3009 3010 if (checkForNewParameters_l()) { 3011 mixBufferSize = mFrameCount*mFrameSize; 3012 updateWaitTime(); 3013 activeSleepTime = activeSleepTimeUs(); 3014 idleSleepTime = idleSleepTimeUs(); 3015 } 3016 3017 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3018 3019 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3020 outputTracks.add(mOutputTracks[i]); 3021 } 3022 3023 // put audio hardware into standby after short delay 3024 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3025 mSuspended)) { 3026 if (!mStandby) { 3027 for (size_t i = 0; i < outputTracks.size(); i++) { 3028 outputTracks[i]->stop(); 3029 } 3030 mStandby = true; 3031 mBytesWritten = 0; 3032 } 3033 3034 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3035 // we're about to wait, flush the binder command buffer 3036 IPCThreadState::self()->flushCommands(); 3037 outputTracks.clear(); 3038 3039 if (exitPending()) break; 3040 3041 releaseWakeLock_l(); 3042 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3043 mWaitWorkCV.wait(mLock); 3044 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3045 acquireWakeLock_l(); 3046 3047 mPrevMixerStatus = MIXER_IDLE; 3048 if (!mMasterMute) { 3049 char value[PROPERTY_VALUE_MAX]; 3050 property_get("ro.audio.silent", value, "0"); 3051 if (atoi(value)) { 3052 ALOGD("Silence is golden"); 3053 setMasterMute(true); 3054 } 3055 } 3056 3057 standbyTime = systemTime() + kStandbyTimeInNsecs; 3058 sleepTime = idleSleepTime; 3059 continue; 3060 } 3061 } 3062 3063 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3064 3065 // prevent any changes in effect chain list and in each effect chain 3066 // during mixing and effect process as the audio buffers could be deleted 3067 // or modified if an effect is created or deleted 3068 lockEffectChains_l(effectChains); 3069 } 3070 3071 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3072 // mix buffers... 3073 if (outputsReady(outputTracks)) { 3074 mAudioMixer->process(); 3075 } else { 3076 memset(mMixBuffer, 0, mixBufferSize); 3077 } 3078 sleepTime = 0; 3079 writeFrames = mFrameCount; 3080 } else { 3081 if (sleepTime == 0) { 3082 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3083 sleepTime = activeSleepTime; 3084 } else { 3085 sleepTime = idleSleepTime; 3086 } 3087 } else if (mBytesWritten != 0) { 3088 // flush remaining overflow buffers in output tracks 3089 for (size_t i = 0; i < outputTracks.size(); i++) { 3090 if (outputTracks[i]->isActive()) { 3091 sleepTime = 0; 3092 writeFrames = 0; 3093 memset(mMixBuffer, 0, mixBufferSize); 3094 break; 3095 } 3096 } 3097 } 3098 } 3099 3100 if (mSuspended) { 3101 sleepTime = suspendSleepTimeUs(); 3102 } 3103 // sleepTime == 0 means we must write to audio hardware 3104 if (sleepTime == 0) { 3105 for (size_t i = 0; i < effectChains.size(); i ++) { 3106 effectChains[i]->process_l(); 3107 } 3108 // enable changes in effect chain 3109 unlockEffectChains(effectChains); 3110 3111 standbyTime = systemTime() + kStandbyTimeInNsecs; 3112 for (size_t i = 0; i < outputTracks.size(); i++) { 3113 outputTracks[i]->write(mMixBuffer, writeFrames); 3114 } 3115 mStandby = false; 3116 mBytesWritten += mixBufferSize; 3117 } else { 3118 // enable changes in effect chain 3119 unlockEffectChains(effectChains); 3120 usleep(sleepTime); 3121 } 3122 3123 // finally let go of all our tracks, without the lock held 3124 // since we can't guarantee the destructors won't acquire that 3125 // same lock. 3126 tracksToRemove.clear(); 3127 outputTracks.clear(); 3128 3129 // Effect chains will be actually deleted here if they were removed from 3130 // mEffectChains list during mixing or effects processing 3131 effectChains.clear(); 3132 } 3133 3134 releaseWakeLock(); 3135 3136 return false; 3137} 3138 3139void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3140{ 3141 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3142 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3143 this, 3144 mSampleRate, 3145 mFormat, 3146 mChannelMask, 3147 frameCount); 3148 if (outputTrack->cblk() != NULL) { 3149 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3150 mOutputTracks.add(outputTrack); 3151 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3152 updateWaitTime(); 3153 } 3154} 3155 3156void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3157{ 3158 Mutex::Autolock _l(mLock); 3159 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3160 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3161 mOutputTracks[i]->destroy(); 3162 mOutputTracks.removeAt(i); 3163 updateWaitTime(); 3164 return; 3165 } 3166 } 3167 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3168} 3169 3170void AudioFlinger::DuplicatingThread::updateWaitTime() 3171{ 3172 mWaitTimeMs = UINT_MAX; 3173 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3174 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3175 if (strong != 0) { 3176 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3177 if (waitTimeMs < mWaitTimeMs) { 3178 mWaitTimeMs = waitTimeMs; 3179 } 3180 } 3181 } 3182} 3183 3184 3185bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3186{ 3187 for (size_t i = 0; i < outputTracks.size(); i++) { 3188 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3189 if (thread == 0) { 3190 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3191 return false; 3192 } 3193 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3194 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3195 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3196 return false; 3197 } 3198 } 3199 return true; 3200} 3201 3202uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3203{ 3204 return (mWaitTimeMs * 1000) / 2; 3205} 3206 3207// ---------------------------------------------------------------------------- 3208 3209// TrackBase constructor must be called with AudioFlinger::mLock held 3210AudioFlinger::ThreadBase::TrackBase::TrackBase( 3211 const wp<ThreadBase>& thread, 3212 const sp<Client>& client, 3213 uint32_t sampleRate, 3214 audio_format_t format, 3215 uint32_t channelMask, 3216 int frameCount, 3217 uint32_t flags, 3218 const sp<IMemory>& sharedBuffer, 3219 int sessionId) 3220 : RefBase(), 3221 mThread(thread), 3222 mClient(client), 3223 mCblk(NULL), 3224 // mBuffer 3225 // mBufferEnd 3226 mFrameCount(0), 3227 mState(IDLE), 3228 mFormat(format), 3229 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3230 mSessionId(sessionId) 3231 // mChannelCount 3232 // mChannelMask 3233{ 3234 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3235 3236 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3237 size_t size = sizeof(audio_track_cblk_t); 3238 uint8_t channelCount = popcount(channelMask); 3239 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3240 if (sharedBuffer == 0) { 3241 size += bufferSize; 3242 } 3243 3244 if (client != NULL) { 3245 mCblkMemory = client->heap()->allocate(size); 3246 if (mCblkMemory != 0) { 3247 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3248 if (mCblk != NULL) { // construct the shared structure in-place. 3249 new(mCblk) audio_track_cblk_t(); 3250 // clear all buffers 3251 mCblk->frameCount = frameCount; 3252 mCblk->sampleRate = sampleRate; 3253 mChannelCount = channelCount; 3254 mChannelMask = channelMask; 3255 if (sharedBuffer == 0) { 3256 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3257 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3258 // Force underrun condition to avoid false underrun callback until first data is 3259 // written to buffer (other flags are cleared) 3260 mCblk->flags = CBLK_UNDERRUN_ON; 3261 } else { 3262 mBuffer = sharedBuffer->pointer(); 3263 } 3264 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3265 } 3266 } else { 3267 ALOGE("not enough memory for AudioTrack size=%u", size); 3268 client->heap()->dump("AudioTrack"); 3269 return; 3270 } 3271 } else { 3272 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3273 // construct the shared structure in-place. 3274 new(mCblk) audio_track_cblk_t(); 3275 // clear all buffers 3276 mCblk->frameCount = frameCount; 3277 mCblk->sampleRate = sampleRate; 3278 mChannelCount = channelCount; 3279 mChannelMask = channelMask; 3280 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3281 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3282 // Force underrun condition to avoid false underrun callback until first data is 3283 // written to buffer (other flags are cleared) 3284 mCblk->flags = CBLK_UNDERRUN_ON; 3285 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3286 } 3287} 3288 3289AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3290{ 3291 if (mCblk != NULL) { 3292 if (mClient == 0) { 3293 delete mCblk; 3294 } else { 3295 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3296 } 3297 } 3298 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3299 if (mClient != 0) { 3300 // Client destructor must run with AudioFlinger mutex locked 3301 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3302 // If the client's reference count drops to zero, the associated destructor 3303 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3304 // relying on the automatic clear() at end of scope. 3305 mClient.clear(); 3306 } 3307} 3308 3309void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3310{ 3311 buffer->raw = NULL; 3312 mFrameCount = buffer->frameCount; 3313 step(); 3314 buffer->frameCount = 0; 3315} 3316 3317bool AudioFlinger::ThreadBase::TrackBase::step() { 3318 bool result; 3319 audio_track_cblk_t* cblk = this->cblk(); 3320 3321 result = cblk->stepServer(mFrameCount); 3322 if (!result) { 3323 ALOGV("stepServer failed acquiring cblk mutex"); 3324 mFlags |= STEPSERVER_FAILED; 3325 } 3326 return result; 3327} 3328 3329void AudioFlinger::ThreadBase::TrackBase::reset() { 3330 audio_track_cblk_t* cblk = this->cblk(); 3331 3332 cblk->user = 0; 3333 cblk->server = 0; 3334 cblk->userBase = 0; 3335 cblk->serverBase = 0; 3336 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3337 ALOGV("TrackBase::reset"); 3338} 3339 3340int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3341 return (int)mCblk->sampleRate; 3342} 3343 3344void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3345 audio_track_cblk_t* cblk = this->cblk(); 3346 size_t frameSize = cblk->frameSize; 3347 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3348 int8_t *bufferEnd = bufferStart + frames * frameSize; 3349 3350 // Check validity of returned pointer in case the track control block would have been corrupted. 3351 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3352 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3353 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3354 server %d, serverBase %d, user %d, userBase %d", 3355 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3356 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3357 return NULL; 3358 } 3359 3360 return bufferStart; 3361} 3362 3363// ---------------------------------------------------------------------------- 3364 3365// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3366AudioFlinger::PlaybackThread::Track::Track( 3367 const wp<ThreadBase>& thread, 3368 const sp<Client>& client, 3369 audio_stream_type_t streamType, 3370 uint32_t sampleRate, 3371 audio_format_t format, 3372 uint32_t channelMask, 3373 int frameCount, 3374 const sp<IMemory>& sharedBuffer, 3375 int sessionId) 3376 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3377 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3378 mAuxEffectId(0), mHasVolumeController(false) 3379{ 3380 if (mCblk != NULL) { 3381 sp<ThreadBase> baseThread = thread.promote(); 3382 if (baseThread != 0) { 3383 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3384 mName = playbackThread->getTrackName_l(); 3385 mMainBuffer = playbackThread->mixBuffer(); 3386 } 3387 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3388 if (mName < 0) { 3389 ALOGE("no more track names available"); 3390 } 3391 mStreamType = streamType; 3392 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3393 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3394 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3395 } 3396} 3397 3398AudioFlinger::PlaybackThread::Track::~Track() 3399{ 3400 ALOGV("PlaybackThread::Track destructor"); 3401 sp<ThreadBase> thread = mThread.promote(); 3402 if (thread != 0) { 3403 Mutex::Autolock _l(thread->mLock); 3404 mState = TERMINATED; 3405 } 3406} 3407 3408void AudioFlinger::PlaybackThread::Track::destroy() 3409{ 3410 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3411 // by removing it from mTracks vector, so there is a risk that this Tracks's 3412 // desctructor is called. As the destructor needs to lock mLock, 3413 // we must acquire a strong reference on this Track before locking mLock 3414 // here so that the destructor is called only when exiting this function. 3415 // On the other hand, as long as Track::destroy() is only called by 3416 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3417 // this Track with its member mTrack. 3418 sp<Track> keep(this); 3419 { // scope for mLock 3420 sp<ThreadBase> thread = mThread.promote(); 3421 if (thread != 0) { 3422 if (!isOutputTrack()) { 3423 if (mState == ACTIVE || mState == RESUMING) { 3424 AudioSystem::stopOutput(thread->id(), 3425 (audio_stream_type_t)mStreamType, 3426 mSessionId); 3427 3428 // to track the speaker usage 3429 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3430 } 3431 AudioSystem::releaseOutput(thread->id()); 3432 } 3433 Mutex::Autolock _l(thread->mLock); 3434 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3435 playbackThread->destroyTrack_l(this); 3436 } 3437 } 3438} 3439 3440void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3441{ 3442 uint32_t vlr = mCblk->getVolumeLR(); 3443 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", 3444 mName - AudioMixer::TRACK0, 3445 (mClient == 0) ? getpid() : mClient->pid(), 3446 mStreamType, 3447 mFormat, 3448 mChannelMask, 3449 mSessionId, 3450 mFrameCount, 3451 mState, 3452 mMute, 3453 mFillingUpStatus, 3454 mCblk->sampleRate, 3455 vlr & 0xFFFF, 3456 vlr >> 16, 3457 mCblk->server, 3458 mCblk->user, 3459 (int)mMainBuffer, 3460 (int)mAuxBuffer); 3461} 3462 3463status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3464{ 3465 audio_track_cblk_t* cblk = this->cblk(); 3466 uint32_t framesReady; 3467 uint32_t framesReq = buffer->frameCount; 3468 3469 // Check if last stepServer failed, try to step now 3470 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3471 if (!step()) goto getNextBuffer_exit; 3472 ALOGV("stepServer recovered"); 3473 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3474 } 3475 3476 framesReady = cblk->framesReady(); 3477 3478 if (CC_LIKELY(framesReady)) { 3479 uint32_t s = cblk->server; 3480 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3481 3482 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3483 if (framesReq > framesReady) { 3484 framesReq = framesReady; 3485 } 3486 if (s + framesReq > bufferEnd) { 3487 framesReq = bufferEnd - s; 3488 } 3489 3490 buffer->raw = getBuffer(s, framesReq); 3491 if (buffer->raw == NULL) goto getNextBuffer_exit; 3492 3493 buffer->frameCount = framesReq; 3494 return NO_ERROR; 3495 } 3496 3497getNextBuffer_exit: 3498 buffer->raw = NULL; 3499 buffer->frameCount = 0; 3500 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3501 return NOT_ENOUGH_DATA; 3502} 3503 3504bool AudioFlinger::PlaybackThread::Track::isReady() const { 3505 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3506 3507 if (mCblk->framesReady() >= mCblk->frameCount || 3508 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3509 mFillingUpStatus = FS_FILLED; 3510 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3511 return true; 3512 } 3513 return false; 3514} 3515 3516status_t AudioFlinger::PlaybackThread::Track::start() 3517{ 3518 status_t status = NO_ERROR; 3519 ALOGV("start(%d), calling thread %d session %d", 3520 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3521 sp<ThreadBase> thread = mThread.promote(); 3522 if (thread != 0) { 3523 Mutex::Autolock _l(thread->mLock); 3524 track_state state = mState; 3525 // here the track could be either new, or restarted 3526 // in both cases "unstop" the track 3527 if (mState == PAUSED) { 3528 mState = TrackBase::RESUMING; 3529 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3530 } else { 3531 mState = TrackBase::ACTIVE; 3532 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3533 } 3534 3535 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3536 thread->mLock.unlock(); 3537 status = AudioSystem::startOutput(thread->id(), 3538 (audio_stream_type_t)mStreamType, 3539 mSessionId); 3540 thread->mLock.lock(); 3541 3542 // to track the speaker usage 3543 if (status == NO_ERROR) { 3544 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3545 } 3546 } 3547 if (status == NO_ERROR) { 3548 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3549 playbackThread->addTrack_l(this); 3550 } else { 3551 mState = state; 3552 } 3553 } else { 3554 status = BAD_VALUE; 3555 } 3556 return status; 3557} 3558 3559void AudioFlinger::PlaybackThread::Track::stop() 3560{ 3561 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3562 sp<ThreadBase> thread = mThread.promote(); 3563 if (thread != 0) { 3564 Mutex::Autolock _l(thread->mLock); 3565 track_state state = mState; 3566 if (mState > STOPPED) { 3567 mState = STOPPED; 3568 // If the track is not active (PAUSED and buffers full), flush buffers 3569 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3570 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3571 reset(); 3572 } 3573 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3574 } 3575 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3576 thread->mLock.unlock(); 3577 AudioSystem::stopOutput(thread->id(), 3578 (audio_stream_type_t)mStreamType, 3579 mSessionId); 3580 thread->mLock.lock(); 3581 3582 // to track the speaker usage 3583 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3584 } 3585 } 3586} 3587 3588void AudioFlinger::PlaybackThread::Track::pause() 3589{ 3590 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3591 sp<ThreadBase> thread = mThread.promote(); 3592 if (thread != 0) { 3593 Mutex::Autolock _l(thread->mLock); 3594 if (mState == ACTIVE || mState == RESUMING) { 3595 mState = PAUSING; 3596 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3597 if (!isOutputTrack()) { 3598 thread->mLock.unlock(); 3599 AudioSystem::stopOutput(thread->id(), 3600 (audio_stream_type_t)mStreamType, 3601 mSessionId); 3602 thread->mLock.lock(); 3603 3604 // to track the speaker usage 3605 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3606 } 3607 } 3608 } 3609} 3610 3611void AudioFlinger::PlaybackThread::Track::flush() 3612{ 3613 ALOGV("flush(%d)", mName); 3614 sp<ThreadBase> thread = mThread.promote(); 3615 if (thread != 0) { 3616 Mutex::Autolock _l(thread->mLock); 3617 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3618 return; 3619 } 3620 // No point remaining in PAUSED state after a flush => go to 3621 // STOPPED state 3622 mState = STOPPED; 3623 3624 // do not reset the track if it is still in the process of being stopped or paused. 3625 // this will be done by prepareTracks_l() when the track is stopped. 3626 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3627 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3628 reset(); 3629 } 3630 } 3631} 3632 3633void AudioFlinger::PlaybackThread::Track::reset() 3634{ 3635 // Do not reset twice to avoid discarding data written just after a flush and before 3636 // the audioflinger thread detects the track is stopped. 3637 if (!mResetDone) { 3638 TrackBase::reset(); 3639 // Force underrun condition to avoid false underrun callback until first data is 3640 // written to buffer 3641 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3642 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3643 mFillingUpStatus = FS_FILLING; 3644 mResetDone = true; 3645 } 3646} 3647 3648void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3649{ 3650 mMute = muted; 3651} 3652 3653status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3654{ 3655 status_t status = DEAD_OBJECT; 3656 sp<ThreadBase> thread = mThread.promote(); 3657 if (thread != 0) { 3658 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3659 status = playbackThread->attachAuxEffect(this, EffectId); 3660 } 3661 return status; 3662} 3663 3664void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3665{ 3666 mAuxEffectId = EffectId; 3667 mAuxBuffer = buffer; 3668} 3669 3670// ---------------------------------------------------------------------------- 3671 3672// RecordTrack constructor must be called with AudioFlinger::mLock held 3673AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3674 const wp<ThreadBase>& thread, 3675 const sp<Client>& client, 3676 uint32_t sampleRate, 3677 audio_format_t format, 3678 uint32_t channelMask, 3679 int frameCount, 3680 uint32_t flags, 3681 int sessionId) 3682 : TrackBase(thread, client, sampleRate, format, 3683 channelMask, frameCount, flags, 0, sessionId), 3684 mOverflow(false) 3685{ 3686 if (mCblk != NULL) { 3687 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3688 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3689 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3690 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3691 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3692 } else { 3693 mCblk->frameSize = sizeof(int8_t); 3694 } 3695 } 3696} 3697 3698AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3699{ 3700 sp<ThreadBase> thread = mThread.promote(); 3701 if (thread != 0) { 3702 AudioSystem::releaseInput(thread->id()); 3703 } 3704} 3705 3706status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3707{ 3708 audio_track_cblk_t* cblk = this->cblk(); 3709 uint32_t framesAvail; 3710 uint32_t framesReq = buffer->frameCount; 3711 3712 // Check if last stepServer failed, try to step now 3713 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3714 if (!step()) goto getNextBuffer_exit; 3715 ALOGV("stepServer recovered"); 3716 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3717 } 3718 3719 framesAvail = cblk->framesAvailable_l(); 3720 3721 if (CC_LIKELY(framesAvail)) { 3722 uint32_t s = cblk->server; 3723 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3724 3725 if (framesReq > framesAvail) { 3726 framesReq = framesAvail; 3727 } 3728 if (s + framesReq > bufferEnd) { 3729 framesReq = bufferEnd - s; 3730 } 3731 3732 buffer->raw = getBuffer(s, framesReq); 3733 if (buffer->raw == NULL) goto getNextBuffer_exit; 3734 3735 buffer->frameCount = framesReq; 3736 return NO_ERROR; 3737 } 3738 3739getNextBuffer_exit: 3740 buffer->raw = NULL; 3741 buffer->frameCount = 0; 3742 return NOT_ENOUGH_DATA; 3743} 3744 3745status_t AudioFlinger::RecordThread::RecordTrack::start() 3746{ 3747 sp<ThreadBase> thread = mThread.promote(); 3748 if (thread != 0) { 3749 RecordThread *recordThread = (RecordThread *)thread.get(); 3750 return recordThread->start(this); 3751 } else { 3752 return BAD_VALUE; 3753 } 3754} 3755 3756void AudioFlinger::RecordThread::RecordTrack::stop() 3757{ 3758 sp<ThreadBase> thread = mThread.promote(); 3759 if (thread != 0) { 3760 RecordThread *recordThread = (RecordThread *)thread.get(); 3761 recordThread->stop(this); 3762 TrackBase::reset(); 3763 // Force overerrun condition to avoid false overrun callback until first data is 3764 // read from buffer 3765 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3766 } 3767} 3768 3769void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3770{ 3771 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3772 (mClient == 0) ? getpid() : mClient->pid(), 3773 mFormat, 3774 mChannelMask, 3775 mSessionId, 3776 mFrameCount, 3777 mState, 3778 mCblk->sampleRate, 3779 mCblk->server, 3780 mCblk->user); 3781} 3782 3783 3784// ---------------------------------------------------------------------------- 3785 3786AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3787 const wp<ThreadBase>& thread, 3788 DuplicatingThread *sourceThread, 3789 uint32_t sampleRate, 3790 audio_format_t format, 3791 uint32_t channelMask, 3792 int frameCount) 3793 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3794 mActive(false), mSourceThread(sourceThread) 3795{ 3796 3797 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3798 if (mCblk != NULL) { 3799 mCblk->flags |= CBLK_DIRECTION_OUT; 3800 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3801 mOutBuffer.frameCount = 0; 3802 playbackThread->mTracks.add(this); 3803 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3804 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3805 mCblk, mBuffer, mCblk->buffers, 3806 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3807 } else { 3808 ALOGW("Error creating output track on thread %p", playbackThread); 3809 } 3810} 3811 3812AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3813{ 3814 clearBufferQueue(); 3815} 3816 3817status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3818{ 3819 status_t status = Track::start(); 3820 if (status != NO_ERROR) { 3821 return status; 3822 } 3823 3824 mActive = true; 3825 mRetryCount = 127; 3826 return status; 3827} 3828 3829void AudioFlinger::PlaybackThread::OutputTrack::stop() 3830{ 3831 Track::stop(); 3832 clearBufferQueue(); 3833 mOutBuffer.frameCount = 0; 3834 mActive = false; 3835} 3836 3837bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3838{ 3839 Buffer *pInBuffer; 3840 Buffer inBuffer; 3841 uint32_t channelCount = mChannelCount; 3842 bool outputBufferFull = false; 3843 inBuffer.frameCount = frames; 3844 inBuffer.i16 = data; 3845 3846 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3847 3848 if (!mActive && frames != 0) { 3849 start(); 3850 sp<ThreadBase> thread = mThread.promote(); 3851 if (thread != 0) { 3852 MixerThread *mixerThread = (MixerThread *)thread.get(); 3853 if (mCblk->frameCount > frames){ 3854 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3855 uint32_t startFrames = (mCblk->frameCount - frames); 3856 pInBuffer = new Buffer; 3857 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3858 pInBuffer->frameCount = startFrames; 3859 pInBuffer->i16 = pInBuffer->mBuffer; 3860 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3861 mBufferQueue.add(pInBuffer); 3862 } else { 3863 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3864 } 3865 } 3866 } 3867 } 3868 3869 while (waitTimeLeftMs) { 3870 // First write pending buffers, then new data 3871 if (mBufferQueue.size()) { 3872 pInBuffer = mBufferQueue.itemAt(0); 3873 } else { 3874 pInBuffer = &inBuffer; 3875 } 3876 3877 if (pInBuffer->frameCount == 0) { 3878 break; 3879 } 3880 3881 if (mOutBuffer.frameCount == 0) { 3882 mOutBuffer.frameCount = pInBuffer->frameCount; 3883 nsecs_t startTime = systemTime(); 3884 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3885 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3886 outputBufferFull = true; 3887 break; 3888 } 3889 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3890 if (waitTimeLeftMs >= waitTimeMs) { 3891 waitTimeLeftMs -= waitTimeMs; 3892 } else { 3893 waitTimeLeftMs = 0; 3894 } 3895 } 3896 3897 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3898 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3899 mCblk->stepUser(outFrames); 3900 pInBuffer->frameCount -= outFrames; 3901 pInBuffer->i16 += outFrames * channelCount; 3902 mOutBuffer.frameCount -= outFrames; 3903 mOutBuffer.i16 += outFrames * channelCount; 3904 3905 if (pInBuffer->frameCount == 0) { 3906 if (mBufferQueue.size()) { 3907 mBufferQueue.removeAt(0); 3908 delete [] pInBuffer->mBuffer; 3909 delete pInBuffer; 3910 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3911 } else { 3912 break; 3913 } 3914 } 3915 } 3916 3917 // If we could not write all frames, allocate a buffer and queue it for next time. 3918 if (inBuffer.frameCount) { 3919 sp<ThreadBase> thread = mThread.promote(); 3920 if (thread != 0 && !thread->standby()) { 3921 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3922 pInBuffer = new Buffer; 3923 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3924 pInBuffer->frameCount = inBuffer.frameCount; 3925 pInBuffer->i16 = pInBuffer->mBuffer; 3926 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3927 mBufferQueue.add(pInBuffer); 3928 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3929 } else { 3930 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3931 } 3932 } 3933 } 3934 3935 // Calling write() with a 0 length buffer, means that no more data will be written: 3936 // If no more buffers are pending, fill output track buffer to make sure it is started 3937 // by output mixer. 3938 if (frames == 0 && mBufferQueue.size() == 0) { 3939 if (mCblk->user < mCblk->frameCount) { 3940 frames = mCblk->frameCount - mCblk->user; 3941 pInBuffer = new Buffer; 3942 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3943 pInBuffer->frameCount = frames; 3944 pInBuffer->i16 = pInBuffer->mBuffer; 3945 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3946 mBufferQueue.add(pInBuffer); 3947 } else if (mActive) { 3948 stop(); 3949 } 3950 } 3951 3952 return outputBufferFull; 3953} 3954 3955status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3956{ 3957 int active; 3958 status_t result; 3959 audio_track_cblk_t* cblk = mCblk; 3960 uint32_t framesReq = buffer->frameCount; 3961 3962// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3963 buffer->frameCount = 0; 3964 3965 uint32_t framesAvail = cblk->framesAvailable(); 3966 3967 3968 if (framesAvail == 0) { 3969 Mutex::Autolock _l(cblk->lock); 3970 goto start_loop_here; 3971 while (framesAvail == 0) { 3972 active = mActive; 3973 if (CC_UNLIKELY(!active)) { 3974 ALOGV("Not active and NO_MORE_BUFFERS"); 3975 return NO_MORE_BUFFERS; 3976 } 3977 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3978 if (result != NO_ERROR) { 3979 return NO_MORE_BUFFERS; 3980 } 3981 // read the server count again 3982 start_loop_here: 3983 framesAvail = cblk->framesAvailable_l(); 3984 } 3985 } 3986 3987// if (framesAvail < framesReq) { 3988// return NO_MORE_BUFFERS; 3989// } 3990 3991 if (framesReq > framesAvail) { 3992 framesReq = framesAvail; 3993 } 3994 3995 uint32_t u = cblk->user; 3996 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3997 3998 if (u + framesReq > bufferEnd) { 3999 framesReq = bufferEnd - u; 4000 } 4001 4002 buffer->frameCount = framesReq; 4003 buffer->raw = (void *)cblk->buffer(u); 4004 return NO_ERROR; 4005} 4006 4007 4008void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4009{ 4010 size_t size = mBufferQueue.size(); 4011 Buffer *pBuffer; 4012 4013 for (size_t i = 0; i < size; i++) { 4014 pBuffer = mBufferQueue.itemAt(i); 4015 delete [] pBuffer->mBuffer; 4016 delete pBuffer; 4017 } 4018 mBufferQueue.clear(); 4019} 4020 4021// ---------------------------------------------------------------------------- 4022 4023AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4024 : RefBase(), 4025 mAudioFlinger(audioFlinger), 4026 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4027 mPid(pid) 4028{ 4029 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4030} 4031 4032// Client destructor must be called with AudioFlinger::mLock held 4033AudioFlinger::Client::~Client() 4034{ 4035 mAudioFlinger->removeClient_l(mPid); 4036} 4037 4038sp<MemoryDealer> AudioFlinger::Client::heap() const 4039{ 4040 return mMemoryDealer; 4041} 4042 4043// ---------------------------------------------------------------------------- 4044 4045AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4046 const sp<IAudioFlingerClient>& client, 4047 pid_t pid) 4048 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4049{ 4050} 4051 4052AudioFlinger::NotificationClient::~NotificationClient() 4053{ 4054} 4055 4056void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4057{ 4058 sp<NotificationClient> keep(this); 4059 { 4060 mAudioFlinger->removeNotificationClient(mPid); 4061 } 4062} 4063 4064// ---------------------------------------------------------------------------- 4065 4066AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4067 : BnAudioTrack(), 4068 mTrack(track) 4069{ 4070} 4071 4072AudioFlinger::TrackHandle::~TrackHandle() { 4073 // just stop the track on deletion, associated resources 4074 // will be freed from the main thread once all pending buffers have 4075 // been played. Unless it's not in the active track list, in which 4076 // case we free everything now... 4077 mTrack->destroy(); 4078} 4079 4080sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4081 return mTrack->getCblk(); 4082} 4083 4084status_t AudioFlinger::TrackHandle::start() { 4085 return mTrack->start(); 4086} 4087 4088void AudioFlinger::TrackHandle::stop() { 4089 mTrack->stop(); 4090} 4091 4092void AudioFlinger::TrackHandle::flush() { 4093 mTrack->flush(); 4094} 4095 4096void AudioFlinger::TrackHandle::mute(bool e) { 4097 mTrack->mute(e); 4098} 4099 4100void AudioFlinger::TrackHandle::pause() { 4101 mTrack->pause(); 4102} 4103 4104status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4105{ 4106 return mTrack->attachAuxEffect(EffectId); 4107} 4108 4109status_t AudioFlinger::TrackHandle::onTransact( 4110 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4111{ 4112 return BnAudioTrack::onTransact(code, data, reply, flags); 4113} 4114 4115// ---------------------------------------------------------------------------- 4116 4117sp<IAudioRecord> AudioFlinger::openRecord( 4118 pid_t pid, 4119 int input, 4120 uint32_t sampleRate, 4121 audio_format_t format, 4122 uint32_t channelMask, 4123 int frameCount, 4124 uint32_t flags, 4125 int *sessionId, 4126 status_t *status) 4127{ 4128 sp<RecordThread::RecordTrack> recordTrack; 4129 sp<RecordHandle> recordHandle; 4130 sp<Client> client; 4131 wp<Client> wclient; 4132 status_t lStatus; 4133 RecordThread *thread; 4134 size_t inFrameCount; 4135 int lSessionId; 4136 4137 // check calling permissions 4138 if (!recordingAllowed()) { 4139 lStatus = PERMISSION_DENIED; 4140 goto Exit; 4141 } 4142 4143 // add client to list 4144 { // scope for mLock 4145 Mutex::Autolock _l(mLock); 4146 thread = checkRecordThread_l(input); 4147 if (thread == NULL) { 4148 lStatus = BAD_VALUE; 4149 goto Exit; 4150 } 4151 4152 wclient = mClients.valueFor(pid); 4153 if (wclient != NULL) { 4154 client = wclient.promote(); 4155 } else { 4156 client = new Client(this, pid); 4157 mClients.add(pid, client); 4158 } 4159 4160 // If no audio session id is provided, create one here 4161 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4162 lSessionId = *sessionId; 4163 } else { 4164 lSessionId = nextUniqueId(); 4165 if (sessionId != NULL) { 4166 *sessionId = lSessionId; 4167 } 4168 } 4169 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4170 recordTrack = thread->createRecordTrack_l(client, 4171 sampleRate, 4172 format, 4173 channelMask, 4174 frameCount, 4175 flags, 4176 lSessionId, 4177 &lStatus); 4178 } 4179 if (lStatus != NO_ERROR) { 4180 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4181 // destructor is called by the TrackBase destructor with mLock held 4182 client.clear(); 4183 recordTrack.clear(); 4184 goto Exit; 4185 } 4186 4187 // return to handle to client 4188 recordHandle = new RecordHandle(recordTrack); 4189 lStatus = NO_ERROR; 4190 4191Exit: 4192 if (status) { 4193 *status = lStatus; 4194 } 4195 return recordHandle; 4196} 4197 4198// ---------------------------------------------------------------------------- 4199 4200AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4201 : BnAudioRecord(), 4202 mRecordTrack(recordTrack) 4203{ 4204} 4205 4206AudioFlinger::RecordHandle::~RecordHandle() { 4207 stop(); 4208} 4209 4210sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4211 return mRecordTrack->getCblk(); 4212} 4213 4214status_t AudioFlinger::RecordHandle::start() { 4215 ALOGV("RecordHandle::start()"); 4216 return mRecordTrack->start(); 4217} 4218 4219void AudioFlinger::RecordHandle::stop() { 4220 ALOGV("RecordHandle::stop()"); 4221 mRecordTrack->stop(); 4222} 4223 4224status_t AudioFlinger::RecordHandle::onTransact( 4225 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4226{ 4227 return BnAudioRecord::onTransact(code, data, reply, flags); 4228} 4229 4230// ---------------------------------------------------------------------------- 4231 4232AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4233 AudioStreamIn *input, 4234 uint32_t sampleRate, 4235 uint32_t channels, 4236 int id, 4237 uint32_t device) : 4238 ThreadBase(audioFlinger, id, device, RECORD), 4239 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4240 // mRsmpInIndex and mInputBytes set by readInputParameters() 4241 mReqChannelCount(popcount(channels)), 4242 mReqSampleRate(sampleRate) 4243 // mBytesRead is only meaningful while active, and so is cleared in start() 4244 // (but might be better to also clear here for dump?) 4245{ 4246 snprintf(mName, kNameLength, "AudioIn_%d", id); 4247 4248 readInputParameters(); 4249} 4250 4251 4252AudioFlinger::RecordThread::~RecordThread() 4253{ 4254 delete[] mRsmpInBuffer; 4255 delete mResampler; 4256 delete[] mRsmpOutBuffer; 4257} 4258 4259void AudioFlinger::RecordThread::onFirstRef() 4260{ 4261 run(mName, PRIORITY_URGENT_AUDIO); 4262} 4263 4264status_t AudioFlinger::RecordThread::readyToRun() 4265{ 4266 status_t status = initCheck(); 4267 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4268 return status; 4269} 4270 4271bool AudioFlinger::RecordThread::threadLoop() 4272{ 4273 AudioBufferProvider::Buffer buffer; 4274 sp<RecordTrack> activeTrack; 4275 Vector< sp<EffectChain> > effectChains; 4276 4277 nsecs_t lastWarning = 0; 4278 4279 acquireWakeLock(); 4280 4281 // start recording 4282 while (!exitPending()) { 4283 4284 processConfigEvents(); 4285 4286 { // scope for mLock 4287 Mutex::Autolock _l(mLock); 4288 checkForNewParameters_l(); 4289 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4290 if (!mStandby) { 4291 mInput->stream->common.standby(&mInput->stream->common); 4292 mStandby = true; 4293 } 4294 4295 if (exitPending()) break; 4296 4297 releaseWakeLock_l(); 4298 ALOGV("RecordThread: loop stopping"); 4299 // go to sleep 4300 mWaitWorkCV.wait(mLock); 4301 ALOGV("RecordThread: loop starting"); 4302 acquireWakeLock_l(); 4303 continue; 4304 } 4305 if (mActiveTrack != 0) { 4306 if (mActiveTrack->mState == TrackBase::PAUSING) { 4307 if (!mStandby) { 4308 mInput->stream->common.standby(&mInput->stream->common); 4309 mStandby = true; 4310 } 4311 mActiveTrack.clear(); 4312 mStartStopCond.broadcast(); 4313 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4314 if (mReqChannelCount != mActiveTrack->channelCount()) { 4315 mActiveTrack.clear(); 4316 mStartStopCond.broadcast(); 4317 } else if (mBytesRead != 0) { 4318 // record start succeeds only if first read from audio input 4319 // succeeds 4320 if (mBytesRead > 0) { 4321 mActiveTrack->mState = TrackBase::ACTIVE; 4322 } else { 4323 mActiveTrack.clear(); 4324 } 4325 mStartStopCond.broadcast(); 4326 } 4327 mStandby = false; 4328 } 4329 } 4330 lockEffectChains_l(effectChains); 4331 } 4332 4333 if (mActiveTrack != 0) { 4334 if (mActiveTrack->mState != TrackBase::ACTIVE && 4335 mActiveTrack->mState != TrackBase::RESUMING) { 4336 unlockEffectChains(effectChains); 4337 usleep(kRecordThreadSleepUs); 4338 continue; 4339 } 4340 for (size_t i = 0; i < effectChains.size(); i ++) { 4341 effectChains[i]->process_l(); 4342 } 4343 4344 buffer.frameCount = mFrameCount; 4345 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4346 size_t framesOut = buffer.frameCount; 4347 if (mResampler == NULL) { 4348 // no resampling 4349 while (framesOut) { 4350 size_t framesIn = mFrameCount - mRsmpInIndex; 4351 if (framesIn) { 4352 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4353 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4354 if (framesIn > framesOut) 4355 framesIn = framesOut; 4356 mRsmpInIndex += framesIn; 4357 framesOut -= framesIn; 4358 if ((int)mChannelCount == mReqChannelCount || 4359 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4360 memcpy(dst, src, framesIn * mFrameSize); 4361 } else { 4362 int16_t *src16 = (int16_t *)src; 4363 int16_t *dst16 = (int16_t *)dst; 4364 if (mChannelCount == 1) { 4365 while (framesIn--) { 4366 *dst16++ = *src16; 4367 *dst16++ = *src16++; 4368 } 4369 } else { 4370 while (framesIn--) { 4371 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4372 src16 += 2; 4373 } 4374 } 4375 } 4376 } 4377 if (framesOut && mFrameCount == mRsmpInIndex) { 4378 if (framesOut == mFrameCount && 4379 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4380 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4381 framesOut = 0; 4382 } else { 4383 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4384 mRsmpInIndex = 0; 4385 } 4386 if (mBytesRead < 0) { 4387 ALOGE("Error reading audio input"); 4388 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4389 // Force input into standby so that it tries to 4390 // recover at next read attempt 4391 mInput->stream->common.standby(&mInput->stream->common); 4392 usleep(kRecordThreadSleepUs); 4393 } 4394 mRsmpInIndex = mFrameCount; 4395 framesOut = 0; 4396 buffer.frameCount = 0; 4397 } 4398 } 4399 } 4400 } else { 4401 // resampling 4402 4403 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4404 // alter output frame count as if we were expecting stereo samples 4405 if (mChannelCount == 1 && mReqChannelCount == 1) { 4406 framesOut >>= 1; 4407 } 4408 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4409 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4410 // are 32 bit aligned which should be always true. 4411 if (mChannelCount == 2 && mReqChannelCount == 1) { 4412 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4413 // the resampler always outputs stereo samples: do post stereo to mono conversion 4414 int16_t *src = (int16_t *)mRsmpOutBuffer; 4415 int16_t *dst = buffer.i16; 4416 while (framesOut--) { 4417 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4418 src += 2; 4419 } 4420 } else { 4421 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4422 } 4423 4424 } 4425 mActiveTrack->releaseBuffer(&buffer); 4426 mActiveTrack->overflow(); 4427 } 4428 // client isn't retrieving buffers fast enough 4429 else { 4430 if (!mActiveTrack->setOverflow()) { 4431 nsecs_t now = systemTime(); 4432 if ((now - lastWarning) > kWarningThrottleNs) { 4433 ALOGW("RecordThread: buffer overflow"); 4434 lastWarning = now; 4435 } 4436 } 4437 // Release the processor for a while before asking for a new buffer. 4438 // This will give the application more chance to read from the buffer and 4439 // clear the overflow. 4440 usleep(kRecordThreadSleepUs); 4441 } 4442 } 4443 // enable changes in effect chain 4444 unlockEffectChains(effectChains); 4445 effectChains.clear(); 4446 } 4447 4448 if (!mStandby) { 4449 mInput->stream->common.standby(&mInput->stream->common); 4450 } 4451 mActiveTrack.clear(); 4452 4453 mStartStopCond.broadcast(); 4454 4455 releaseWakeLock(); 4456 4457 ALOGV("RecordThread %p exiting", this); 4458 return false; 4459} 4460 4461 4462sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4463 const sp<AudioFlinger::Client>& client, 4464 uint32_t sampleRate, 4465 audio_format_t format, 4466 int channelMask, 4467 int frameCount, 4468 uint32_t flags, 4469 int sessionId, 4470 status_t *status) 4471{ 4472 sp<RecordTrack> track; 4473 status_t lStatus; 4474 4475 lStatus = initCheck(); 4476 if (lStatus != NO_ERROR) { 4477 ALOGE("Audio driver not initialized."); 4478 goto Exit; 4479 } 4480 4481 { // scope for mLock 4482 Mutex::Autolock _l(mLock); 4483 4484 track = new RecordTrack(this, client, sampleRate, 4485 format, channelMask, frameCount, flags, sessionId); 4486 4487 if (track->getCblk() == 0) { 4488 lStatus = NO_MEMORY; 4489 goto Exit; 4490 } 4491 4492 mTrack = track.get(); 4493 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4494 bool suspend = audio_is_bluetooth_sco_device( 4495 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4496 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4497 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4498 } 4499 lStatus = NO_ERROR; 4500 4501Exit: 4502 if (status) { 4503 *status = lStatus; 4504 } 4505 return track; 4506} 4507 4508status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4509{ 4510 ALOGV("RecordThread::start"); 4511 sp <ThreadBase> strongMe = this; 4512 status_t status = NO_ERROR; 4513 { 4514 AutoMutex lock(mLock); 4515 if (mActiveTrack != 0) { 4516 if (recordTrack != mActiveTrack.get()) { 4517 status = -EBUSY; 4518 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4519 mActiveTrack->mState = TrackBase::ACTIVE; 4520 } 4521 return status; 4522 } 4523 4524 recordTrack->mState = TrackBase::IDLE; 4525 mActiveTrack = recordTrack; 4526 mLock.unlock(); 4527 status_t status = AudioSystem::startInput(mId); 4528 mLock.lock(); 4529 if (status != NO_ERROR) { 4530 mActiveTrack.clear(); 4531 return status; 4532 } 4533 mRsmpInIndex = mFrameCount; 4534 mBytesRead = 0; 4535 if (mResampler != NULL) { 4536 mResampler->reset(); 4537 } 4538 mActiveTrack->mState = TrackBase::RESUMING; 4539 // signal thread to start 4540 ALOGV("Signal record thread"); 4541 mWaitWorkCV.signal(); 4542 // do not wait for mStartStopCond if exiting 4543 if (mExiting) { 4544 mActiveTrack.clear(); 4545 status = INVALID_OPERATION; 4546 goto startError; 4547 } 4548 mStartStopCond.wait(mLock); 4549 if (mActiveTrack == 0) { 4550 ALOGV("Record failed to start"); 4551 status = BAD_VALUE; 4552 goto startError; 4553 } 4554 ALOGV("Record started OK"); 4555 return status; 4556 } 4557startError: 4558 AudioSystem::stopInput(mId); 4559 return status; 4560} 4561 4562void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4563 ALOGV("RecordThread::stop"); 4564 sp <ThreadBase> strongMe = this; 4565 { 4566 AutoMutex lock(mLock); 4567 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4568 mActiveTrack->mState = TrackBase::PAUSING; 4569 // do not wait for mStartStopCond if exiting 4570 if (mExiting) { 4571 return; 4572 } 4573 mStartStopCond.wait(mLock); 4574 // if we have been restarted, recordTrack == mActiveTrack.get() here 4575 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4576 mLock.unlock(); 4577 AudioSystem::stopInput(mId); 4578 mLock.lock(); 4579 ALOGV("Record stopped OK"); 4580 } 4581 } 4582 } 4583} 4584 4585status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4586{ 4587 const size_t SIZE = 256; 4588 char buffer[SIZE]; 4589 String8 result; 4590 4591 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4592 result.append(buffer); 4593 4594 if (mActiveTrack != 0) { 4595 result.append("Active Track:\n"); 4596 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4597 mActiveTrack->dump(buffer, SIZE); 4598 result.append(buffer); 4599 4600 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4601 result.append(buffer); 4602 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4603 result.append(buffer); 4604 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4605 result.append(buffer); 4606 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4607 result.append(buffer); 4608 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4609 result.append(buffer); 4610 4611 4612 } else { 4613 result.append("No record client\n"); 4614 } 4615 write(fd, result.string(), result.size()); 4616 4617 dumpBase(fd, args); 4618 dumpEffectChains(fd, args); 4619 4620 return NO_ERROR; 4621} 4622 4623status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4624{ 4625 size_t framesReq = buffer->frameCount; 4626 size_t framesReady = mFrameCount - mRsmpInIndex; 4627 int channelCount; 4628 4629 if (framesReady == 0) { 4630 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4631 if (mBytesRead < 0) { 4632 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4633 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4634 // Force input into standby so that it tries to 4635 // recover at next read attempt 4636 mInput->stream->common.standby(&mInput->stream->common); 4637 usleep(kRecordThreadSleepUs); 4638 } 4639 buffer->raw = NULL; 4640 buffer->frameCount = 0; 4641 return NOT_ENOUGH_DATA; 4642 } 4643 mRsmpInIndex = 0; 4644 framesReady = mFrameCount; 4645 } 4646 4647 if (framesReq > framesReady) { 4648 framesReq = framesReady; 4649 } 4650 4651 if (mChannelCount == 1 && mReqChannelCount == 2) { 4652 channelCount = 1; 4653 } else { 4654 channelCount = 2; 4655 } 4656 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4657 buffer->frameCount = framesReq; 4658 return NO_ERROR; 4659} 4660 4661void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4662{ 4663 mRsmpInIndex += buffer->frameCount; 4664 buffer->frameCount = 0; 4665} 4666 4667bool AudioFlinger::RecordThread::checkForNewParameters_l() 4668{ 4669 bool reconfig = false; 4670 4671 while (!mNewParameters.isEmpty()) { 4672 status_t status = NO_ERROR; 4673 String8 keyValuePair = mNewParameters[0]; 4674 AudioParameter param = AudioParameter(keyValuePair); 4675 int value; 4676 audio_format_t reqFormat = mFormat; 4677 int reqSamplingRate = mReqSampleRate; 4678 int reqChannelCount = mReqChannelCount; 4679 4680 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4681 reqSamplingRate = value; 4682 reconfig = true; 4683 } 4684 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4685 reqFormat = (audio_format_t) value; 4686 reconfig = true; 4687 } 4688 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4689 reqChannelCount = popcount(value); 4690 reconfig = true; 4691 } 4692 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4693 // do not accept frame count changes if tracks are open as the track buffer 4694 // size depends on frame count and correct behavior would not be garantied 4695 // if frame count is changed after track creation 4696 if (mActiveTrack != 0) { 4697 status = INVALID_OPERATION; 4698 } else { 4699 reconfig = true; 4700 } 4701 } 4702 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4703 // forward device change to effects that have requested to be 4704 // aware of attached audio device. 4705 for (size_t i = 0; i < mEffectChains.size(); i++) { 4706 mEffectChains[i]->setDevice_l(value); 4707 } 4708 // store input device and output device but do not forward output device to audio HAL. 4709 // Note that status is ignored by the caller for output device 4710 // (see AudioFlinger::setParameters() 4711 if (value & AUDIO_DEVICE_OUT_ALL) { 4712 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4713 status = BAD_VALUE; 4714 } else { 4715 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4716 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4717 if (mTrack != NULL) { 4718 bool suspend = audio_is_bluetooth_sco_device( 4719 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4720 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4721 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4722 } 4723 } 4724 mDevice |= (uint32_t)value; 4725 } 4726 if (status == NO_ERROR) { 4727 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4728 if (status == INVALID_OPERATION) { 4729 mInput->stream->common.standby(&mInput->stream->common); 4730 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4731 } 4732 if (reconfig) { 4733 if (status == BAD_VALUE && 4734 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4735 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4736 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4737 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4738 (reqChannelCount < 3)) { 4739 status = NO_ERROR; 4740 } 4741 if (status == NO_ERROR) { 4742 readInputParameters(); 4743 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4744 } 4745 } 4746 } 4747 4748 mNewParameters.removeAt(0); 4749 4750 mParamStatus = status; 4751 mParamCond.signal(); 4752 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4753 // already timed out waiting for the status and will never signal the condition. 4754 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4755 } 4756 return reconfig; 4757} 4758 4759String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4760{ 4761 char *s; 4762 String8 out_s8 = String8(); 4763 4764 Mutex::Autolock _l(mLock); 4765 if (initCheck() != NO_ERROR) { 4766 return out_s8; 4767 } 4768 4769 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4770 out_s8 = String8(s); 4771 free(s); 4772 return out_s8; 4773} 4774 4775void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4776 AudioSystem::OutputDescriptor desc; 4777 void *param2 = NULL; 4778 4779 switch (event) { 4780 case AudioSystem::INPUT_OPENED: 4781 case AudioSystem::INPUT_CONFIG_CHANGED: 4782 desc.channels = mChannelMask; 4783 desc.samplingRate = mSampleRate; 4784 desc.format = mFormat; 4785 desc.frameCount = mFrameCount; 4786 desc.latency = 0; 4787 param2 = &desc; 4788 break; 4789 4790 case AudioSystem::INPUT_CLOSED: 4791 default: 4792 break; 4793 } 4794 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4795} 4796 4797void AudioFlinger::RecordThread::readInputParameters() 4798{ 4799 delete mRsmpInBuffer; 4800 // mRsmpInBuffer is always assigned a new[] below 4801 delete mRsmpOutBuffer; 4802 mRsmpOutBuffer = NULL; 4803 delete mResampler; 4804 mResampler = NULL; 4805 4806 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4807 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4808 mChannelCount = (uint16_t)popcount(mChannelMask); 4809 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4810 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4811 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4812 mFrameCount = mInputBytes / mFrameSize; 4813 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4814 4815 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4816 { 4817 int channelCount; 4818 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4819 // stereo to mono post process as the resampler always outputs stereo. 4820 if (mChannelCount == 1 && mReqChannelCount == 2) { 4821 channelCount = 1; 4822 } else { 4823 channelCount = 2; 4824 } 4825 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4826 mResampler->setSampleRate(mSampleRate); 4827 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4828 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4829 4830 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4831 if (mChannelCount == 1 && mReqChannelCount == 1) { 4832 mFrameCount >>= 1; 4833 } 4834 4835 } 4836 mRsmpInIndex = mFrameCount; 4837} 4838 4839unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4840{ 4841 Mutex::Autolock _l(mLock); 4842 if (initCheck() != NO_ERROR) { 4843 return 0; 4844 } 4845 4846 return mInput->stream->get_input_frames_lost(mInput->stream); 4847} 4848 4849uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4850{ 4851 Mutex::Autolock _l(mLock); 4852 uint32_t result = 0; 4853 if (getEffectChain_l(sessionId) != 0) { 4854 result = EFFECT_SESSION; 4855 } 4856 4857 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4858 result |= TRACK_SESSION; 4859 } 4860 4861 return result; 4862} 4863 4864AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4865{ 4866 Mutex::Autolock _l(mLock); 4867 return mTrack; 4868} 4869 4870AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4871{ 4872 Mutex::Autolock _l(mLock); 4873 return mInput; 4874} 4875 4876AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4877{ 4878 Mutex::Autolock _l(mLock); 4879 AudioStreamIn *input = mInput; 4880 mInput = NULL; 4881 return input; 4882} 4883 4884// this method must always be called either with ThreadBase mLock held or inside the thread loop 4885audio_stream_t* AudioFlinger::RecordThread::stream() 4886{ 4887 if (mInput == NULL) { 4888 return NULL; 4889 } 4890 return &mInput->stream->common; 4891} 4892 4893 4894// ---------------------------------------------------------------------------- 4895 4896int AudioFlinger::openOutput(uint32_t *pDevices, 4897 uint32_t *pSamplingRate, 4898 audio_format_t *pFormat, 4899 uint32_t *pChannels, 4900 uint32_t *pLatencyMs, 4901 uint32_t flags) 4902{ 4903 status_t status; 4904 PlaybackThread *thread = NULL; 4905 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4906 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4907 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4908 uint32_t channels = pChannels ? *pChannels : 0; 4909 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4910 audio_stream_out_t *outStream; 4911 audio_hw_device_t *outHwDev; 4912 4913 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4914 pDevices ? *pDevices : 0, 4915 samplingRate, 4916 format, 4917 channels, 4918 flags); 4919 4920 if (pDevices == NULL || *pDevices == 0) { 4921 return 0; 4922 } 4923 4924 Mutex::Autolock _l(mLock); 4925 4926 outHwDev = findSuitableHwDev_l(*pDevices); 4927 if (outHwDev == NULL) 4928 return 0; 4929 4930 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4931 &channels, &samplingRate, &outStream); 4932 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4933 outStream, 4934 samplingRate, 4935 format, 4936 channels, 4937 status); 4938 4939 mHardwareStatus = AUDIO_HW_IDLE; 4940 if (outStream != NULL) { 4941 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4942 int id = nextUniqueId(); 4943 4944 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4945 (format != AUDIO_FORMAT_PCM_16_BIT) || 4946 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4947 thread = new DirectOutputThread(this, output, id, *pDevices); 4948 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4949 } else { 4950 thread = new MixerThread(this, output, id, *pDevices); 4951 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4952 } 4953 mPlaybackThreads.add(id, thread); 4954 4955 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 4956 if (pFormat != NULL) *pFormat = format; 4957 if (pChannels != NULL) *pChannels = channels; 4958 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 4959 4960 // notify client processes of the new output creation 4961 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4962 return id; 4963 } 4964 4965 return 0; 4966} 4967 4968int AudioFlinger::openDuplicateOutput(int output1, int output2) 4969{ 4970 Mutex::Autolock _l(mLock); 4971 MixerThread *thread1 = checkMixerThread_l(output1); 4972 MixerThread *thread2 = checkMixerThread_l(output2); 4973 4974 if (thread1 == NULL || thread2 == NULL) { 4975 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4976 return 0; 4977 } 4978 4979 int id = nextUniqueId(); 4980 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4981 thread->addOutputTrack(thread2); 4982 mPlaybackThreads.add(id, thread); 4983 // notify client processes of the new output creation 4984 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4985 return id; 4986} 4987 4988status_t AudioFlinger::closeOutput(int output) 4989{ 4990 // keep strong reference on the playback thread so that 4991 // it is not destroyed while exit() is executed 4992 sp <PlaybackThread> thread; 4993 { 4994 Mutex::Autolock _l(mLock); 4995 thread = checkPlaybackThread_l(output); 4996 if (thread == NULL) { 4997 return BAD_VALUE; 4998 } 4999 5000 ALOGV("closeOutput() %d", output); 5001 5002 if (thread->type() == ThreadBase::MIXER) { 5003 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5004 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5005 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5006 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5007 } 5008 } 5009 } 5010 void *param2 = NULL; 5011 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5012 mPlaybackThreads.removeItem(output); 5013 } 5014 thread->exit(); 5015 5016 if (thread->type() != ThreadBase::DUPLICATING) { 5017 AudioStreamOut *out = thread->clearOutput(); 5018 assert(out != NULL); 5019 // from now on thread->mOutput is NULL 5020 out->hwDev->close_output_stream(out->hwDev, out->stream); 5021 delete out; 5022 } 5023 return NO_ERROR; 5024} 5025 5026status_t AudioFlinger::suspendOutput(int output) 5027{ 5028 Mutex::Autolock _l(mLock); 5029 PlaybackThread *thread = checkPlaybackThread_l(output); 5030 5031 if (thread == NULL) { 5032 return BAD_VALUE; 5033 } 5034 5035 ALOGV("suspendOutput() %d", output); 5036 thread->suspend(); 5037 5038 return NO_ERROR; 5039} 5040 5041status_t AudioFlinger::restoreOutput(int output) 5042{ 5043 Mutex::Autolock _l(mLock); 5044 PlaybackThread *thread = checkPlaybackThread_l(output); 5045 5046 if (thread == NULL) { 5047 return BAD_VALUE; 5048 } 5049 5050 ALOGV("restoreOutput() %d", output); 5051 5052 thread->restore(); 5053 5054 return NO_ERROR; 5055} 5056 5057int AudioFlinger::openInput(uint32_t *pDevices, 5058 uint32_t *pSamplingRate, 5059 audio_format_t *pFormat, 5060 uint32_t *pChannels, 5061 audio_in_acoustics_t acoustics) 5062{ 5063 status_t status; 5064 RecordThread *thread = NULL; 5065 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5066 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5067 uint32_t channels = pChannels ? *pChannels : 0; 5068 uint32_t reqSamplingRate = samplingRate; 5069 audio_format_t reqFormat = format; 5070 uint32_t reqChannels = channels; 5071 audio_stream_in_t *inStream; 5072 audio_hw_device_t *inHwDev; 5073 5074 if (pDevices == NULL || *pDevices == 0) { 5075 return 0; 5076 } 5077 5078 Mutex::Autolock _l(mLock); 5079 5080 inHwDev = findSuitableHwDev_l(*pDevices); 5081 if (inHwDev == NULL) 5082 return 0; 5083 5084 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5085 &channels, &samplingRate, 5086 acoustics, 5087 &inStream); 5088 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5089 inStream, 5090 samplingRate, 5091 format, 5092 channels, 5093 acoustics, 5094 status); 5095 5096 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5097 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5098 // or stereo to mono conversions on 16 bit PCM inputs. 5099 if (inStream == NULL && status == BAD_VALUE && 5100 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5101 (samplingRate <= 2 * reqSamplingRate) && 5102 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5103 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5104 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5105 &channels, &samplingRate, 5106 acoustics, 5107 &inStream); 5108 } 5109 5110 if (inStream != NULL) { 5111 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5112 5113 int id = nextUniqueId(); 5114 // Start record thread 5115 // RecorThread require both input and output device indication to forward to audio 5116 // pre processing modules 5117 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5118 thread = new RecordThread(this, 5119 input, 5120 reqSamplingRate, 5121 reqChannels, 5122 id, 5123 device); 5124 mRecordThreads.add(id, thread); 5125 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5126 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5127 if (pFormat != NULL) *pFormat = format; 5128 if (pChannels != NULL) *pChannels = reqChannels; 5129 5130 input->stream->common.standby(&input->stream->common); 5131 5132 // notify client processes of the new input creation 5133 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5134 return id; 5135 } 5136 5137 return 0; 5138} 5139 5140status_t AudioFlinger::closeInput(int input) 5141{ 5142 // keep strong reference on the record thread so that 5143 // it is not destroyed while exit() is executed 5144 sp <RecordThread> thread; 5145 { 5146 Mutex::Autolock _l(mLock); 5147 thread = checkRecordThread_l(input); 5148 if (thread == NULL) { 5149 return BAD_VALUE; 5150 } 5151 5152 ALOGV("closeInput() %d", input); 5153 void *param2 = NULL; 5154 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5155 mRecordThreads.removeItem(input); 5156 } 5157 thread->exit(); 5158 5159 AudioStreamIn *in = thread->clearInput(); 5160 assert(in != NULL); 5161 // from now on thread->mInput is NULL 5162 in->hwDev->close_input_stream(in->hwDev, in->stream); 5163 delete in; 5164 5165 return NO_ERROR; 5166} 5167 5168status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5169{ 5170 Mutex::Autolock _l(mLock); 5171 MixerThread *dstThread = checkMixerThread_l(output); 5172 if (dstThread == NULL) { 5173 ALOGW("setStreamOutput() bad output id %d", output); 5174 return BAD_VALUE; 5175 } 5176 5177 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5178 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5179 5180 dstThread->setStreamValid(stream, true); 5181 5182 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5183 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5184 if (thread != dstThread && 5185 thread->type() != ThreadBase::DIRECT) { 5186 MixerThread *srcThread = (MixerThread *)thread; 5187 srcThread->setStreamValid(stream, false); 5188 srcThread->invalidateTracks(stream); 5189 } 5190 } 5191 5192 return NO_ERROR; 5193} 5194 5195 5196int AudioFlinger::newAudioSessionId() 5197{ 5198 return nextUniqueId(); 5199} 5200 5201void AudioFlinger::acquireAudioSessionId(int audioSession) 5202{ 5203 Mutex::Autolock _l(mLock); 5204 pid_t caller = IPCThreadState::self()->getCallingPid(); 5205 ALOGV("acquiring %d from %d", audioSession, caller); 5206 int num = mAudioSessionRefs.size(); 5207 for (int i = 0; i< num; i++) { 5208 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5209 if (ref->sessionid == audioSession && ref->pid == caller) { 5210 ref->cnt++; 5211 ALOGV(" incremented refcount to %d", ref->cnt); 5212 return; 5213 } 5214 } 5215 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5216 ALOGV(" added new entry for %d", audioSession); 5217} 5218 5219void AudioFlinger::releaseAudioSessionId(int audioSession) 5220{ 5221 Mutex::Autolock _l(mLock); 5222 pid_t caller = IPCThreadState::self()->getCallingPid(); 5223 ALOGV("releasing %d from %d", audioSession, caller); 5224 int num = mAudioSessionRefs.size(); 5225 for (int i = 0; i< num; i++) { 5226 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5227 if (ref->sessionid == audioSession && ref->pid == caller) { 5228 ref->cnt--; 5229 ALOGV(" decremented refcount to %d", ref->cnt); 5230 if (ref->cnt == 0) { 5231 mAudioSessionRefs.removeAt(i); 5232 delete ref; 5233 purgeStaleEffects_l(); 5234 } 5235 return; 5236 } 5237 } 5238 ALOGW("session id %d not found for pid %d", audioSession, caller); 5239} 5240 5241void AudioFlinger::purgeStaleEffects_l() { 5242 5243 ALOGV("purging stale effects"); 5244 5245 Vector< sp<EffectChain> > chains; 5246 5247 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5248 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5249 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5250 sp<EffectChain> ec = t->mEffectChains[j]; 5251 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5252 chains.push(ec); 5253 } 5254 } 5255 } 5256 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5257 sp<RecordThread> t = mRecordThreads.valueAt(i); 5258 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5259 sp<EffectChain> ec = t->mEffectChains[j]; 5260 chains.push(ec); 5261 } 5262 } 5263 5264 for (size_t i = 0; i < chains.size(); i++) { 5265 sp<EffectChain> ec = chains[i]; 5266 int sessionid = ec->sessionId(); 5267 sp<ThreadBase> t = ec->mThread.promote(); 5268 if (t == 0) { 5269 continue; 5270 } 5271 size_t numsessionrefs = mAudioSessionRefs.size(); 5272 bool found = false; 5273 for (size_t k = 0; k < numsessionrefs; k++) { 5274 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5275 if (ref->sessionid == sessionid) { 5276 ALOGV(" session %d still exists for %d with %d refs", 5277 sessionid, ref->pid, ref->cnt); 5278 found = true; 5279 break; 5280 } 5281 } 5282 if (!found) { 5283 // remove all effects from the chain 5284 while (ec->mEffects.size()) { 5285 sp<EffectModule> effect = ec->mEffects[0]; 5286 effect->unPin(); 5287 Mutex::Autolock _l (t->mLock); 5288 t->removeEffect_l(effect); 5289 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5290 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5291 if (handle != 0) { 5292 handle->mEffect.clear(); 5293 if (handle->mHasControl && handle->mEnabled) { 5294 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5295 } 5296 } 5297 } 5298 AudioSystem::unregisterEffect(effect->id()); 5299 } 5300 } 5301 } 5302 return; 5303} 5304 5305// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5306AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5307{ 5308 PlaybackThread *thread = NULL; 5309 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5310 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5311 } 5312 return thread; 5313} 5314 5315// checkMixerThread_l() must be called with AudioFlinger::mLock held 5316AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5317{ 5318 PlaybackThread *thread = checkPlaybackThread_l(output); 5319 if (thread != NULL) { 5320 if (thread->type() == ThreadBase::DIRECT) { 5321 thread = NULL; 5322 } 5323 } 5324 return (MixerThread *)thread; 5325} 5326 5327// checkRecordThread_l() must be called with AudioFlinger::mLock held 5328AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5329{ 5330 RecordThread *thread = NULL; 5331 if (mRecordThreads.indexOfKey(input) >= 0) { 5332 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5333 } 5334 return thread; 5335} 5336 5337uint32_t AudioFlinger::nextUniqueId() 5338{ 5339 return android_atomic_inc(&mNextUniqueId); 5340} 5341 5342AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5343{ 5344 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5345 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5346 AudioStreamOut *output = thread->getOutput(); 5347 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5348 return thread; 5349 } 5350 } 5351 return NULL; 5352} 5353 5354uint32_t AudioFlinger::primaryOutputDevice_l() 5355{ 5356 PlaybackThread *thread = primaryPlaybackThread_l(); 5357 5358 if (thread == NULL) { 5359 return 0; 5360 } 5361 5362 return thread->device(); 5363} 5364 5365 5366// ---------------------------------------------------------------------------- 5367// Effect management 5368// ---------------------------------------------------------------------------- 5369 5370 5371status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5372{ 5373 Mutex::Autolock _l(mLock); 5374 return EffectQueryNumberEffects(numEffects); 5375} 5376 5377status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5378{ 5379 Mutex::Autolock _l(mLock); 5380 return EffectQueryEffect(index, descriptor); 5381} 5382 5383status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5384 effect_descriptor_t *descriptor) const 5385{ 5386 Mutex::Autolock _l(mLock); 5387 return EffectGetDescriptor(pUuid, descriptor); 5388} 5389 5390 5391sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5392 effect_descriptor_t *pDesc, 5393 const sp<IEffectClient>& effectClient, 5394 int32_t priority, 5395 int io, 5396 int sessionId, 5397 status_t *status, 5398 int *id, 5399 int *enabled) 5400{ 5401 status_t lStatus = NO_ERROR; 5402 sp<EffectHandle> handle; 5403 effect_descriptor_t desc; 5404 sp<Client> client; 5405 wp<Client> wclient; 5406 5407 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5408 pid, effectClient.get(), priority, sessionId, io); 5409 5410 if (pDesc == NULL) { 5411 lStatus = BAD_VALUE; 5412 goto Exit; 5413 } 5414 5415 // check audio settings permission for global effects 5416 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5417 lStatus = PERMISSION_DENIED; 5418 goto Exit; 5419 } 5420 5421 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5422 // that can only be created by audio policy manager (running in same process) 5423 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5424 lStatus = PERMISSION_DENIED; 5425 goto Exit; 5426 } 5427 5428 if (io == 0) { 5429 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5430 // output must be specified by AudioPolicyManager when using session 5431 // AUDIO_SESSION_OUTPUT_STAGE 5432 lStatus = BAD_VALUE; 5433 goto Exit; 5434 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5435 // if the output returned by getOutputForEffect() is removed before we lock the 5436 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5437 // and we will exit safely 5438 io = AudioSystem::getOutputForEffect(&desc); 5439 } 5440 } 5441 5442 { 5443 Mutex::Autolock _l(mLock); 5444 5445 5446 if (!EffectIsNullUuid(&pDesc->uuid)) { 5447 // if uuid is specified, request effect descriptor 5448 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5449 if (lStatus < 0) { 5450 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5451 goto Exit; 5452 } 5453 } else { 5454 // if uuid is not specified, look for an available implementation 5455 // of the required type in effect factory 5456 if (EffectIsNullUuid(&pDesc->type)) { 5457 ALOGW("createEffect() no effect type"); 5458 lStatus = BAD_VALUE; 5459 goto Exit; 5460 } 5461 uint32_t numEffects = 0; 5462 effect_descriptor_t d; 5463 d.flags = 0; // prevent compiler warning 5464 bool found = false; 5465 5466 lStatus = EffectQueryNumberEffects(&numEffects); 5467 if (lStatus < 0) { 5468 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5469 goto Exit; 5470 } 5471 for (uint32_t i = 0; i < numEffects; i++) { 5472 lStatus = EffectQueryEffect(i, &desc); 5473 if (lStatus < 0) { 5474 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5475 continue; 5476 } 5477 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5478 // If matching type found save effect descriptor. If the session is 5479 // 0 and the effect is not auxiliary, continue enumeration in case 5480 // an auxiliary version of this effect type is available 5481 found = true; 5482 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5483 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5484 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5485 break; 5486 } 5487 } 5488 } 5489 if (!found) { 5490 lStatus = BAD_VALUE; 5491 ALOGW("createEffect() effect not found"); 5492 goto Exit; 5493 } 5494 // For same effect type, chose auxiliary version over insert version if 5495 // connect to output mix (Compliance to OpenSL ES) 5496 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5497 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5498 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5499 } 5500 } 5501 5502 // Do not allow auxiliary effects on a session different from 0 (output mix) 5503 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5504 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5505 lStatus = INVALID_OPERATION; 5506 goto Exit; 5507 } 5508 5509 // check recording permission for visualizer 5510 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5511 !recordingAllowed()) { 5512 lStatus = PERMISSION_DENIED; 5513 goto Exit; 5514 } 5515 5516 // return effect descriptor 5517 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5518 5519 // If output is not specified try to find a matching audio session ID in one of the 5520 // output threads. 5521 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5522 // because of code checking output when entering the function. 5523 // Note: io is never 0 when creating an effect on an input 5524 if (io == 0) { 5525 // look for the thread where the specified audio session is present 5526 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5527 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5528 io = mPlaybackThreads.keyAt(i); 5529 break; 5530 } 5531 } 5532 if (io == 0) { 5533 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5534 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5535 io = mRecordThreads.keyAt(i); 5536 break; 5537 } 5538 } 5539 } 5540 // If no output thread contains the requested session ID, default to 5541 // first output. The effect chain will be moved to the correct output 5542 // thread when a track with the same session ID is created 5543 if (io == 0 && mPlaybackThreads.size()) { 5544 io = mPlaybackThreads.keyAt(0); 5545 } 5546 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5547 } 5548 ThreadBase *thread = checkRecordThread_l(io); 5549 if (thread == NULL) { 5550 thread = checkPlaybackThread_l(io); 5551 if (thread == NULL) { 5552 ALOGE("createEffect() unknown output thread"); 5553 lStatus = BAD_VALUE; 5554 goto Exit; 5555 } 5556 } 5557 5558 wclient = mClients.valueFor(pid); 5559 5560 if (wclient != NULL) { 5561 client = wclient.promote(); 5562 } else { 5563 client = new Client(this, pid); 5564 mClients.add(pid, client); 5565 } 5566 5567 // create effect on selected output thread 5568 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5569 &desc, enabled, &lStatus); 5570 if (handle != 0 && id != NULL) { 5571 *id = handle->id(); 5572 } 5573 } 5574 5575Exit: 5576 if(status) { 5577 *status = lStatus; 5578 } 5579 return handle; 5580} 5581 5582status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5583{ 5584 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5585 sessionId, srcOutput, dstOutput); 5586 Mutex::Autolock _l(mLock); 5587 if (srcOutput == dstOutput) { 5588 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5589 return NO_ERROR; 5590 } 5591 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5592 if (srcThread == NULL) { 5593 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5594 return BAD_VALUE; 5595 } 5596 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5597 if (dstThread == NULL) { 5598 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5599 return BAD_VALUE; 5600 } 5601 5602 Mutex::Autolock _dl(dstThread->mLock); 5603 Mutex::Autolock _sl(srcThread->mLock); 5604 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5605 5606 return NO_ERROR; 5607} 5608 5609// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5610status_t AudioFlinger::moveEffectChain_l(int sessionId, 5611 AudioFlinger::PlaybackThread *srcThread, 5612 AudioFlinger::PlaybackThread *dstThread, 5613 bool reRegister) 5614{ 5615 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5616 sessionId, srcThread, dstThread); 5617 5618 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5619 if (chain == 0) { 5620 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5621 sessionId, srcThread); 5622 return INVALID_OPERATION; 5623 } 5624 5625 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5626 // so that a new chain is created with correct parameters when first effect is added. This is 5627 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5628 // removed. 5629 srcThread->removeEffectChain_l(chain); 5630 5631 // transfer all effects one by one so that new effect chain is created on new thread with 5632 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5633 int dstOutput = dstThread->id(); 5634 sp<EffectChain> dstChain; 5635 uint32_t strategy = 0; // prevent compiler warning 5636 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5637 while (effect != 0) { 5638 srcThread->removeEffect_l(effect); 5639 dstThread->addEffect_l(effect); 5640 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5641 if (effect->state() == EffectModule::ACTIVE || 5642 effect->state() == EffectModule::STOPPING) { 5643 effect->start(); 5644 } 5645 // if the move request is not received from audio policy manager, the effect must be 5646 // re-registered with the new strategy and output 5647 if (dstChain == 0) { 5648 dstChain = effect->chain().promote(); 5649 if (dstChain == 0) { 5650 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5651 srcThread->addEffect_l(effect); 5652 return NO_INIT; 5653 } 5654 strategy = dstChain->strategy(); 5655 } 5656 if (reRegister) { 5657 AudioSystem::unregisterEffect(effect->id()); 5658 AudioSystem::registerEffect(&effect->desc(), 5659 dstOutput, 5660 strategy, 5661 sessionId, 5662 effect->id()); 5663 } 5664 effect = chain->getEffectFromId_l(0); 5665 } 5666 5667 return NO_ERROR; 5668} 5669 5670 5671// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5672sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5673 const sp<AudioFlinger::Client>& client, 5674 const sp<IEffectClient>& effectClient, 5675 int32_t priority, 5676 int sessionId, 5677 effect_descriptor_t *desc, 5678 int *enabled, 5679 status_t *status 5680 ) 5681{ 5682 sp<EffectModule> effect; 5683 sp<EffectHandle> handle; 5684 status_t lStatus; 5685 sp<EffectChain> chain; 5686 bool chainCreated = false; 5687 bool effectCreated = false; 5688 bool effectRegistered = false; 5689 5690 lStatus = initCheck(); 5691 if (lStatus != NO_ERROR) { 5692 ALOGW("createEffect_l() Audio driver not initialized."); 5693 goto Exit; 5694 } 5695 5696 // Do not allow effects with session ID 0 on direct output or duplicating threads 5697 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5698 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5699 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5700 desc->name, sessionId); 5701 lStatus = BAD_VALUE; 5702 goto Exit; 5703 } 5704 // Only Pre processor effects are allowed on input threads and only on input threads 5705 if ((mType == RECORD && 5706 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5707 (mType != RECORD && 5708 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5709 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5710 desc->name, desc->flags, mType); 5711 lStatus = BAD_VALUE; 5712 goto Exit; 5713 } 5714 5715 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5716 5717 { // scope for mLock 5718 Mutex::Autolock _l(mLock); 5719 5720 // check for existing effect chain with the requested audio session 5721 chain = getEffectChain_l(sessionId); 5722 if (chain == 0) { 5723 // create a new chain for this session 5724 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5725 chain = new EffectChain(this, sessionId); 5726 addEffectChain_l(chain); 5727 chain->setStrategy(getStrategyForSession_l(sessionId)); 5728 chainCreated = true; 5729 } else { 5730 effect = chain->getEffectFromDesc_l(desc); 5731 } 5732 5733 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5734 5735 if (effect == 0) { 5736 int id = mAudioFlinger->nextUniqueId(); 5737 // Check CPU and memory usage 5738 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5739 if (lStatus != NO_ERROR) { 5740 goto Exit; 5741 } 5742 effectRegistered = true; 5743 // create a new effect module if none present in the chain 5744 effect = new EffectModule(this, chain, desc, id, sessionId); 5745 lStatus = effect->status(); 5746 if (lStatus != NO_ERROR) { 5747 goto Exit; 5748 } 5749 lStatus = chain->addEffect_l(effect); 5750 if (lStatus != NO_ERROR) { 5751 goto Exit; 5752 } 5753 effectCreated = true; 5754 5755 effect->setDevice(mDevice); 5756 effect->setMode(mAudioFlinger->getMode()); 5757 } 5758 // create effect handle and connect it to effect module 5759 handle = new EffectHandle(effect, client, effectClient, priority); 5760 lStatus = effect->addHandle(handle); 5761 if (enabled != NULL) { 5762 *enabled = (int)effect->isEnabled(); 5763 } 5764 } 5765 5766Exit: 5767 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5768 Mutex::Autolock _l(mLock); 5769 if (effectCreated) { 5770 chain->removeEffect_l(effect); 5771 } 5772 if (effectRegistered) { 5773 AudioSystem::unregisterEffect(effect->id()); 5774 } 5775 if (chainCreated) { 5776 removeEffectChain_l(chain); 5777 } 5778 handle.clear(); 5779 } 5780 5781 if(status) { 5782 *status = lStatus; 5783 } 5784 return handle; 5785} 5786 5787sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5788{ 5789 sp<EffectChain> chain = getEffectChain_l(sessionId); 5790 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 5791} 5792 5793// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5794// PlaybackThread::mLock held 5795status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5796{ 5797 // check for existing effect chain with the requested audio session 5798 int sessionId = effect->sessionId(); 5799 sp<EffectChain> chain = getEffectChain_l(sessionId); 5800 bool chainCreated = false; 5801 5802 if (chain == 0) { 5803 // create a new chain for this session 5804 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5805 chain = new EffectChain(this, sessionId); 5806 addEffectChain_l(chain); 5807 chain->setStrategy(getStrategyForSession_l(sessionId)); 5808 chainCreated = true; 5809 } 5810 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5811 5812 if (chain->getEffectFromId_l(effect->id()) != 0) { 5813 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5814 this, effect->desc().name, chain.get()); 5815 return BAD_VALUE; 5816 } 5817 5818 status_t status = chain->addEffect_l(effect); 5819 if (status != NO_ERROR) { 5820 if (chainCreated) { 5821 removeEffectChain_l(chain); 5822 } 5823 return status; 5824 } 5825 5826 effect->setDevice(mDevice); 5827 effect->setMode(mAudioFlinger->getMode()); 5828 return NO_ERROR; 5829} 5830 5831void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5832 5833 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5834 effect_descriptor_t desc = effect->desc(); 5835 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5836 detachAuxEffect_l(effect->id()); 5837 } 5838 5839 sp<EffectChain> chain = effect->chain().promote(); 5840 if (chain != 0) { 5841 // remove effect chain if removing last effect 5842 if (chain->removeEffect_l(effect) == 0) { 5843 removeEffectChain_l(chain); 5844 } 5845 } else { 5846 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5847 } 5848} 5849 5850void AudioFlinger::ThreadBase::lockEffectChains_l( 5851 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5852{ 5853 effectChains = mEffectChains; 5854 for (size_t i = 0; i < mEffectChains.size(); i++) { 5855 mEffectChains[i]->lock(); 5856 } 5857} 5858 5859void AudioFlinger::ThreadBase::unlockEffectChains( 5860 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5861{ 5862 for (size_t i = 0; i < effectChains.size(); i++) { 5863 effectChains[i]->unlock(); 5864 } 5865} 5866 5867sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5868{ 5869 Mutex::Autolock _l(mLock); 5870 return getEffectChain_l(sessionId); 5871} 5872 5873sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5874{ 5875 size_t size = mEffectChains.size(); 5876 for (size_t i = 0; i < size; i++) { 5877 if (mEffectChains[i]->sessionId() == sessionId) { 5878 return mEffectChains[i]; 5879 } 5880 } 5881 return 0; 5882} 5883 5884void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5885{ 5886 Mutex::Autolock _l(mLock); 5887 size_t size = mEffectChains.size(); 5888 for (size_t i = 0; i < size; i++) { 5889 mEffectChains[i]->setMode_l(mode); 5890 } 5891} 5892 5893void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5894 const wp<EffectHandle>& handle, 5895 bool unpiniflast) { 5896 5897 Mutex::Autolock _l(mLock); 5898 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5899 // delete the effect module if removing last handle on it 5900 if (effect->removeHandle(handle) == 0) { 5901 if (!effect->isPinned() || unpiniflast) { 5902 removeEffect_l(effect); 5903 AudioSystem::unregisterEffect(effect->id()); 5904 } 5905 } 5906} 5907 5908status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5909{ 5910 int session = chain->sessionId(); 5911 int16_t *buffer = mMixBuffer; 5912 bool ownsBuffer = false; 5913 5914 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5915 if (session > 0) { 5916 // Only one effect chain can be present in direct output thread and it uses 5917 // the mix buffer as input 5918 if (mType != DIRECT) { 5919 size_t numSamples = mFrameCount * mChannelCount; 5920 buffer = new int16_t[numSamples]; 5921 memset(buffer, 0, numSamples * sizeof(int16_t)); 5922 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5923 ownsBuffer = true; 5924 } 5925 5926 // Attach all tracks with same session ID to this chain. 5927 for (size_t i = 0; i < mTracks.size(); ++i) { 5928 sp<Track> track = mTracks[i]; 5929 if (session == track->sessionId()) { 5930 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5931 track->setMainBuffer(buffer); 5932 chain->incTrackCnt(); 5933 } 5934 } 5935 5936 // indicate all active tracks in the chain 5937 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5938 sp<Track> track = mActiveTracks[i].promote(); 5939 if (track == 0) continue; 5940 if (session == track->sessionId()) { 5941 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5942 chain->incActiveTrackCnt(); 5943 } 5944 } 5945 } 5946 5947 chain->setInBuffer(buffer, ownsBuffer); 5948 chain->setOutBuffer(mMixBuffer); 5949 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5950 // chains list in order to be processed last as it contains output stage effects 5951 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5952 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5953 // after track specific effects and before output stage 5954 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5955 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5956 // Effect chain for other sessions are inserted at beginning of effect 5957 // chains list to be processed before output mix effects. Relative order between other 5958 // sessions is not important 5959 size_t size = mEffectChains.size(); 5960 size_t i = 0; 5961 for (i = 0; i < size; i++) { 5962 if (mEffectChains[i]->sessionId() < session) break; 5963 } 5964 mEffectChains.insertAt(chain, i); 5965 checkSuspendOnAddEffectChain_l(chain); 5966 5967 return NO_ERROR; 5968} 5969 5970size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5971{ 5972 int session = chain->sessionId(); 5973 5974 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5975 5976 for (size_t i = 0; i < mEffectChains.size(); i++) { 5977 if (chain == mEffectChains[i]) { 5978 mEffectChains.removeAt(i); 5979 // detach all active tracks from the chain 5980 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5981 sp<Track> track = mActiveTracks[i].promote(); 5982 if (track == 0) continue; 5983 if (session == track->sessionId()) { 5984 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5985 chain.get(), session); 5986 chain->decActiveTrackCnt(); 5987 } 5988 } 5989 5990 // detach all tracks with same session ID from this chain 5991 for (size_t i = 0; i < mTracks.size(); ++i) { 5992 sp<Track> track = mTracks[i]; 5993 if (session == track->sessionId()) { 5994 track->setMainBuffer(mMixBuffer); 5995 chain->decTrackCnt(); 5996 } 5997 } 5998 break; 5999 } 6000 } 6001 return mEffectChains.size(); 6002} 6003 6004status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6005 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6006{ 6007 Mutex::Autolock _l(mLock); 6008 return attachAuxEffect_l(track, EffectId); 6009} 6010 6011status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6012 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6013{ 6014 status_t status = NO_ERROR; 6015 6016 if (EffectId == 0) { 6017 track->setAuxBuffer(0, NULL); 6018 } else { 6019 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6020 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6021 if (effect != 0) { 6022 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6023 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6024 } else { 6025 status = INVALID_OPERATION; 6026 } 6027 } else { 6028 status = BAD_VALUE; 6029 } 6030 } 6031 return status; 6032} 6033 6034void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6035{ 6036 for (size_t i = 0; i < mTracks.size(); ++i) { 6037 sp<Track> track = mTracks[i]; 6038 if (track->auxEffectId() == effectId) { 6039 attachAuxEffect_l(track, 0); 6040 } 6041 } 6042} 6043 6044status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6045{ 6046 // only one chain per input thread 6047 if (mEffectChains.size() != 0) { 6048 return INVALID_OPERATION; 6049 } 6050 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6051 6052 chain->setInBuffer(NULL); 6053 chain->setOutBuffer(NULL); 6054 6055 checkSuspendOnAddEffectChain_l(chain); 6056 6057 mEffectChains.add(chain); 6058 6059 return NO_ERROR; 6060} 6061 6062size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6063{ 6064 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6065 ALOGW_IF(mEffectChains.size() != 1, 6066 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6067 chain.get(), mEffectChains.size(), this); 6068 if (mEffectChains.size() == 1) { 6069 mEffectChains.removeAt(0); 6070 } 6071 return 0; 6072} 6073 6074// ---------------------------------------------------------------------------- 6075// EffectModule implementation 6076// ---------------------------------------------------------------------------- 6077 6078#undef LOG_TAG 6079#define LOG_TAG "AudioFlinger::EffectModule" 6080 6081AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6082 const wp<AudioFlinger::EffectChain>& chain, 6083 effect_descriptor_t *desc, 6084 int id, 6085 int sessionId) 6086 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6087 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6088{ 6089 ALOGV("Constructor %p", this); 6090 int lStatus; 6091 sp<ThreadBase> thread = mThread.promote(); 6092 if (thread == 0) { 6093 return; 6094 } 6095 6096 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6097 6098 // create effect engine from effect factory 6099 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6100 6101 if (mStatus != NO_ERROR) { 6102 return; 6103 } 6104 lStatus = init(); 6105 if (lStatus < 0) { 6106 mStatus = lStatus; 6107 goto Error; 6108 } 6109 6110 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6111 mPinned = true; 6112 } 6113 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6114 return; 6115Error: 6116 EffectRelease(mEffectInterface); 6117 mEffectInterface = NULL; 6118 ALOGV("Constructor Error %d", mStatus); 6119} 6120 6121AudioFlinger::EffectModule::~EffectModule() 6122{ 6123 ALOGV("Destructor %p", this); 6124 if (mEffectInterface != NULL) { 6125 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6126 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6127 sp<ThreadBase> thread = mThread.promote(); 6128 if (thread != 0) { 6129 audio_stream_t *stream = thread->stream(); 6130 if (stream != NULL) { 6131 stream->remove_audio_effect(stream, mEffectInterface); 6132 } 6133 } 6134 } 6135 // release effect engine 6136 EffectRelease(mEffectInterface); 6137 } 6138} 6139 6140status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6141{ 6142 status_t status; 6143 6144 Mutex::Autolock _l(mLock); 6145 // First handle in mHandles has highest priority and controls the effect module 6146 int priority = handle->priority(); 6147 size_t size = mHandles.size(); 6148 sp<EffectHandle> h; 6149 size_t i; 6150 for (i = 0; i < size; i++) { 6151 h = mHandles[i].promote(); 6152 if (h == 0) continue; 6153 if (h->priority() <= priority) break; 6154 } 6155 // if inserted in first place, move effect control from previous owner to this handle 6156 if (i == 0) { 6157 bool enabled = false; 6158 if (h != 0) { 6159 enabled = h->enabled(); 6160 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6161 } 6162 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6163 status = NO_ERROR; 6164 } else { 6165 status = ALREADY_EXISTS; 6166 } 6167 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6168 mHandles.insertAt(handle, i); 6169 return status; 6170} 6171 6172size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6173{ 6174 Mutex::Autolock _l(mLock); 6175 size_t size = mHandles.size(); 6176 size_t i; 6177 for (i = 0; i < size; i++) { 6178 if (mHandles[i] == handle) break; 6179 } 6180 if (i == size) { 6181 return size; 6182 } 6183 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6184 6185 bool enabled = false; 6186 EffectHandle *hdl = handle.unsafe_get(); 6187 if (hdl != NULL) { 6188 ALOGV("removeHandle() unsafe_get OK"); 6189 enabled = hdl->enabled(); 6190 } 6191 mHandles.removeAt(i); 6192 size = mHandles.size(); 6193 // if removed from first place, move effect control from this handle to next in line 6194 if (i == 0 && size != 0) { 6195 sp<EffectHandle> h = mHandles[0].promote(); 6196 if (h != 0) { 6197 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6198 } 6199 } 6200 6201 // Prevent calls to process() and other functions on effect interface from now on. 6202 // The effect engine will be released by the destructor when the last strong reference on 6203 // this object is released which can happen after next process is called. 6204 if (size == 0 && !mPinned) { 6205 mState = DESTROYED; 6206 } 6207 6208 return size; 6209} 6210 6211sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6212{ 6213 Mutex::Autolock _l(mLock); 6214 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6215} 6216 6217void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6218{ 6219 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6220 // keep a strong reference on this EffectModule to avoid calling the 6221 // destructor before we exit 6222 sp<EffectModule> keep(this); 6223 { 6224 sp<ThreadBase> thread = mThread.promote(); 6225 if (thread != 0) { 6226 thread->disconnectEffect(keep, handle, unpiniflast); 6227 } 6228 } 6229} 6230 6231void AudioFlinger::EffectModule::updateState() { 6232 Mutex::Autolock _l(mLock); 6233 6234 switch (mState) { 6235 case RESTART: 6236 reset_l(); 6237 // FALL THROUGH 6238 6239 case STARTING: 6240 // clear auxiliary effect input buffer for next accumulation 6241 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6242 memset(mConfig.inputCfg.buffer.raw, 6243 0, 6244 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6245 } 6246 start_l(); 6247 mState = ACTIVE; 6248 break; 6249 case STOPPING: 6250 stop_l(); 6251 mDisableWaitCnt = mMaxDisableWaitCnt; 6252 mState = STOPPED; 6253 break; 6254 case STOPPED: 6255 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6256 // turn off sequence. 6257 if (--mDisableWaitCnt == 0) { 6258 reset_l(); 6259 mState = IDLE; 6260 } 6261 break; 6262 default: //IDLE , ACTIVE, DESTROYED 6263 break; 6264 } 6265} 6266 6267void AudioFlinger::EffectModule::process() 6268{ 6269 Mutex::Autolock _l(mLock); 6270 6271 if (mState == DESTROYED || mEffectInterface == NULL || 6272 mConfig.inputCfg.buffer.raw == NULL || 6273 mConfig.outputCfg.buffer.raw == NULL) { 6274 return; 6275 } 6276 6277 if (isProcessEnabled()) { 6278 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6279 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6280 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6281 mConfig.inputCfg.buffer.s32, 6282 mConfig.inputCfg.buffer.frameCount/2); 6283 } 6284 6285 // do the actual processing in the effect engine 6286 int ret = (*mEffectInterface)->process(mEffectInterface, 6287 &mConfig.inputCfg.buffer, 6288 &mConfig.outputCfg.buffer); 6289 6290 // force transition to IDLE state when engine is ready 6291 if (mState == STOPPED && ret == -ENODATA) { 6292 mDisableWaitCnt = 1; 6293 } 6294 6295 // clear auxiliary effect input buffer for next accumulation 6296 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6297 memset(mConfig.inputCfg.buffer.raw, 0, 6298 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6299 } 6300 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6301 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6302 // If an insert effect is idle and input buffer is different from output buffer, 6303 // accumulate input onto output 6304 sp<EffectChain> chain = mChain.promote(); 6305 if (chain != 0 && chain->activeTrackCnt() != 0) { 6306 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6307 int16_t *in = mConfig.inputCfg.buffer.s16; 6308 int16_t *out = mConfig.outputCfg.buffer.s16; 6309 for (size_t i = 0; i < frameCnt; i++) { 6310 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6311 } 6312 } 6313 } 6314} 6315 6316void AudioFlinger::EffectModule::reset_l() 6317{ 6318 if (mEffectInterface == NULL) { 6319 return; 6320 } 6321 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6322} 6323 6324status_t AudioFlinger::EffectModule::configure() 6325{ 6326 uint32_t channels; 6327 if (mEffectInterface == NULL) { 6328 return NO_INIT; 6329 } 6330 6331 sp<ThreadBase> thread = mThread.promote(); 6332 if (thread == 0) { 6333 return DEAD_OBJECT; 6334 } 6335 6336 // TODO: handle configuration of effects replacing track process 6337 if (thread->channelCount() == 1) { 6338 channels = AUDIO_CHANNEL_OUT_MONO; 6339 } else { 6340 channels = AUDIO_CHANNEL_OUT_STEREO; 6341 } 6342 6343 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6344 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6345 } else { 6346 mConfig.inputCfg.channels = channels; 6347 } 6348 mConfig.outputCfg.channels = channels; 6349 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6350 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6351 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6352 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6353 mConfig.inputCfg.bufferProvider.cookie = NULL; 6354 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6355 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6356 mConfig.outputCfg.bufferProvider.cookie = NULL; 6357 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6358 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6359 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6360 // Insert effect: 6361 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6362 // always overwrites output buffer: input buffer == output buffer 6363 // - in other sessions: 6364 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6365 // other effect: overwrites output buffer: input buffer == output buffer 6366 // Auxiliary effect: 6367 // accumulates in output buffer: input buffer != output buffer 6368 // Therefore: accumulate <=> input buffer != output buffer 6369 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6370 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6371 } else { 6372 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6373 } 6374 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6375 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6376 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6377 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6378 6379 ALOGV("configure() %p thread %p buffer %p framecount %d", 6380 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6381 6382 status_t cmdStatus; 6383 uint32_t size = sizeof(int); 6384 status_t status = (*mEffectInterface)->command(mEffectInterface, 6385 EFFECT_CMD_SET_CONFIG, 6386 sizeof(effect_config_t), 6387 &mConfig, 6388 &size, 6389 &cmdStatus); 6390 if (status == 0) { 6391 status = cmdStatus; 6392 } 6393 6394 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6395 (1000 * mConfig.outputCfg.buffer.frameCount); 6396 6397 return status; 6398} 6399 6400status_t AudioFlinger::EffectModule::init() 6401{ 6402 Mutex::Autolock _l(mLock); 6403 if (mEffectInterface == NULL) { 6404 return NO_INIT; 6405 } 6406 status_t cmdStatus; 6407 uint32_t size = sizeof(status_t); 6408 status_t status = (*mEffectInterface)->command(mEffectInterface, 6409 EFFECT_CMD_INIT, 6410 0, 6411 NULL, 6412 &size, 6413 &cmdStatus); 6414 if (status == 0) { 6415 status = cmdStatus; 6416 } 6417 return status; 6418} 6419 6420status_t AudioFlinger::EffectModule::start() 6421{ 6422 Mutex::Autolock _l(mLock); 6423 return start_l(); 6424} 6425 6426status_t AudioFlinger::EffectModule::start_l() 6427{ 6428 if (mEffectInterface == NULL) { 6429 return NO_INIT; 6430 } 6431 status_t cmdStatus; 6432 uint32_t size = sizeof(status_t); 6433 status_t status = (*mEffectInterface)->command(mEffectInterface, 6434 EFFECT_CMD_ENABLE, 6435 0, 6436 NULL, 6437 &size, 6438 &cmdStatus); 6439 if (status == 0) { 6440 status = cmdStatus; 6441 } 6442 if (status == 0 && 6443 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6444 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6445 sp<ThreadBase> thread = mThread.promote(); 6446 if (thread != 0) { 6447 audio_stream_t *stream = thread->stream(); 6448 if (stream != NULL) { 6449 stream->add_audio_effect(stream, mEffectInterface); 6450 } 6451 } 6452 } 6453 return status; 6454} 6455 6456status_t AudioFlinger::EffectModule::stop() 6457{ 6458 Mutex::Autolock _l(mLock); 6459 return stop_l(); 6460} 6461 6462status_t AudioFlinger::EffectModule::stop_l() 6463{ 6464 if (mEffectInterface == NULL) { 6465 return NO_INIT; 6466 } 6467 status_t cmdStatus; 6468 uint32_t size = sizeof(status_t); 6469 status_t status = (*mEffectInterface)->command(mEffectInterface, 6470 EFFECT_CMD_DISABLE, 6471 0, 6472 NULL, 6473 &size, 6474 &cmdStatus); 6475 if (status == 0) { 6476 status = cmdStatus; 6477 } 6478 if (status == 0 && 6479 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6480 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6481 sp<ThreadBase> thread = mThread.promote(); 6482 if (thread != 0) { 6483 audio_stream_t *stream = thread->stream(); 6484 if (stream != NULL) { 6485 stream->remove_audio_effect(stream, mEffectInterface); 6486 } 6487 } 6488 } 6489 return status; 6490} 6491 6492status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6493 uint32_t cmdSize, 6494 void *pCmdData, 6495 uint32_t *replySize, 6496 void *pReplyData) 6497{ 6498 Mutex::Autolock _l(mLock); 6499// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6500 6501 if (mState == DESTROYED || mEffectInterface == NULL) { 6502 return NO_INIT; 6503 } 6504 status_t status = (*mEffectInterface)->command(mEffectInterface, 6505 cmdCode, 6506 cmdSize, 6507 pCmdData, 6508 replySize, 6509 pReplyData); 6510 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6511 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6512 for (size_t i = 1; i < mHandles.size(); i++) { 6513 sp<EffectHandle> h = mHandles[i].promote(); 6514 if (h != 0) { 6515 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6516 } 6517 } 6518 } 6519 return status; 6520} 6521 6522status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6523{ 6524 6525 Mutex::Autolock _l(mLock); 6526 ALOGV("setEnabled %p enabled %d", this, enabled); 6527 6528 if (enabled != isEnabled()) { 6529 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6530 if (enabled && status != NO_ERROR) { 6531 return status; 6532 } 6533 6534 switch (mState) { 6535 // going from disabled to enabled 6536 case IDLE: 6537 mState = STARTING; 6538 break; 6539 case STOPPED: 6540 mState = RESTART; 6541 break; 6542 case STOPPING: 6543 mState = ACTIVE; 6544 break; 6545 6546 // going from enabled to disabled 6547 case RESTART: 6548 mState = STOPPED; 6549 break; 6550 case STARTING: 6551 mState = IDLE; 6552 break; 6553 case ACTIVE: 6554 mState = STOPPING; 6555 break; 6556 case DESTROYED: 6557 return NO_ERROR; // simply ignore as we are being destroyed 6558 } 6559 for (size_t i = 1; i < mHandles.size(); i++) { 6560 sp<EffectHandle> h = mHandles[i].promote(); 6561 if (h != 0) { 6562 h->setEnabled(enabled); 6563 } 6564 } 6565 } 6566 return NO_ERROR; 6567} 6568 6569bool AudioFlinger::EffectModule::isEnabled() const 6570{ 6571 switch (mState) { 6572 case RESTART: 6573 case STARTING: 6574 case ACTIVE: 6575 return true; 6576 case IDLE: 6577 case STOPPING: 6578 case STOPPED: 6579 case DESTROYED: 6580 default: 6581 return false; 6582 } 6583} 6584 6585bool AudioFlinger::EffectModule::isProcessEnabled() const 6586{ 6587 switch (mState) { 6588 case RESTART: 6589 case ACTIVE: 6590 case STOPPING: 6591 case STOPPED: 6592 return true; 6593 case IDLE: 6594 case STARTING: 6595 case DESTROYED: 6596 default: 6597 return false; 6598 } 6599} 6600 6601status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6602{ 6603 Mutex::Autolock _l(mLock); 6604 status_t status = NO_ERROR; 6605 6606 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6607 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6608 if (isProcessEnabled() && 6609 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6610 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6611 status_t cmdStatus; 6612 uint32_t volume[2]; 6613 uint32_t *pVolume = NULL; 6614 uint32_t size = sizeof(volume); 6615 volume[0] = *left; 6616 volume[1] = *right; 6617 if (controller) { 6618 pVolume = volume; 6619 } 6620 status = (*mEffectInterface)->command(mEffectInterface, 6621 EFFECT_CMD_SET_VOLUME, 6622 size, 6623 volume, 6624 &size, 6625 pVolume); 6626 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6627 *left = volume[0]; 6628 *right = volume[1]; 6629 } 6630 } 6631 return status; 6632} 6633 6634status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6635{ 6636 Mutex::Autolock _l(mLock); 6637 status_t status = NO_ERROR; 6638 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6639 // audio pre processing modules on RecordThread can receive both output and 6640 // input device indication in the same call 6641 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6642 if (dev) { 6643 status_t cmdStatus; 6644 uint32_t size = sizeof(status_t); 6645 6646 status = (*mEffectInterface)->command(mEffectInterface, 6647 EFFECT_CMD_SET_DEVICE, 6648 sizeof(uint32_t), 6649 &dev, 6650 &size, 6651 &cmdStatus); 6652 if (status == NO_ERROR) { 6653 status = cmdStatus; 6654 } 6655 } 6656 dev = device & AUDIO_DEVICE_IN_ALL; 6657 if (dev) { 6658 status_t cmdStatus; 6659 uint32_t size = sizeof(status_t); 6660 6661 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6662 EFFECT_CMD_SET_INPUT_DEVICE, 6663 sizeof(uint32_t), 6664 &dev, 6665 &size, 6666 &cmdStatus); 6667 if (status2 == NO_ERROR) { 6668 status2 = cmdStatus; 6669 } 6670 if (status == NO_ERROR) { 6671 status = status2; 6672 } 6673 } 6674 } 6675 return status; 6676} 6677 6678status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6679{ 6680 Mutex::Autolock _l(mLock); 6681 status_t status = NO_ERROR; 6682 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6683 status_t cmdStatus; 6684 uint32_t size = sizeof(status_t); 6685 status = (*mEffectInterface)->command(mEffectInterface, 6686 EFFECT_CMD_SET_AUDIO_MODE, 6687 sizeof(audio_mode_t), 6688 &mode, 6689 &size, 6690 &cmdStatus); 6691 if (status == NO_ERROR) { 6692 status = cmdStatus; 6693 } 6694 } 6695 return status; 6696} 6697 6698void AudioFlinger::EffectModule::setSuspended(bool suspended) 6699{ 6700 Mutex::Autolock _l(mLock); 6701 mSuspended = suspended; 6702} 6703 6704bool AudioFlinger::EffectModule::suspended() const 6705{ 6706 Mutex::Autolock _l(mLock); 6707 return mSuspended; 6708} 6709 6710status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6711{ 6712 const size_t SIZE = 256; 6713 char buffer[SIZE]; 6714 String8 result; 6715 6716 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6717 result.append(buffer); 6718 6719 bool locked = tryLock(mLock); 6720 // failed to lock - AudioFlinger is probably deadlocked 6721 if (!locked) { 6722 result.append("\t\tCould not lock Fx mutex:\n"); 6723 } 6724 6725 result.append("\t\tSession Status State Engine:\n"); 6726 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6727 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6728 result.append(buffer); 6729 6730 result.append("\t\tDescriptor:\n"); 6731 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6732 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6733 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6734 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6735 result.append(buffer); 6736 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6737 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6738 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6739 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6740 result.append(buffer); 6741 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6742 mDescriptor.apiVersion, 6743 mDescriptor.flags); 6744 result.append(buffer); 6745 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6746 mDescriptor.name); 6747 result.append(buffer); 6748 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6749 mDescriptor.implementor); 6750 result.append(buffer); 6751 6752 result.append("\t\t- Input configuration:\n"); 6753 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6754 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6755 (uint32_t)mConfig.inputCfg.buffer.raw, 6756 mConfig.inputCfg.buffer.frameCount, 6757 mConfig.inputCfg.samplingRate, 6758 mConfig.inputCfg.channels, 6759 mConfig.inputCfg.format); 6760 result.append(buffer); 6761 6762 result.append("\t\t- Output configuration:\n"); 6763 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6764 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6765 (uint32_t)mConfig.outputCfg.buffer.raw, 6766 mConfig.outputCfg.buffer.frameCount, 6767 mConfig.outputCfg.samplingRate, 6768 mConfig.outputCfg.channels, 6769 mConfig.outputCfg.format); 6770 result.append(buffer); 6771 6772 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6773 result.append(buffer); 6774 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6775 for (size_t i = 0; i < mHandles.size(); ++i) { 6776 sp<EffectHandle> handle = mHandles[i].promote(); 6777 if (handle != 0) { 6778 handle->dump(buffer, SIZE); 6779 result.append(buffer); 6780 } 6781 } 6782 6783 result.append("\n"); 6784 6785 write(fd, result.string(), result.length()); 6786 6787 if (locked) { 6788 mLock.unlock(); 6789 } 6790 6791 return NO_ERROR; 6792} 6793 6794// ---------------------------------------------------------------------------- 6795// EffectHandle implementation 6796// ---------------------------------------------------------------------------- 6797 6798#undef LOG_TAG 6799#define LOG_TAG "AudioFlinger::EffectHandle" 6800 6801AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6802 const sp<AudioFlinger::Client>& client, 6803 const sp<IEffectClient>& effectClient, 6804 int32_t priority) 6805 : BnEffect(), 6806 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6807 mPriority(priority), mHasControl(false), mEnabled(false) 6808{ 6809 ALOGV("constructor %p", this); 6810 6811 if (client == 0) { 6812 return; 6813 } 6814 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6815 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6816 if (mCblkMemory != 0) { 6817 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6818 6819 if (mCblk != NULL) { 6820 new(mCblk) effect_param_cblk_t(); 6821 mBuffer = (uint8_t *)mCblk + bufOffset; 6822 } 6823 } else { 6824 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6825 return; 6826 } 6827} 6828 6829AudioFlinger::EffectHandle::~EffectHandle() 6830{ 6831 ALOGV("Destructor %p", this); 6832 disconnect(false); 6833 ALOGV("Destructor DONE %p", this); 6834} 6835 6836status_t AudioFlinger::EffectHandle::enable() 6837{ 6838 ALOGV("enable %p", this); 6839 if (!mHasControl) return INVALID_OPERATION; 6840 if (mEffect == 0) return DEAD_OBJECT; 6841 6842 if (mEnabled) { 6843 return NO_ERROR; 6844 } 6845 6846 mEnabled = true; 6847 6848 sp<ThreadBase> thread = mEffect->thread().promote(); 6849 if (thread != 0) { 6850 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6851 } 6852 6853 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6854 if (mEffect->suspended()) { 6855 return NO_ERROR; 6856 } 6857 6858 status_t status = mEffect->setEnabled(true); 6859 if (status != NO_ERROR) { 6860 if (thread != 0) { 6861 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6862 } 6863 mEnabled = false; 6864 } 6865 return status; 6866} 6867 6868status_t AudioFlinger::EffectHandle::disable() 6869{ 6870 ALOGV("disable %p", this); 6871 if (!mHasControl) return INVALID_OPERATION; 6872 if (mEffect == 0) return DEAD_OBJECT; 6873 6874 if (!mEnabled) { 6875 return NO_ERROR; 6876 } 6877 mEnabled = false; 6878 6879 if (mEffect->suspended()) { 6880 return NO_ERROR; 6881 } 6882 6883 status_t status = mEffect->setEnabled(false); 6884 6885 sp<ThreadBase> thread = mEffect->thread().promote(); 6886 if (thread != 0) { 6887 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6888 } 6889 6890 return status; 6891} 6892 6893void AudioFlinger::EffectHandle::disconnect() 6894{ 6895 disconnect(true); 6896} 6897 6898void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6899{ 6900 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6901 if (mEffect == 0) { 6902 return; 6903 } 6904 mEffect->disconnect(this, unpiniflast); 6905 6906 if (mHasControl && mEnabled) { 6907 sp<ThreadBase> thread = mEffect->thread().promote(); 6908 if (thread != 0) { 6909 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6910 } 6911 } 6912 6913 // release sp on module => module destructor can be called now 6914 mEffect.clear(); 6915 if (mClient != 0) { 6916 if (mCblk != NULL) { 6917 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 6918 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6919 } 6920 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 6921 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6922 mClient.clear(); 6923 } 6924} 6925 6926status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6927 uint32_t cmdSize, 6928 void *pCmdData, 6929 uint32_t *replySize, 6930 void *pReplyData) 6931{ 6932// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6933// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6934 6935 // only get parameter command is permitted for applications not controlling the effect 6936 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6937 return INVALID_OPERATION; 6938 } 6939 if (mEffect == 0) return DEAD_OBJECT; 6940 if (mClient == 0) return INVALID_OPERATION; 6941 6942 // handle commands that are not forwarded transparently to effect engine 6943 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6944 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6945 // no risk to block the whole media server process or mixer threads is we are stuck here 6946 Mutex::Autolock _l(mCblk->lock); 6947 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6948 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6949 mCblk->serverIndex = 0; 6950 mCblk->clientIndex = 0; 6951 return BAD_VALUE; 6952 } 6953 status_t status = NO_ERROR; 6954 while (mCblk->serverIndex < mCblk->clientIndex) { 6955 int reply; 6956 uint32_t rsize = sizeof(int); 6957 int *p = (int *)(mBuffer + mCblk->serverIndex); 6958 int size = *p++; 6959 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6960 ALOGW("command(): invalid parameter block size"); 6961 break; 6962 } 6963 effect_param_t *param = (effect_param_t *)p; 6964 if (param->psize == 0 || param->vsize == 0) { 6965 ALOGW("command(): null parameter or value size"); 6966 mCblk->serverIndex += size; 6967 continue; 6968 } 6969 uint32_t psize = sizeof(effect_param_t) + 6970 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6971 param->vsize; 6972 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6973 psize, 6974 p, 6975 &rsize, 6976 &reply); 6977 // stop at first error encountered 6978 if (ret != NO_ERROR) { 6979 status = ret; 6980 *(int *)pReplyData = reply; 6981 break; 6982 } else if (reply != NO_ERROR) { 6983 *(int *)pReplyData = reply; 6984 break; 6985 } 6986 mCblk->serverIndex += size; 6987 } 6988 mCblk->serverIndex = 0; 6989 mCblk->clientIndex = 0; 6990 return status; 6991 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6992 *(int *)pReplyData = NO_ERROR; 6993 return enable(); 6994 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6995 *(int *)pReplyData = NO_ERROR; 6996 return disable(); 6997 } 6998 6999 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7000} 7001 7002void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7003{ 7004 ALOGV("setControl %p control %d", this, hasControl); 7005 7006 mHasControl = hasControl; 7007 mEnabled = enabled; 7008 7009 if (signal && mEffectClient != 0) { 7010 mEffectClient->controlStatusChanged(hasControl); 7011 } 7012} 7013 7014void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7015 uint32_t cmdSize, 7016 void *pCmdData, 7017 uint32_t replySize, 7018 void *pReplyData) 7019{ 7020 if (mEffectClient != 0) { 7021 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7022 } 7023} 7024 7025 7026 7027void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7028{ 7029 if (mEffectClient != 0) { 7030 mEffectClient->enableStatusChanged(enabled); 7031 } 7032} 7033 7034status_t AudioFlinger::EffectHandle::onTransact( 7035 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7036{ 7037 return BnEffect::onTransact(code, data, reply, flags); 7038} 7039 7040 7041void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7042{ 7043 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7044 7045 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7046 (mClient == 0) ? getpid() : mClient->pid(), 7047 mPriority, 7048 mHasControl, 7049 !locked, 7050 mCblk ? mCblk->clientIndex : 0, 7051 mCblk ? mCblk->serverIndex : 0 7052 ); 7053 7054 if (locked) { 7055 mCblk->lock.unlock(); 7056 } 7057} 7058 7059#undef LOG_TAG 7060#define LOG_TAG "AudioFlinger::EffectChain" 7061 7062AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7063 int sessionId) 7064 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7065 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7066 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7067{ 7068 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7069 sp<ThreadBase> thread = mThread.promote(); 7070 if (thread == 0) { 7071 return; 7072 } 7073 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7074 thread->frameCount(); 7075} 7076 7077AudioFlinger::EffectChain::~EffectChain() 7078{ 7079 if (mOwnInBuffer) { 7080 delete mInBuffer; 7081 } 7082 7083} 7084 7085// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7086sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7087{ 7088 size_t size = mEffects.size(); 7089 7090 for (size_t i = 0; i < size; i++) { 7091 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7092 return mEffects[i]; 7093 } 7094 } 7095 return 0; 7096} 7097 7098// getEffectFromId_l() must be called with ThreadBase::mLock held 7099sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7100{ 7101 size_t size = mEffects.size(); 7102 7103 for (size_t i = 0; i < size; i++) { 7104 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7105 if (id == 0 || mEffects[i]->id() == id) { 7106 return mEffects[i]; 7107 } 7108 } 7109 return 0; 7110} 7111 7112// getEffectFromType_l() must be called with ThreadBase::mLock held 7113sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7114 const effect_uuid_t *type) 7115{ 7116 size_t size = mEffects.size(); 7117 7118 for (size_t i = 0; i < size; i++) { 7119 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7120 return mEffects[i]; 7121 } 7122 } 7123 return 0; 7124} 7125 7126// Must be called with EffectChain::mLock locked 7127void AudioFlinger::EffectChain::process_l() 7128{ 7129 sp<ThreadBase> thread = mThread.promote(); 7130 if (thread == 0) { 7131 ALOGW("process_l(): cannot promote mixer thread"); 7132 return; 7133 } 7134 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7135 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7136 // always process effects unless no more tracks are on the session and the effect tail 7137 // has been rendered 7138 bool doProcess = true; 7139 if (!isGlobalSession) { 7140 bool tracksOnSession = (trackCnt() != 0); 7141 7142 if (!tracksOnSession && mTailBufferCount == 0) { 7143 doProcess = false; 7144 } 7145 7146 if (activeTrackCnt() == 0) { 7147 // if no track is active and the effect tail has not been rendered, 7148 // the input buffer must be cleared here as the mixer process will not do it 7149 if (tracksOnSession || mTailBufferCount > 0) { 7150 size_t numSamples = thread->frameCount() * thread->channelCount(); 7151 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7152 if (mTailBufferCount > 0) { 7153 mTailBufferCount--; 7154 } 7155 } 7156 } 7157 } 7158 7159 size_t size = mEffects.size(); 7160 if (doProcess) { 7161 for (size_t i = 0; i < size; i++) { 7162 mEffects[i]->process(); 7163 } 7164 } 7165 for (size_t i = 0; i < size; i++) { 7166 mEffects[i]->updateState(); 7167 } 7168} 7169 7170// addEffect_l() must be called with PlaybackThread::mLock held 7171status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7172{ 7173 effect_descriptor_t desc = effect->desc(); 7174 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7175 7176 Mutex::Autolock _l(mLock); 7177 effect->setChain(this); 7178 sp<ThreadBase> thread = mThread.promote(); 7179 if (thread == 0) { 7180 return NO_INIT; 7181 } 7182 effect->setThread(thread); 7183 7184 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7185 // Auxiliary effects are inserted at the beginning of mEffects vector as 7186 // they are processed first and accumulated in chain input buffer 7187 mEffects.insertAt(effect, 0); 7188 7189 // the input buffer for auxiliary effect contains mono samples in 7190 // 32 bit format. This is to avoid saturation in AudoMixer 7191 // accumulation stage. Saturation is done in EffectModule::process() before 7192 // calling the process in effect engine 7193 size_t numSamples = thread->frameCount(); 7194 int32_t *buffer = new int32_t[numSamples]; 7195 memset(buffer, 0, numSamples * sizeof(int32_t)); 7196 effect->setInBuffer((int16_t *)buffer); 7197 // auxiliary effects output samples to chain input buffer for further processing 7198 // by insert effects 7199 effect->setOutBuffer(mInBuffer); 7200 } else { 7201 // Insert effects are inserted at the end of mEffects vector as they are processed 7202 // after track and auxiliary effects. 7203 // Insert effect order as a function of indicated preference: 7204 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7205 // another effect is present 7206 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7207 // last effect claiming first position 7208 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7209 // first effect claiming last position 7210 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7211 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7212 // already present 7213 7214 int size = (int)mEffects.size(); 7215 int idx_insert = size; 7216 int idx_insert_first = -1; 7217 int idx_insert_last = -1; 7218 7219 for (int i = 0; i < size; i++) { 7220 effect_descriptor_t d = mEffects[i]->desc(); 7221 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7222 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7223 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7224 // check invalid effect chaining combinations 7225 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7226 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7227 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7228 return INVALID_OPERATION; 7229 } 7230 // remember position of first insert effect and by default 7231 // select this as insert position for new effect 7232 if (idx_insert == size) { 7233 idx_insert = i; 7234 } 7235 // remember position of last insert effect claiming 7236 // first position 7237 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7238 idx_insert_first = i; 7239 } 7240 // remember position of first insert effect claiming 7241 // last position 7242 if (iPref == EFFECT_FLAG_INSERT_LAST && 7243 idx_insert_last == -1) { 7244 idx_insert_last = i; 7245 } 7246 } 7247 } 7248 7249 // modify idx_insert from first position if needed 7250 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7251 if (idx_insert_last != -1) { 7252 idx_insert = idx_insert_last; 7253 } else { 7254 idx_insert = size; 7255 } 7256 } else { 7257 if (idx_insert_first != -1) { 7258 idx_insert = idx_insert_first + 1; 7259 } 7260 } 7261 7262 // always read samples from chain input buffer 7263 effect->setInBuffer(mInBuffer); 7264 7265 // if last effect in the chain, output samples to chain 7266 // output buffer, otherwise to chain input buffer 7267 if (idx_insert == size) { 7268 if (idx_insert != 0) { 7269 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7270 mEffects[idx_insert-1]->configure(); 7271 } 7272 effect->setOutBuffer(mOutBuffer); 7273 } else { 7274 effect->setOutBuffer(mInBuffer); 7275 } 7276 mEffects.insertAt(effect, idx_insert); 7277 7278 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7279 } 7280 effect->configure(); 7281 return NO_ERROR; 7282} 7283 7284// removeEffect_l() must be called with PlaybackThread::mLock held 7285size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7286{ 7287 Mutex::Autolock _l(mLock); 7288 int size = (int)mEffects.size(); 7289 int i; 7290 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7291 7292 for (i = 0; i < size; i++) { 7293 if (effect == mEffects[i]) { 7294 // calling stop here will remove pre-processing effect from the audio HAL. 7295 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7296 // the middle of a read from audio HAL 7297 if (mEffects[i]->state() == EffectModule::ACTIVE || 7298 mEffects[i]->state() == EffectModule::STOPPING) { 7299 mEffects[i]->stop(); 7300 } 7301 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7302 delete[] effect->inBuffer(); 7303 } else { 7304 if (i == size - 1 && i != 0) { 7305 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7306 mEffects[i - 1]->configure(); 7307 } 7308 } 7309 mEffects.removeAt(i); 7310 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7311 break; 7312 } 7313 } 7314 7315 return mEffects.size(); 7316} 7317 7318// setDevice_l() must be called with PlaybackThread::mLock held 7319void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7320{ 7321 size_t size = mEffects.size(); 7322 for (size_t i = 0; i < size; i++) { 7323 mEffects[i]->setDevice(device); 7324 } 7325} 7326 7327// setMode_l() must be called with PlaybackThread::mLock held 7328void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7329{ 7330 size_t size = mEffects.size(); 7331 for (size_t i = 0; i < size; i++) { 7332 mEffects[i]->setMode(mode); 7333 } 7334} 7335 7336// setVolume_l() must be called with PlaybackThread::mLock held 7337bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7338{ 7339 uint32_t newLeft = *left; 7340 uint32_t newRight = *right; 7341 bool hasControl = false; 7342 int ctrlIdx = -1; 7343 size_t size = mEffects.size(); 7344 7345 // first update volume controller 7346 for (size_t i = size; i > 0; i--) { 7347 if (mEffects[i - 1]->isProcessEnabled() && 7348 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7349 ctrlIdx = i - 1; 7350 hasControl = true; 7351 break; 7352 } 7353 } 7354 7355 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7356 if (hasControl) { 7357 *left = mNewLeftVolume; 7358 *right = mNewRightVolume; 7359 } 7360 return hasControl; 7361 } 7362 7363 mVolumeCtrlIdx = ctrlIdx; 7364 mLeftVolume = newLeft; 7365 mRightVolume = newRight; 7366 7367 // second get volume update from volume controller 7368 if (ctrlIdx >= 0) { 7369 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7370 mNewLeftVolume = newLeft; 7371 mNewRightVolume = newRight; 7372 } 7373 // then indicate volume to all other effects in chain. 7374 // Pass altered volume to effects before volume controller 7375 // and requested volume to effects after controller 7376 uint32_t lVol = newLeft; 7377 uint32_t rVol = newRight; 7378 7379 for (size_t i = 0; i < size; i++) { 7380 if ((int)i == ctrlIdx) continue; 7381 // this also works for ctrlIdx == -1 when there is no volume controller 7382 if ((int)i > ctrlIdx) { 7383 lVol = *left; 7384 rVol = *right; 7385 } 7386 mEffects[i]->setVolume(&lVol, &rVol, false); 7387 } 7388 *left = newLeft; 7389 *right = newRight; 7390 7391 return hasControl; 7392} 7393 7394status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7395{ 7396 const size_t SIZE = 256; 7397 char buffer[SIZE]; 7398 String8 result; 7399 7400 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7401 result.append(buffer); 7402 7403 bool locked = tryLock(mLock); 7404 // failed to lock - AudioFlinger is probably deadlocked 7405 if (!locked) { 7406 result.append("\tCould not lock mutex:\n"); 7407 } 7408 7409 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7410 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7411 mEffects.size(), 7412 (uint32_t)mInBuffer, 7413 (uint32_t)mOutBuffer, 7414 mActiveTrackCnt); 7415 result.append(buffer); 7416 write(fd, result.string(), result.size()); 7417 7418 for (size_t i = 0; i < mEffects.size(); ++i) { 7419 sp<EffectModule> effect = mEffects[i]; 7420 if (effect != 0) { 7421 effect->dump(fd, args); 7422 } 7423 } 7424 7425 if (locked) { 7426 mLock.unlock(); 7427 } 7428 7429 return NO_ERROR; 7430} 7431 7432// must be called with ThreadBase::mLock held 7433void AudioFlinger::EffectChain::setEffectSuspended_l( 7434 const effect_uuid_t *type, bool suspend) 7435{ 7436 sp<SuspendedEffectDesc> desc; 7437 // use effect type UUID timelow as key as there is no real risk of identical 7438 // timeLow fields among effect type UUIDs. 7439 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7440 if (suspend) { 7441 if (index >= 0) { 7442 desc = mSuspendedEffects.valueAt(index); 7443 } else { 7444 desc = new SuspendedEffectDesc(); 7445 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7446 mSuspendedEffects.add(type->timeLow, desc); 7447 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7448 } 7449 if (desc->mRefCount++ == 0) { 7450 sp<EffectModule> effect = getEffectIfEnabled(type); 7451 if (effect != 0) { 7452 desc->mEffect = effect; 7453 effect->setSuspended(true); 7454 effect->setEnabled(false); 7455 } 7456 } 7457 } else { 7458 if (index < 0) { 7459 return; 7460 } 7461 desc = mSuspendedEffects.valueAt(index); 7462 if (desc->mRefCount <= 0) { 7463 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7464 desc->mRefCount = 1; 7465 } 7466 if (--desc->mRefCount == 0) { 7467 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7468 if (desc->mEffect != 0) { 7469 sp<EffectModule> effect = desc->mEffect.promote(); 7470 if (effect != 0) { 7471 effect->setSuspended(false); 7472 sp<EffectHandle> handle = effect->controlHandle(); 7473 if (handle != 0) { 7474 effect->setEnabled(handle->enabled()); 7475 } 7476 } 7477 desc->mEffect.clear(); 7478 } 7479 mSuspendedEffects.removeItemsAt(index); 7480 } 7481 } 7482} 7483 7484// must be called with ThreadBase::mLock held 7485void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7486{ 7487 sp<SuspendedEffectDesc> desc; 7488 7489 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7490 if (suspend) { 7491 if (index >= 0) { 7492 desc = mSuspendedEffects.valueAt(index); 7493 } else { 7494 desc = new SuspendedEffectDesc(); 7495 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7496 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7497 } 7498 if (desc->mRefCount++ == 0) { 7499 Vector< sp<EffectModule> > effects; 7500 getSuspendEligibleEffects(effects); 7501 for (size_t i = 0; i < effects.size(); i++) { 7502 setEffectSuspended_l(&effects[i]->desc().type, true); 7503 } 7504 } 7505 } else { 7506 if (index < 0) { 7507 return; 7508 } 7509 desc = mSuspendedEffects.valueAt(index); 7510 if (desc->mRefCount <= 0) { 7511 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7512 desc->mRefCount = 1; 7513 } 7514 if (--desc->mRefCount == 0) { 7515 Vector<const effect_uuid_t *> types; 7516 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7517 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7518 continue; 7519 } 7520 types.add(&mSuspendedEffects.valueAt(i)->mType); 7521 } 7522 for (size_t i = 0; i < types.size(); i++) { 7523 setEffectSuspended_l(types[i], false); 7524 } 7525 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7526 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7527 } 7528 } 7529} 7530 7531 7532// The volume effect is used for automated tests only 7533#ifndef OPENSL_ES_H_ 7534static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7535 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7536const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7537#endif //OPENSL_ES_H_ 7538 7539bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7540{ 7541 // auxiliary effects and visualizer are never suspended on output mix 7542 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7543 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7544 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7545 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7546 return false; 7547 } 7548 return true; 7549} 7550 7551void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 7552{ 7553 effects.clear(); 7554 for (size_t i = 0; i < mEffects.size(); i++) { 7555 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 7556 effects.add(mEffects[i]); 7557 } 7558 } 7559} 7560 7561sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7562 const effect_uuid_t *type) 7563{ 7564 sp<EffectModule> effect = getEffectFromType_l(type); 7565 return effect != 0 && effect->isEnabled() ? effect : 0; 7566} 7567 7568void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7569 bool enabled) 7570{ 7571 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7572 if (enabled) { 7573 if (index < 0) { 7574 // if the effect is not suspend check if all effects are suspended 7575 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7576 if (index < 0) { 7577 return; 7578 } 7579 if (!isEffectEligibleForSuspend(effect->desc())) { 7580 return; 7581 } 7582 setEffectSuspended_l(&effect->desc().type, enabled); 7583 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7584 if (index < 0) { 7585 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7586 return; 7587 } 7588 } 7589 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7590 effect->desc().type.timeLow); 7591 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7592 // if effect is requested to suspended but was not yet enabled, supend it now. 7593 if (desc->mEffect == 0) { 7594 desc->mEffect = effect; 7595 effect->setEnabled(false); 7596 effect->setSuspended(true); 7597 } 7598 } else { 7599 if (index < 0) { 7600 return; 7601 } 7602 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7603 effect->desc().type.timeLow); 7604 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7605 desc->mEffect.clear(); 7606 effect->setSuspended(false); 7607 } 7608} 7609 7610#undef LOG_TAG 7611#define LOG_TAG "AudioFlinger" 7612 7613// ---------------------------------------------------------------------------- 7614 7615status_t AudioFlinger::onTransact( 7616 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7617{ 7618 return BnAudioFlinger::onTransact(code, data, reply, flags); 7619} 7620 7621}; // namespace android 7622