AudioFlinger.cpp revision 2b213bc220768d2b984239511cd4554a96bc0079
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 == NULL) { 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 if (mPrimaryHardwareDev == NULL) { 199 ALOGE("Primary audio interface not found"); 200 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 201 } 202 203 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 204 // primary HW dev is selected can change so these conditions might not always be equivalent. 205 // When that happens, re-visit all the code that assumes this. 206 207 AutoMutex lock(mHardwareLock); 208 209 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 210 audio_hw_device_t *dev = mAudioHwDevs[i]; 211 212 mHardwareStatus = AUDIO_HW_INIT; 213 rc = dev->init_check(dev); 214 mHardwareStatus = AUDIO_HW_IDLE; 215 if (rc == 0) { 216 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 217 mHardwareStatus = AUDIO_HW_SET_MODE; 218 dev->set_mode(dev, mMode); 219 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 220 dev->set_master_volume(dev, 1.0f); 221 mHardwareStatus = AUDIO_HW_IDLE; 222 } 223 } 224} 225 226AudioFlinger::~AudioFlinger() 227{ 228 229 while (!mRecordThreads.isEmpty()) { 230 // closeInput() will remove first entry from mRecordThreads 231 closeInput(mRecordThreads.keyAt(0)); 232 } 233 while (!mPlaybackThreads.isEmpty()) { 234 // closeOutput() will remove first entry from mPlaybackThreads 235 closeOutput(mPlaybackThreads.keyAt(0)); 236 } 237 238 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 239 // no mHardwareLock needed, as there are no other references to this 240 audio_hw_device_close(mAudioHwDevs[i]); 241 } 242} 243 244audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 245{ 246 /* first matching HW device is returned */ 247 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 248 audio_hw_device_t *dev = mAudioHwDevs[i]; 249 if ((dev->get_supported_devices(dev) & devices) == devices) 250 return dev; 251 } 252 return NULL; 253} 254 255status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 256{ 257 const size_t SIZE = 256; 258 char buffer[SIZE]; 259 String8 result; 260 261 result.append("Clients:\n"); 262 for (size_t i = 0; i < mClients.size(); ++i) { 263 sp<Client> client = mClients.valueAt(i).promote(); 264 if (client != 0) { 265 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 266 result.append(buffer); 267 } 268 } 269 270 result.append("Global session refs:\n"); 271 result.append(" session pid cnt\n"); 272 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 273 AudioSessionRef *r = mAudioSessionRefs[i]; 274 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 275 result.append(buffer); 276 } 277 write(fd, result.string(), result.size()); 278 return NO_ERROR; 279} 280 281 282status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 283{ 284 const size_t SIZE = 256; 285 char buffer[SIZE]; 286 String8 result; 287 hardware_call_state hardwareStatus = mHardwareStatus; 288 289 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 290 result.append(buffer); 291 write(fd, result.string(), result.size()); 292 return NO_ERROR; 293} 294 295status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 296{ 297 const size_t SIZE = 256; 298 char buffer[SIZE]; 299 String8 result; 300 snprintf(buffer, SIZE, "Permission Denial: " 301 "can't dump AudioFlinger from pid=%d, uid=%d\n", 302 IPCThreadState::self()->getCallingPid(), 303 IPCThreadState::self()->getCallingUid()); 304 result.append(buffer); 305 write(fd, result.string(), result.size()); 306 return NO_ERROR; 307} 308 309static bool tryLock(Mutex& mutex) 310{ 311 bool locked = false; 312 for (int i = 0; i < kDumpLockRetries; ++i) { 313 if (mutex.tryLock() == NO_ERROR) { 314 locked = true; 315 break; 316 } 317 usleep(kDumpLockSleepUs); 318 } 319 return locked; 320} 321 322status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 323{ 324 if (!checkCallingPermission(String16("android.permission.DUMP"))) { 325 dumpPermissionDenial(fd, args); 326 } else { 327 // get state of hardware lock 328 bool hardwareLocked = tryLock(mHardwareLock); 329 if (!hardwareLocked) { 330 String8 result(kHardwareLockedString); 331 write(fd, result.string(), result.size()); 332 } else { 333 mHardwareLock.unlock(); 334 } 335 336 bool locked = tryLock(mLock); 337 338 // failed to lock - AudioFlinger is probably deadlocked 339 if (!locked) { 340 String8 result(kDeadlockedString); 341 write(fd, result.string(), result.size()); 342 } 343 344 dumpClients(fd, args); 345 dumpInternals(fd, args); 346 347 // dump playback threads 348 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 349 mPlaybackThreads.valueAt(i)->dump(fd, args); 350 } 351 352 // dump record threads 353 for (size_t i = 0; i < mRecordThreads.size(); i++) { 354 mRecordThreads.valueAt(i)->dump(fd, args); 355 } 356 357 // dump all hardware devs 358 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 359 audio_hw_device_t *dev = mAudioHwDevs[i]; 360 dev->dump(dev, fd); 361 } 362 if (locked) mLock.unlock(); 363 } 364 return NO_ERROR; 365} 366 367sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 368{ 369 // If pid is already in the mClients wp<> map, then use that entry 370 // (for which promote() is always != 0), otherwise create a new entry and Client. 371 sp<Client> client = mClients.valueFor(pid).promote(); 372 if (client == 0) { 373 client = new Client(this, pid); 374 mClients.add(pid, client); 375 } 376 377 return client; 378} 379 380// IAudioFlinger interface 381 382 383sp<IAudioTrack> AudioFlinger::createTrack( 384 pid_t pid, 385 audio_stream_type_t streamType, 386 uint32_t sampleRate, 387 audio_format_t format, 388 uint32_t channelMask, 389 int frameCount, 390 uint32_t flags, 391 const sp<IMemory>& sharedBuffer, 392 audio_io_handle_t output, 393 int *sessionId, 394 status_t *status) 395{ 396 sp<PlaybackThread::Track> track; 397 sp<TrackHandle> trackHandle; 398 sp<Client> client; 399 status_t lStatus; 400 int lSessionId; 401 402 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 403 // but if someone uses binder directly they could bypass that and cause us to crash 404 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 405 ALOGE("createTrack() invalid stream type %d", streamType); 406 lStatus = BAD_VALUE; 407 goto Exit; 408 } 409 410 { 411 Mutex::Autolock _l(mLock); 412 PlaybackThread *thread = checkPlaybackThread_l(output); 413 PlaybackThread *effectThread = NULL; 414 if (thread == NULL) { 415 ALOGE("unknown output thread"); 416 lStatus = BAD_VALUE; 417 goto Exit; 418 } 419 420 client = registerPid_l(pid); 421 422 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 423 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 424 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 425 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 426 if (mPlaybackThreads.keyAt(i) != output) { 427 // prevent same audio session on different output threads 428 uint32_t sessions = t->hasAudioSession(*sessionId); 429 if (sessions & PlaybackThread::TRACK_SESSION) { 430 ALOGE("createTrack() session ID %d already in use", *sessionId); 431 lStatus = BAD_VALUE; 432 goto Exit; 433 } 434 // check if an effect with same session ID is waiting for a track to be created 435 if (sessions & PlaybackThread::EFFECT_SESSION) { 436 effectThread = t.get(); 437 } 438 } 439 } 440 lSessionId = *sessionId; 441 } else { 442 // if no audio session id is provided, create one here 443 lSessionId = nextUniqueId(); 444 if (sessionId != NULL) { 445 *sessionId = lSessionId; 446 } 447 } 448 ALOGV("createTrack() lSessionId: %d", lSessionId); 449 450 track = thread->createTrack_l(client, streamType, sampleRate, format, 451 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 452 453 // move effect chain to this output thread if an effect on same session was waiting 454 // for a track to be created 455 if (lStatus == NO_ERROR && effectThread != NULL) { 456 Mutex::Autolock _dl(thread->mLock); 457 Mutex::Autolock _sl(effectThread->mLock); 458 moveEffectChain_l(lSessionId, effectThread, thread, true); 459 } 460 } 461 if (lStatus == NO_ERROR) { 462 trackHandle = new TrackHandle(track); 463 } else { 464 // remove local strong reference to Client before deleting the Track so that the Client 465 // destructor is called by the TrackBase destructor with mLock held 466 client.clear(); 467 track.clear(); 468 } 469 470Exit: 471 if(status) { 472 *status = lStatus; 473 } 474 return trackHandle; 475} 476 477uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 478{ 479 Mutex::Autolock _l(mLock); 480 PlaybackThread *thread = checkPlaybackThread_l(output); 481 if (thread == NULL) { 482 ALOGW("sampleRate() unknown thread %d", output); 483 return 0; 484 } 485 return thread->sampleRate(); 486} 487 488int AudioFlinger::channelCount(audio_io_handle_t output) const 489{ 490 Mutex::Autolock _l(mLock); 491 PlaybackThread *thread = checkPlaybackThread_l(output); 492 if (thread == NULL) { 493 ALOGW("channelCount() unknown thread %d", output); 494 return 0; 495 } 496 return thread->channelCount(); 497} 498 499audio_format_t AudioFlinger::format(audio_io_handle_t output) const 500{ 501 Mutex::Autolock _l(mLock); 502 PlaybackThread *thread = checkPlaybackThread_l(output); 503 if (thread == NULL) { 504 ALOGW("format() unknown thread %d", output); 505 return AUDIO_FORMAT_INVALID; 506 } 507 return thread->format(); 508} 509 510size_t AudioFlinger::frameCount(audio_io_handle_t output) const 511{ 512 Mutex::Autolock _l(mLock); 513 PlaybackThread *thread = checkPlaybackThread_l(output); 514 if (thread == NULL) { 515 ALOGW("frameCount() unknown thread %d", output); 516 return 0; 517 } 518 return thread->frameCount(); 519} 520 521uint32_t AudioFlinger::latency(audio_io_handle_t output) const 522{ 523 Mutex::Autolock _l(mLock); 524 PlaybackThread *thread = checkPlaybackThread_l(output); 525 if (thread == NULL) { 526 ALOGW("latency() unknown thread %d", output); 527 return 0; 528 } 529 return thread->latency(); 530} 531 532status_t AudioFlinger::setMasterVolume(float value) 533{ 534 status_t ret = initCheck(); 535 if (ret != NO_ERROR) { 536 return ret; 537 } 538 539 // check calling permissions 540 if (!settingsAllowed()) { 541 return PERMISSION_DENIED; 542 } 543 544 // when hw supports master volume, don't scale in sw mixer 545 { // scope for the lock 546 AutoMutex lock(mHardwareLock); 547 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 548 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 549 value = 1.0f; 550 } 551 mHardwareStatus = AUDIO_HW_IDLE; 552 } 553 554 Mutex::Autolock _l(mLock); 555 mMasterVolume = value; 556 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 557 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 558 559 return NO_ERROR; 560} 561 562status_t AudioFlinger::setMode(audio_mode_t mode) 563{ 564 status_t ret = initCheck(); 565 if (ret != NO_ERROR) { 566 return ret; 567 } 568 569 // check calling permissions 570 if (!settingsAllowed()) { 571 return PERMISSION_DENIED; 572 } 573 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 574 ALOGW("Illegal value: setMode(%d)", mode); 575 return BAD_VALUE; 576 } 577 578 { // scope for the lock 579 AutoMutex lock(mHardwareLock); 580 mHardwareStatus = AUDIO_HW_SET_MODE; 581 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 582 mHardwareStatus = AUDIO_HW_IDLE; 583 } 584 585 if (NO_ERROR == ret) { 586 Mutex::Autolock _l(mLock); 587 mMode = mode; 588 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 589 mPlaybackThreads.valueAt(i)->setMode(mode); 590 } 591 592 return ret; 593} 594 595status_t AudioFlinger::setMicMute(bool state) 596{ 597 status_t ret = initCheck(); 598 if (ret != NO_ERROR) { 599 return ret; 600 } 601 602 // check calling permissions 603 if (!settingsAllowed()) { 604 return PERMISSION_DENIED; 605 } 606 607 AutoMutex lock(mHardwareLock); 608 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 609 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 610 mHardwareStatus = AUDIO_HW_IDLE; 611 return ret; 612} 613 614bool AudioFlinger::getMicMute() const 615{ 616 status_t ret = initCheck(); 617 if (ret != NO_ERROR) { 618 return false; 619 } 620 621 bool state = AUDIO_MODE_INVALID; 622 AutoMutex lock(mHardwareLock); 623 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 624 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 625 mHardwareStatus = AUDIO_HW_IDLE; 626 return state; 627} 628 629status_t AudioFlinger::setMasterMute(bool muted) 630{ 631 // check calling permissions 632 if (!settingsAllowed()) { 633 return PERMISSION_DENIED; 634 } 635 636 Mutex::Autolock _l(mLock); 637 mMasterMute = muted; 638 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 639 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 640 641 return NO_ERROR; 642} 643 644float AudioFlinger::masterVolume() const 645{ 646 Mutex::Autolock _l(mLock); 647 return masterVolume_l(); 648} 649 650bool AudioFlinger::masterMute() const 651{ 652 Mutex::Autolock _l(mLock); 653 return masterMute_l(); 654} 655 656status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 657 audio_io_handle_t 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, audio_io_handle_t 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(audio_io_handle_t ioHandle, const String8& keyValuePairs) 743{ 744 status_t result; 745 746 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %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(audio_io_handle_t ioHandle, const String8& keys) const 818{ 819// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %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 AutoMutex lock(mHardwareLock); 855 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 856 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 857 mHardwareStatus = AUDIO_HW_IDLE; 858 return size; 859} 860 861unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 862{ 863 if (ioHandle == 0) { 864 return 0; 865 } 866 867 Mutex::Autolock _l(mLock); 868 869 RecordThread *recordThread = checkRecordThread_l(ioHandle); 870 if (recordThread != NULL) { 871 return recordThread->getInputFramesLost(); 872 } 873 return 0; 874} 875 876status_t AudioFlinger::setVoiceVolume(float value) 877{ 878 status_t ret = initCheck(); 879 if (ret != NO_ERROR) { 880 return ret; 881 } 882 883 // check calling permissions 884 if (!settingsAllowed()) { 885 return PERMISSION_DENIED; 886 } 887 888 AutoMutex lock(mHardwareLock); 889 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 890 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 891 mHardwareStatus = AUDIO_HW_IDLE; 892 893 return ret; 894} 895 896status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 897 audio_io_handle_t output) const 898{ 899 status_t status; 900 901 Mutex::Autolock _l(mLock); 902 903 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 904 if (playbackThread != NULL) { 905 return playbackThread->getRenderPosition(halFrames, dspFrames); 906 } 907 908 return BAD_VALUE; 909} 910 911void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 912{ 913 914 Mutex::Autolock _l(mLock); 915 916 pid_t pid = IPCThreadState::self()->getCallingPid(); 917 if (mNotificationClients.indexOfKey(pid) < 0) { 918 sp<NotificationClient> notificationClient = new NotificationClient(this, 919 client, 920 pid); 921 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 922 923 mNotificationClients.add(pid, notificationClient); 924 925 sp<IBinder> binder = client->asBinder(); 926 binder->linkToDeath(notificationClient); 927 928 // the config change is always sent from playback or record threads to avoid deadlock 929 // with AudioSystem::gLock 930 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 931 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 932 } 933 934 for (size_t i = 0; i < mRecordThreads.size(); i++) { 935 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 936 } 937 } 938} 939 940void AudioFlinger::removeNotificationClient(pid_t pid) 941{ 942 Mutex::Autolock _l(mLock); 943 944 int index = mNotificationClients.indexOfKey(pid); 945 if (index >= 0) { 946 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 947 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 948 mNotificationClients.removeItem(pid); 949 } 950 951 ALOGV("%d died, releasing its sessions", pid); 952 int num = mAudioSessionRefs.size(); 953 bool removed = false; 954 for (int i = 0; i< num; i++) { 955 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 956 ALOGV(" pid %d @ %d", ref->pid, i); 957 if (ref->pid == pid) { 958 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 959 mAudioSessionRefs.removeAt(i); 960 delete ref; 961 removed = true; 962 i--; 963 num--; 964 } 965 } 966 if (removed) { 967 purgeStaleEffects_l(); 968 } 969} 970 971// audioConfigChanged_l() must be called with AudioFlinger::mLock held 972void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 973{ 974 size_t size = mNotificationClients.size(); 975 for (size_t i = 0; i < size; i++) { 976 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 977 param2); 978 } 979} 980 981// removeClient_l() must be called with AudioFlinger::mLock held 982void AudioFlinger::removeClient_l(pid_t pid) 983{ 984 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 985 mClients.removeItem(pid); 986} 987 988 989// ---------------------------------------------------------------------------- 990 991AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 992 uint32_t device, type_t type) 993 : Thread(false), 994 mType(type), 995 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 996 // mChannelMask 997 mChannelCount(0), 998 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 999 mParamStatus(NO_ERROR), 1000 mStandby(false), mId(id), 1001 mDevice(device), 1002 mDeathRecipient(new PMDeathRecipient(this)) 1003{ 1004} 1005 1006AudioFlinger::ThreadBase::~ThreadBase() 1007{ 1008 mParamCond.broadcast(); 1009 // do not lock the mutex in destructor 1010 releaseWakeLock_l(); 1011 if (mPowerManager != 0) { 1012 sp<IBinder> binder = mPowerManager->asBinder(); 1013 binder->unlinkToDeath(mDeathRecipient); 1014 } 1015} 1016 1017void AudioFlinger::ThreadBase::exit() 1018{ 1019 ALOGV("ThreadBase::exit"); 1020 { 1021 // This lock prevents the following race in thread (uniprocessor for illustration): 1022 // if (!exitPending()) { 1023 // // context switch from here to exit() 1024 // // exit() calls requestExit(), what exitPending() observes 1025 // // exit() calls signal(), which is dropped since no waiters 1026 // // context switch back from exit() to here 1027 // mWaitWorkCV.wait(...); 1028 // // now thread is hung 1029 // } 1030 AutoMutex lock(mLock); 1031 requestExit(); 1032 mWaitWorkCV.signal(); 1033 } 1034 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1035 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1036 requestExitAndWait(); 1037} 1038 1039status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1040{ 1041 status_t status; 1042 1043 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1044 Mutex::Autolock _l(mLock); 1045 1046 mNewParameters.add(keyValuePairs); 1047 mWaitWorkCV.signal(); 1048 // wait condition with timeout in case the thread loop has exited 1049 // before the request could be processed 1050 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1051 status = mParamStatus; 1052 mWaitWorkCV.signal(); 1053 } else { 1054 status = TIMED_OUT; 1055 } 1056 return status; 1057} 1058 1059void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1060{ 1061 Mutex::Autolock _l(mLock); 1062 sendConfigEvent_l(event, param); 1063} 1064 1065// sendConfigEvent_l() must be called with ThreadBase::mLock held 1066void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1067{ 1068 ConfigEvent configEvent; 1069 configEvent.mEvent = event; 1070 configEvent.mParam = param; 1071 mConfigEvents.add(configEvent); 1072 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1073 mWaitWorkCV.signal(); 1074} 1075 1076void AudioFlinger::ThreadBase::processConfigEvents() 1077{ 1078 mLock.lock(); 1079 while(!mConfigEvents.isEmpty()) { 1080 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1081 ConfigEvent configEvent = mConfigEvents[0]; 1082 mConfigEvents.removeAt(0); 1083 // release mLock before locking AudioFlinger mLock: lock order is always 1084 // AudioFlinger then ThreadBase to avoid cross deadlock 1085 mLock.unlock(); 1086 mAudioFlinger->mLock.lock(); 1087 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1088 mAudioFlinger->mLock.unlock(); 1089 mLock.lock(); 1090 } 1091 mLock.unlock(); 1092} 1093 1094status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1095{ 1096 const size_t SIZE = 256; 1097 char buffer[SIZE]; 1098 String8 result; 1099 1100 bool locked = tryLock(mLock); 1101 if (!locked) { 1102 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1103 write(fd, buffer, strlen(buffer)); 1104 } 1105 1106 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1107 result.append(buffer); 1108 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1109 result.append(buffer); 1110 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1111 result.append(buffer); 1112 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1113 result.append(buffer); 1114 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1115 result.append(buffer); 1116 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1117 result.append(buffer); 1118 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1119 result.append(buffer); 1120 1121 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1122 result.append(buffer); 1123 result.append(" Index Command"); 1124 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1125 snprintf(buffer, SIZE, "\n %02d ", i); 1126 result.append(buffer); 1127 result.append(mNewParameters[i]); 1128 } 1129 1130 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1131 result.append(buffer); 1132 snprintf(buffer, SIZE, " Index event param\n"); 1133 result.append(buffer); 1134 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1135 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1136 result.append(buffer); 1137 } 1138 result.append("\n"); 1139 1140 write(fd, result.string(), result.size()); 1141 1142 if (locked) { 1143 mLock.unlock(); 1144 } 1145 return NO_ERROR; 1146} 1147 1148status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1149{ 1150 const size_t SIZE = 256; 1151 char buffer[SIZE]; 1152 String8 result; 1153 1154 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1155 write(fd, buffer, strlen(buffer)); 1156 1157 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1158 sp<EffectChain> chain = mEffectChains[i]; 1159 if (chain != 0) { 1160 chain->dump(fd, args); 1161 } 1162 } 1163 return NO_ERROR; 1164} 1165 1166void AudioFlinger::ThreadBase::acquireWakeLock() 1167{ 1168 Mutex::Autolock _l(mLock); 1169 acquireWakeLock_l(); 1170} 1171 1172void AudioFlinger::ThreadBase::acquireWakeLock_l() 1173{ 1174 if (mPowerManager == 0) { 1175 // use checkService() to avoid blocking if power service is not up yet 1176 sp<IBinder> binder = 1177 defaultServiceManager()->checkService(String16("power")); 1178 if (binder == 0) { 1179 ALOGW("Thread %s cannot connect to the power manager service", mName); 1180 } else { 1181 mPowerManager = interface_cast<IPowerManager>(binder); 1182 binder->linkToDeath(mDeathRecipient); 1183 } 1184 } 1185 if (mPowerManager != 0) { 1186 sp<IBinder> binder = new BBinder(); 1187 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1188 binder, 1189 String16(mName)); 1190 if (status == NO_ERROR) { 1191 mWakeLockToken = binder; 1192 } 1193 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1194 } 1195} 1196 1197void AudioFlinger::ThreadBase::releaseWakeLock() 1198{ 1199 Mutex::Autolock _l(mLock); 1200 releaseWakeLock_l(); 1201} 1202 1203void AudioFlinger::ThreadBase::releaseWakeLock_l() 1204{ 1205 if (mWakeLockToken != 0) { 1206 ALOGV("releaseWakeLock_l() %s", mName); 1207 if (mPowerManager != 0) { 1208 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1209 } 1210 mWakeLockToken.clear(); 1211 } 1212} 1213 1214void AudioFlinger::ThreadBase::clearPowerManager() 1215{ 1216 Mutex::Autolock _l(mLock); 1217 releaseWakeLock_l(); 1218 mPowerManager.clear(); 1219} 1220 1221void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1222{ 1223 sp<ThreadBase> thread = mThread.promote(); 1224 if (thread != 0) { 1225 thread->clearPowerManager(); 1226 } 1227 ALOGW("power manager service died !!!"); 1228} 1229 1230void AudioFlinger::ThreadBase::setEffectSuspended( 1231 const effect_uuid_t *type, bool suspend, int sessionId) 1232{ 1233 Mutex::Autolock _l(mLock); 1234 setEffectSuspended_l(type, suspend, sessionId); 1235} 1236 1237void AudioFlinger::ThreadBase::setEffectSuspended_l( 1238 const effect_uuid_t *type, bool suspend, int sessionId) 1239{ 1240 sp<EffectChain> chain = getEffectChain_l(sessionId); 1241 if (chain != 0) { 1242 if (type != NULL) { 1243 chain->setEffectSuspended_l(type, suspend); 1244 } else { 1245 chain->setEffectSuspendedAll_l(suspend); 1246 } 1247 } 1248 1249 updateSuspendedSessions_l(type, suspend, sessionId); 1250} 1251 1252void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1253{ 1254 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1255 if (index < 0) { 1256 return; 1257 } 1258 1259 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1260 mSuspendedSessions.editValueAt(index); 1261 1262 for (size_t i = 0; i < sessionEffects.size(); i++) { 1263 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1264 for (int j = 0; j < desc->mRefCount; j++) { 1265 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1266 chain->setEffectSuspendedAll_l(true); 1267 } else { 1268 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1269 desc->mType.timeLow); 1270 chain->setEffectSuspended_l(&desc->mType, true); 1271 } 1272 } 1273 } 1274} 1275 1276void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1277 bool suspend, 1278 int sessionId) 1279{ 1280 int index = mSuspendedSessions.indexOfKey(sessionId); 1281 1282 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1283 1284 if (suspend) { 1285 if (index >= 0) { 1286 sessionEffects = mSuspendedSessions.editValueAt(index); 1287 } else { 1288 mSuspendedSessions.add(sessionId, sessionEffects); 1289 } 1290 } else { 1291 if (index < 0) { 1292 return; 1293 } 1294 sessionEffects = mSuspendedSessions.editValueAt(index); 1295 } 1296 1297 1298 int key = EffectChain::kKeyForSuspendAll; 1299 if (type != NULL) { 1300 key = type->timeLow; 1301 } 1302 index = sessionEffects.indexOfKey(key); 1303 1304 sp <SuspendedSessionDesc> desc; 1305 if (suspend) { 1306 if (index >= 0) { 1307 desc = sessionEffects.valueAt(index); 1308 } else { 1309 desc = new SuspendedSessionDesc(); 1310 if (type != NULL) { 1311 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1312 } 1313 sessionEffects.add(key, desc); 1314 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1315 } 1316 desc->mRefCount++; 1317 } else { 1318 if (index < 0) { 1319 return; 1320 } 1321 desc = sessionEffects.valueAt(index); 1322 if (--desc->mRefCount == 0) { 1323 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1324 sessionEffects.removeItemsAt(index); 1325 if (sessionEffects.isEmpty()) { 1326 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1327 sessionId); 1328 mSuspendedSessions.removeItem(sessionId); 1329 } 1330 } 1331 } 1332 if (!sessionEffects.isEmpty()) { 1333 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1334 } 1335} 1336 1337void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1338 bool enabled, 1339 int sessionId) 1340{ 1341 Mutex::Autolock _l(mLock); 1342 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1343} 1344 1345void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1346 bool enabled, 1347 int sessionId) 1348{ 1349 if (mType != RECORD) { 1350 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1351 // another session. This gives the priority to well behaved effect control panels 1352 // and applications not using global effects. 1353 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1354 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1355 } 1356 } 1357 1358 sp<EffectChain> chain = getEffectChain_l(sessionId); 1359 if (chain != 0) { 1360 chain->checkSuspendOnEffectEnabled(effect, enabled); 1361 } 1362} 1363 1364// ---------------------------------------------------------------------------- 1365 1366AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1367 AudioStreamOut* output, 1368 audio_io_handle_t id, 1369 uint32_t device, 1370 type_t type) 1371 : ThreadBase(audioFlinger, id, device, type), 1372 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1373 // Assumes constructor is called by AudioFlinger with it's mLock held, 1374 // but it would be safer to explicitly pass initial masterMute as parameter 1375 mMasterMute(audioFlinger->masterMute_l()), 1376 // mStreamTypes[] initialized in constructor body 1377 mOutput(output), 1378 // Assumes constructor is called by AudioFlinger with it's mLock held, 1379 // but it would be safer to explicitly pass initial masterVolume as parameter 1380 mMasterVolume(audioFlinger->masterVolume_l()), 1381 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1382{ 1383 snprintf(mName, kNameLength, "AudioOut_%d", id); 1384 1385 readOutputParameters(); 1386 1387 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1388 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1389 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1390 stream = (audio_stream_type_t) (stream + 1)) { 1391 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1392 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1393 // initialized by stream_type_t default constructor 1394 // mStreamTypes[stream].valid = true; 1395 } 1396} 1397 1398AudioFlinger::PlaybackThread::~PlaybackThread() 1399{ 1400 delete [] mMixBuffer; 1401} 1402 1403status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1404{ 1405 dumpInternals(fd, args); 1406 dumpTracks(fd, args); 1407 dumpEffectChains(fd, args); 1408 return NO_ERROR; 1409} 1410 1411status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1412{ 1413 const size_t SIZE = 256; 1414 char buffer[SIZE]; 1415 String8 result; 1416 1417 snprintf(buffer, SIZE, "Output thread %p 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 < mTracks.size(); ++i) { 1421 sp<Track> track = mTracks[i]; 1422 if (track != 0) { 1423 track->dump(buffer, SIZE); 1424 result.append(buffer); 1425 } 1426 } 1427 1428 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1429 result.append(buffer); 1430 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1431 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1432 sp<Track> track = mActiveTracks[i].promote(); 1433 if (track != 0) { 1434 track->dump(buffer, SIZE); 1435 result.append(buffer); 1436 } 1437 } 1438 write(fd, result.string(), result.size()); 1439 return NO_ERROR; 1440} 1441 1442status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1443{ 1444 const size_t SIZE = 256; 1445 char buffer[SIZE]; 1446 String8 result; 1447 1448 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1449 result.append(buffer); 1450 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1451 result.append(buffer); 1452 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1453 result.append(buffer); 1454 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1455 result.append(buffer); 1456 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1457 result.append(buffer); 1458 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1459 result.append(buffer); 1460 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1461 result.append(buffer); 1462 write(fd, result.string(), result.size()); 1463 1464 dumpBase(fd, args); 1465 1466 return NO_ERROR; 1467} 1468 1469// Thread virtuals 1470status_t AudioFlinger::PlaybackThread::readyToRun() 1471{ 1472 status_t status = initCheck(); 1473 if (status == NO_ERROR) { 1474 ALOGI("AudioFlinger's thread %p ready to run", this); 1475 } else { 1476 ALOGE("No working audio driver found."); 1477 } 1478 return status; 1479} 1480 1481void AudioFlinger::PlaybackThread::onFirstRef() 1482{ 1483 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1484} 1485 1486// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1487sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1488 const sp<AudioFlinger::Client>& client, 1489 audio_stream_type_t streamType, 1490 uint32_t sampleRate, 1491 audio_format_t format, 1492 uint32_t channelMask, 1493 int frameCount, 1494 const sp<IMemory>& sharedBuffer, 1495 int sessionId, 1496 status_t *status) 1497{ 1498 sp<Track> track; 1499 status_t lStatus; 1500 1501 if (mType == DIRECT) { 1502 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1503 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1504 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1505 "for output %p with format %d", 1506 sampleRate, format, channelMask, mOutput, mFormat); 1507 lStatus = BAD_VALUE; 1508 goto Exit; 1509 } 1510 } 1511 } else { 1512 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1513 if (sampleRate > mSampleRate*2) { 1514 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1515 lStatus = BAD_VALUE; 1516 goto Exit; 1517 } 1518 } 1519 1520 lStatus = initCheck(); 1521 if (lStatus != NO_ERROR) { 1522 ALOGE("Audio driver not initialized."); 1523 goto Exit; 1524 } 1525 1526 { // scope for mLock 1527 Mutex::Autolock _l(mLock); 1528 1529 // all tracks in same audio session must share the same routing strategy otherwise 1530 // conflicts will happen when tracks are moved from one output to another by audio policy 1531 // manager 1532 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1533 for (size_t i = 0; i < mTracks.size(); ++i) { 1534 sp<Track> t = mTracks[i]; 1535 if (t != 0) { 1536 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1537 if (sessionId == t->sessionId() && strategy != actual) { 1538 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1539 strategy, actual); 1540 lStatus = BAD_VALUE; 1541 goto Exit; 1542 } 1543 } 1544 } 1545 1546 track = new Track(this, client, streamType, sampleRate, format, 1547 channelMask, frameCount, sharedBuffer, sessionId); 1548 if (track->getCblk() == NULL || track->name() < 0) { 1549 lStatus = NO_MEMORY; 1550 goto Exit; 1551 } 1552 mTracks.add(track); 1553 1554 sp<EffectChain> chain = getEffectChain_l(sessionId); 1555 if (chain != 0) { 1556 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1557 track->setMainBuffer(chain->inBuffer()); 1558 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1559 chain->incTrackCnt(); 1560 } 1561 1562 // invalidate track immediately if the stream type was moved to another thread since 1563 // createTrack() was called by the client process. 1564 if (!mStreamTypes[streamType].valid) { 1565 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1566 this, streamType); 1567 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1568 } 1569 } 1570 lStatus = NO_ERROR; 1571 1572Exit: 1573 if(status) { 1574 *status = lStatus; 1575 } 1576 return track; 1577} 1578 1579uint32_t AudioFlinger::PlaybackThread::latency() const 1580{ 1581 Mutex::Autolock _l(mLock); 1582 if (initCheck() == NO_ERROR) { 1583 return mOutput->stream->get_latency(mOutput->stream); 1584 } else { 1585 return 0; 1586 } 1587} 1588 1589status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1590{ 1591 mMasterVolume = value; 1592 return NO_ERROR; 1593} 1594 1595status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1596{ 1597 mMasterMute = muted; 1598 return NO_ERROR; 1599} 1600 1601status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1602{ 1603 mStreamTypes[stream].volume = value; 1604 return NO_ERROR; 1605} 1606 1607status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1608{ 1609 mStreamTypes[stream].mute = muted; 1610 return NO_ERROR; 1611} 1612 1613float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1614{ 1615 return mStreamTypes[stream].volume; 1616} 1617 1618bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1619{ 1620 return mStreamTypes[stream].mute; 1621} 1622 1623// addTrack_l() must be called with ThreadBase::mLock held 1624status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1625{ 1626 status_t status = ALREADY_EXISTS; 1627 1628 // set retry count for buffer fill 1629 track->mRetryCount = kMaxTrackStartupRetries; 1630 if (mActiveTracks.indexOf(track) < 0) { 1631 // the track is newly added, make sure it fills up all its 1632 // buffers before playing. This is to ensure the client will 1633 // effectively get the latency it requested. 1634 track->mFillingUpStatus = Track::FS_FILLING; 1635 track->mResetDone = false; 1636 mActiveTracks.add(track); 1637 if (track->mainBuffer() != mMixBuffer) { 1638 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1639 if (chain != 0) { 1640 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1641 chain->incActiveTrackCnt(); 1642 } 1643 } 1644 1645 status = NO_ERROR; 1646 } 1647 1648 ALOGV("mWaitWorkCV.broadcast"); 1649 mWaitWorkCV.broadcast(); 1650 1651 return status; 1652} 1653 1654// destroyTrack_l() must be called with ThreadBase::mLock held 1655void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1656{ 1657 track->mState = TrackBase::TERMINATED; 1658 if (mActiveTracks.indexOf(track) < 0) { 1659 removeTrack_l(track); 1660 } 1661} 1662 1663void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1664{ 1665 mTracks.remove(track); 1666 deleteTrackName_l(track->name()); 1667 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1668 if (chain != 0) { 1669 chain->decTrackCnt(); 1670 } 1671} 1672 1673String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1674{ 1675 String8 out_s8 = String8(""); 1676 char *s; 1677 1678 Mutex::Autolock _l(mLock); 1679 if (initCheck() != NO_ERROR) { 1680 return out_s8; 1681 } 1682 1683 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1684 out_s8 = String8(s); 1685 free(s); 1686 return out_s8; 1687} 1688 1689// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1690void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1691 AudioSystem::OutputDescriptor desc; 1692 void *param2 = NULL; 1693 1694 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1695 1696 switch (event) { 1697 case AudioSystem::OUTPUT_OPENED: 1698 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1699 desc.channels = mChannelMask; 1700 desc.samplingRate = mSampleRate; 1701 desc.format = mFormat; 1702 desc.frameCount = mFrameCount; 1703 desc.latency = latency(); 1704 param2 = &desc; 1705 break; 1706 1707 case AudioSystem::STREAM_CONFIG_CHANGED: 1708 param2 = ¶m; 1709 case AudioSystem::OUTPUT_CLOSED: 1710 default: 1711 break; 1712 } 1713 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1714} 1715 1716void AudioFlinger::PlaybackThread::readOutputParameters() 1717{ 1718 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1719 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1720 mChannelCount = (uint16_t)popcount(mChannelMask); 1721 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1722 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1723 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1724 1725 // FIXME - Current mixer implementation only supports stereo output: Always 1726 // Allocate a stereo buffer even if HW output is mono. 1727 delete[] mMixBuffer; 1728 mMixBuffer = new int16_t[mFrameCount * 2]; 1729 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1730 1731 // force reconfiguration of effect chains and engines to take new buffer size and audio 1732 // parameters into account 1733 // Note that mLock is not held when readOutputParameters() is called from the constructor 1734 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1735 // matter. 1736 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1737 Vector< sp<EffectChain> > effectChains = mEffectChains; 1738 for (size_t i = 0; i < effectChains.size(); i ++) { 1739 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1740 } 1741} 1742 1743status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1744{ 1745 if (halFrames == NULL || dspFrames == NULL) { 1746 return BAD_VALUE; 1747 } 1748 Mutex::Autolock _l(mLock); 1749 if (initCheck() != NO_ERROR) { 1750 return INVALID_OPERATION; 1751 } 1752 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1753 1754 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1755} 1756 1757uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1758{ 1759 Mutex::Autolock _l(mLock); 1760 uint32_t result = 0; 1761 if (getEffectChain_l(sessionId) != 0) { 1762 result = EFFECT_SESSION; 1763 } 1764 1765 for (size_t i = 0; i < mTracks.size(); ++i) { 1766 sp<Track> track = mTracks[i]; 1767 if (sessionId == track->sessionId() && 1768 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1769 result |= TRACK_SESSION; 1770 break; 1771 } 1772 } 1773 1774 return result; 1775} 1776 1777uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1778{ 1779 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1780 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1781 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1782 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1783 } 1784 for (size_t i = 0; i < mTracks.size(); i++) { 1785 sp<Track> track = mTracks[i]; 1786 if (sessionId == track->sessionId() && 1787 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1788 return AudioSystem::getStrategyForStream(track->streamType()); 1789 } 1790 } 1791 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1792} 1793 1794 1795AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1796{ 1797 Mutex::Autolock _l(mLock); 1798 return mOutput; 1799} 1800 1801AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1802{ 1803 Mutex::Autolock _l(mLock); 1804 AudioStreamOut *output = mOutput; 1805 mOutput = NULL; 1806 return output; 1807} 1808 1809// this method must always be called either with ThreadBase mLock held or inside the thread loop 1810audio_stream_t* AudioFlinger::PlaybackThread::stream() 1811{ 1812 if (mOutput == NULL) { 1813 return NULL; 1814 } 1815 return &mOutput->stream->common; 1816} 1817 1818uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1819{ 1820 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1821 // decoding and transfer time. So sleeping for half of the latency would likely cause 1822 // underruns 1823 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1824 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1825 } else { 1826 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1827 } 1828} 1829 1830// ---------------------------------------------------------------------------- 1831 1832AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1833 audio_io_handle_t id, uint32_t device, type_t type) 1834 : PlaybackThread(audioFlinger, output, id, device, type), 1835 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1836 mPrevMixerStatus(MIXER_IDLE) 1837{ 1838 // FIXME - Current mixer implementation only supports stereo output 1839 if (mChannelCount == 1) { 1840 ALOGE("Invalid audio hardware channel count"); 1841 } 1842} 1843 1844AudioFlinger::MixerThread::~MixerThread() 1845{ 1846 delete mAudioMixer; 1847} 1848 1849bool AudioFlinger::MixerThread::threadLoop() 1850{ 1851 Vector< sp<Track> > tracksToRemove; 1852 mixer_state mixerStatus = MIXER_IDLE; 1853 nsecs_t standbyTime = systemTime(); 1854 size_t mixBufferSize = mFrameCount * mFrameSize; 1855 // FIXME: Relaxed timing because of a certain device that can't meet latency 1856 // Should be reduced to 2x after the vendor fixes the driver issue 1857 // increase threshold again due to low power audio mode. The way this warning threshold is 1858 // calculated and its usefulness should be reconsidered anyway. 1859 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1860 nsecs_t lastWarning = 0; 1861 bool longStandbyExit = false; 1862 uint32_t activeSleepTime = activeSleepTimeUs(); 1863 uint32_t idleSleepTime = idleSleepTimeUs(); 1864 uint32_t sleepTime = idleSleepTime; 1865 uint32_t sleepTimeShift = 0; 1866 Vector< sp<EffectChain> > effectChains; 1867#ifdef DEBUG_CPU_USAGE 1868 ThreadCpuUsage cpu; 1869 const CentralTendencyStatistics& stats = cpu.statistics(); 1870#endif 1871 1872 acquireWakeLock(); 1873 1874 while (!exitPending()) 1875 { 1876#ifdef DEBUG_CPU_USAGE 1877 cpu.sampleAndEnable(); 1878 unsigned n = stats.n(); 1879 // cpu.elapsed() is expensive, so don't call it every loop 1880 if ((n & 127) == 1) { 1881 long long elapsed = cpu.elapsed(); 1882 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1883 double perLoop = elapsed / (double) n; 1884 double perLoop100 = perLoop * 0.01; 1885 double mean = stats.mean(); 1886 double stddev = stats.stddev(); 1887 double minimum = stats.minimum(); 1888 double maximum = stats.maximum(); 1889 cpu.resetStatistics(); 1890 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", 1891 elapsed * .000000001, n, perLoop * .000001, 1892 mean * .001, 1893 stddev * .001, 1894 minimum * .001, 1895 maximum * .001, 1896 mean / perLoop100, 1897 stddev / perLoop100, 1898 minimum / perLoop100, 1899 maximum / perLoop100); 1900 } 1901 } 1902#endif 1903 processConfigEvents(); 1904 1905 mixerStatus = MIXER_IDLE; 1906 { // scope for mLock 1907 1908 Mutex::Autolock _l(mLock); 1909 1910 if (checkForNewParameters_l()) { 1911 mixBufferSize = mFrameCount * mFrameSize; 1912 // FIXME: Relaxed timing because of a certain device that can't meet latency 1913 // Should be reduced to 2x after the vendor fixes the driver issue 1914 // increase threshold again due to low power audio mode. The way this warning 1915 // threshold is calculated and its usefulness should be reconsidered anyway. 1916 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1917 activeSleepTime = activeSleepTimeUs(); 1918 idleSleepTime = idleSleepTimeUs(); 1919 } 1920 1921 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1922 1923 // put audio hardware into standby after short delay 1924 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1925 mSuspended)) { 1926 if (!mStandby) { 1927 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 1928 mOutput->stream->common.standby(&mOutput->stream->common); 1929 mStandby = true; 1930 mBytesWritten = 0; 1931 } 1932 1933 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1934 // we're about to wait, flush the binder command buffer 1935 IPCThreadState::self()->flushCommands(); 1936 1937 if (exitPending()) break; 1938 1939 releaseWakeLock_l(); 1940 // wait until we have something to do... 1941 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 1942 mWaitWorkCV.wait(mLock); 1943 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 1944 acquireWakeLock_l(); 1945 1946 mPrevMixerStatus = MIXER_IDLE; 1947 if (!mMasterMute) { 1948 char value[PROPERTY_VALUE_MAX]; 1949 property_get("ro.audio.silent", value, "0"); 1950 if (atoi(value)) { 1951 ALOGD("Silence is golden"); 1952 setMasterMute(true); 1953 } 1954 } 1955 1956 standbyTime = systemTime() + kStandbyTimeInNsecs; 1957 sleepTime = idleSleepTime; 1958 sleepTimeShift = 0; 1959 continue; 1960 } 1961 } 1962 1963 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1964 1965 // prevent any changes in effect chain list and in each effect chain 1966 // during mixing and effect process as the audio buffers could be deleted 1967 // or modified if an effect is created or deleted 1968 lockEffectChains_l(effectChains); 1969 } 1970 1971 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1972 // mix buffers... 1973 mAudioMixer->process(); 1974 // increase sleep time progressively when application underrun condition clears. 1975 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1976 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1977 // such that we would underrun the audio HAL. 1978 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 1979 sleepTimeShift--; 1980 } 1981 sleepTime = 0; 1982 standbyTime = systemTime() + kStandbyTimeInNsecs; 1983 //TODO: delay standby when effects have a tail 1984 } else { 1985 // If no tracks are ready, sleep once for the duration of an output 1986 // buffer size, then write 0s to the output 1987 if (sleepTime == 0) { 1988 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1989 sleepTime = activeSleepTime >> sleepTimeShift; 1990 if (sleepTime < kMinThreadSleepTimeUs) { 1991 sleepTime = kMinThreadSleepTimeUs; 1992 } 1993 // reduce sleep time in case of consecutive application underruns to avoid 1994 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 1995 // duration we would end up writing less data than needed by the audio HAL if 1996 // the condition persists. 1997 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 1998 sleepTimeShift++; 1999 } 2000 } else { 2001 sleepTime = idleSleepTime; 2002 } 2003 } else if (mBytesWritten != 0 || 2004 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2005 memset (mMixBuffer, 0, mixBufferSize); 2006 sleepTime = 0; 2007 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2008 } 2009 // TODO add standby time extension fct of effect tail 2010 } 2011 2012 if (mSuspended) { 2013 sleepTime = suspendSleepTimeUs(); 2014 } 2015 // sleepTime == 0 means we must write to audio hardware 2016 if (sleepTime == 0) { 2017 for (size_t i = 0; i < effectChains.size(); i ++) { 2018 effectChains[i]->process_l(); 2019 } 2020 // enable changes in effect chain 2021 unlockEffectChains(effectChains); 2022 mLastWriteTime = systemTime(); 2023 mInWrite = true; 2024 mBytesWritten += mixBufferSize; 2025 2026 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2027 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2028 mNumWrites++; 2029 mInWrite = false; 2030 nsecs_t now = systemTime(); 2031 nsecs_t delta = now - mLastWriteTime; 2032 if (!mStandby && delta > maxPeriod) { 2033 mNumDelayedWrites++; 2034 if ((now - lastWarning) > kWarningThrottleNs) { 2035 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2036 ns2ms(delta), mNumDelayedWrites, this); 2037 lastWarning = now; 2038 } 2039 if (mStandby) { 2040 longStandbyExit = true; 2041 } 2042 } 2043 mStandby = false; 2044 } else { 2045 // enable changes in effect chain 2046 unlockEffectChains(effectChains); 2047 usleep(sleepTime); 2048 } 2049 2050 // finally let go of all our tracks, without the lock held 2051 // since we can't guarantee the destructors won't acquire that 2052 // same lock. 2053 tracksToRemove.clear(); 2054 2055 // Effect chains will be actually deleted here if they were removed from 2056 // mEffectChains list during mixing or effects processing 2057 effectChains.clear(); 2058 } 2059 2060 if (!mStandby) { 2061 mOutput->stream->common.standby(&mOutput->stream->common); 2062 } 2063 2064 releaseWakeLock(); 2065 2066 ALOGV("MixerThread %p exiting", this); 2067 return false; 2068} 2069 2070// prepareTracks_l() must be called with ThreadBase::mLock held 2071AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2072 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2073{ 2074 2075 mixer_state mixerStatus = MIXER_IDLE; 2076 // find out which tracks need to be processed 2077 size_t count = activeTracks.size(); 2078 size_t mixedTracks = 0; 2079 size_t tracksWithEffect = 0; 2080 2081 float masterVolume = mMasterVolume; 2082 bool masterMute = mMasterMute; 2083 2084 if (masterMute) { 2085 masterVolume = 0; 2086 } 2087 // Delegate master volume control to effect in output mix effect chain if needed 2088 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2089 if (chain != 0) { 2090 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2091 chain->setVolume_l(&v, &v); 2092 masterVolume = (float)((v + (1 << 23)) >> 24); 2093 chain.clear(); 2094 } 2095 2096 for (size_t i=0 ; i<count ; i++) { 2097 sp<Track> t = activeTracks[i].promote(); 2098 if (t == 0) continue; 2099 2100 // this const just means the local variable doesn't change 2101 Track* const track = t.get(); 2102 audio_track_cblk_t* cblk = track->cblk(); 2103 2104 // The first time a track is added we wait 2105 // for all its buffers to be filled before processing it 2106 int name = track->name(); 2107 // make sure that we have enough frames to mix one full buffer. 2108 // enforce this condition only once to enable draining the buffer in case the client 2109 // app does not call stop() and relies on underrun to stop: 2110 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2111 // during last round 2112 uint32_t minFrames = 1; 2113 if (!track->isStopped() && !track->isPausing() && 2114 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2115 if (t->sampleRate() == (int)mSampleRate) { 2116 minFrames = mFrameCount; 2117 } else { 2118 // +1 for rounding and +1 for additional sample needed for interpolation 2119 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2120 // add frames already consumed but not yet released by the resampler 2121 // because cblk->framesReady() will include these frames 2122 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2123 // the minimum track buffer size is normally twice the number of frames necessary 2124 // to fill one buffer and the resampler should not leave more than one buffer worth 2125 // of unreleased frames after each pass, but just in case... 2126 ALOG_ASSERT(minFrames <= cblk->frameCount); 2127 } 2128 } 2129 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2130 !track->isPaused() && !track->isTerminated()) 2131 { 2132 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2133 2134 mixedTracks++; 2135 2136 // track->mainBuffer() != mMixBuffer means there is an effect chain 2137 // connected to the track 2138 chain.clear(); 2139 if (track->mainBuffer() != mMixBuffer) { 2140 chain = getEffectChain_l(track->sessionId()); 2141 // Delegate volume control to effect in track effect chain if needed 2142 if (chain != 0) { 2143 tracksWithEffect++; 2144 } else { 2145 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2146 name, track->sessionId()); 2147 } 2148 } 2149 2150 2151 int param = AudioMixer::VOLUME; 2152 if (track->mFillingUpStatus == Track::FS_FILLED) { 2153 // no ramp for the first volume setting 2154 track->mFillingUpStatus = Track::FS_ACTIVE; 2155 if (track->mState == TrackBase::RESUMING) { 2156 track->mState = TrackBase::ACTIVE; 2157 param = AudioMixer::RAMP_VOLUME; 2158 } 2159 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2160 } else if (cblk->server != 0) { 2161 // If the track is stopped before the first frame was mixed, 2162 // do not apply ramp 2163 param = AudioMixer::RAMP_VOLUME; 2164 } 2165 2166 // compute volume for this track 2167 uint32_t vl, vr, va; 2168 if (track->isMuted() || track->isPausing() || 2169 mStreamTypes[track->streamType()].mute) { 2170 vl = vr = va = 0; 2171 if (track->isPausing()) { 2172 track->setPaused(); 2173 } 2174 } else { 2175 2176 // read original volumes with volume control 2177 float typeVolume = mStreamTypes[track->streamType()].volume; 2178 float v = masterVolume * typeVolume; 2179 uint32_t vlr = cblk->getVolumeLR(); 2180 vl = vlr & 0xFFFF; 2181 vr = vlr >> 16; 2182 // track volumes come from shared memory, so can't be trusted and must be clamped 2183 if (vl > MAX_GAIN_INT) { 2184 ALOGV("Track left volume out of range: %04X", vl); 2185 vl = MAX_GAIN_INT; 2186 } 2187 if (vr > MAX_GAIN_INT) { 2188 ALOGV("Track right volume out of range: %04X", vr); 2189 vr = MAX_GAIN_INT; 2190 } 2191 // now apply the master volume and stream type volume 2192 vl = (uint32_t)(v * vl) << 12; 2193 vr = (uint32_t)(v * vr) << 12; 2194 // assuming master volume and stream type volume each go up to 1.0, 2195 // vl and vr are now in 8.24 format 2196 2197 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2198 // send level comes from shared memory and so may be corrupt 2199 if (sendLevel >= MAX_GAIN_INT) { 2200 ALOGV("Track send level out of range: %04X", sendLevel); 2201 sendLevel = MAX_GAIN_INT; 2202 } 2203 va = (uint32_t)(v * sendLevel); 2204 } 2205 // Delegate volume control to effect in track effect chain if needed 2206 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2207 // Do not ramp volume if volume is controlled by effect 2208 param = AudioMixer::VOLUME; 2209 track->mHasVolumeController = true; 2210 } else { 2211 // force no volume ramp when volume controller was just disabled or removed 2212 // from effect chain to avoid volume spike 2213 if (track->mHasVolumeController) { 2214 param = AudioMixer::VOLUME; 2215 } 2216 track->mHasVolumeController = false; 2217 } 2218 2219 // Convert volumes from 8.24 to 4.12 format 2220 int16_t left, right, aux; 2221 // This additional clamping is needed in case chain->setVolume_l() overshot 2222 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2223 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2224 left = int16_t(v_clamped); 2225 v_clamped = (vr + (1 << 11)) >> 12; 2226 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2227 right = int16_t(v_clamped); 2228 2229 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2230 aux = int16_t(va); 2231 2232 // XXX: these things DON'T need to be done each time 2233 mAudioMixer->setBufferProvider(name, track); 2234 mAudioMixer->enable(name); 2235 2236 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2237 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2238 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2239 mAudioMixer->setParameter( 2240 name, 2241 AudioMixer::TRACK, 2242 AudioMixer::FORMAT, (void *)track->format()); 2243 mAudioMixer->setParameter( 2244 name, 2245 AudioMixer::TRACK, 2246 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2247 mAudioMixer->setParameter( 2248 name, 2249 AudioMixer::RESAMPLE, 2250 AudioMixer::SAMPLE_RATE, 2251 (void *)(cblk->sampleRate)); 2252 mAudioMixer->setParameter( 2253 name, 2254 AudioMixer::TRACK, 2255 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2256 mAudioMixer->setParameter( 2257 name, 2258 AudioMixer::TRACK, 2259 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2260 2261 // reset retry count 2262 track->mRetryCount = kMaxTrackRetries; 2263 // If one track is ready, set the mixer ready if: 2264 // - the mixer was not ready during previous round OR 2265 // - no other track is not ready 2266 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2267 mixerStatus != MIXER_TRACKS_ENABLED) { 2268 mixerStatus = MIXER_TRACKS_READY; 2269 } 2270 } else { 2271 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2272 if (track->isStopped()) { 2273 track->reset(); 2274 } 2275 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2276 // We have consumed all the buffers of this track. 2277 // Remove it from the list of active tracks. 2278 tracksToRemove->add(track); 2279 } else { 2280 // No buffers for this track. Give it a few chances to 2281 // fill a buffer, then remove it from active list. 2282 if (--(track->mRetryCount) <= 0) { 2283 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2284 tracksToRemove->add(track); 2285 // indicate to client process that the track was disabled because of underrun 2286 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2287 // If one track is not ready, mark the mixer also not ready if: 2288 // - the mixer was ready during previous round OR 2289 // - no other track is ready 2290 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2291 mixerStatus != MIXER_TRACKS_READY) { 2292 mixerStatus = MIXER_TRACKS_ENABLED; 2293 } 2294 } 2295 mAudioMixer->disable(name); 2296 } 2297 } 2298 2299 // remove all the tracks that need to be... 2300 count = tracksToRemove->size(); 2301 if (CC_UNLIKELY(count)) { 2302 for (size_t i=0 ; i<count ; i++) { 2303 const sp<Track>& track = tracksToRemove->itemAt(i); 2304 mActiveTracks.remove(track); 2305 if (track->mainBuffer() != mMixBuffer) { 2306 chain = getEffectChain_l(track->sessionId()); 2307 if (chain != 0) { 2308 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2309 chain->decActiveTrackCnt(); 2310 } 2311 } 2312 if (track->isTerminated()) { 2313 removeTrack_l(track); 2314 } 2315 } 2316 } 2317 2318 // mix buffer must be cleared if all tracks are connected to an 2319 // effect chain as in this case the mixer will not write to 2320 // mix buffer and track effects will accumulate into it 2321 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2322 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2323 } 2324 2325 mPrevMixerStatus = mixerStatus; 2326 return mixerStatus; 2327} 2328 2329void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2330{ 2331 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2332 this, streamType, mTracks.size()); 2333 Mutex::Autolock _l(mLock); 2334 2335 size_t size = mTracks.size(); 2336 for (size_t i = 0; i < size; i++) { 2337 sp<Track> t = mTracks[i]; 2338 if (t->streamType() == streamType) { 2339 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2340 t->mCblk->cv.signal(); 2341 } 2342 } 2343} 2344 2345void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2346{ 2347 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2348 this, streamType, valid); 2349 Mutex::Autolock _l(mLock); 2350 2351 mStreamTypes[streamType].valid = valid; 2352} 2353 2354// getTrackName_l() must be called with ThreadBase::mLock held 2355int AudioFlinger::MixerThread::getTrackName_l() 2356{ 2357 return mAudioMixer->getTrackName(); 2358} 2359 2360// deleteTrackName_l() must be called with ThreadBase::mLock held 2361void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2362{ 2363 ALOGV("remove track (%d) and delete from mixer", name); 2364 mAudioMixer->deleteTrackName(name); 2365} 2366 2367// checkForNewParameters_l() must be called with ThreadBase::mLock held 2368bool AudioFlinger::MixerThread::checkForNewParameters_l() 2369{ 2370 bool reconfig = false; 2371 2372 while (!mNewParameters.isEmpty()) { 2373 status_t status = NO_ERROR; 2374 String8 keyValuePair = mNewParameters[0]; 2375 AudioParameter param = AudioParameter(keyValuePair); 2376 int value; 2377 2378 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2379 reconfig = true; 2380 } 2381 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2382 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2383 status = BAD_VALUE; 2384 } else { 2385 reconfig = true; 2386 } 2387 } 2388 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2389 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2390 status = BAD_VALUE; 2391 } else { 2392 reconfig = true; 2393 } 2394 } 2395 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2396 // do not accept frame count changes if tracks are open as the track buffer 2397 // size depends on frame count and correct behavior would not be guaranteed 2398 // if frame count is changed after track creation 2399 if (!mTracks.isEmpty()) { 2400 status = INVALID_OPERATION; 2401 } else { 2402 reconfig = true; 2403 } 2404 } 2405 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2406 // when changing the audio output device, call addBatteryData to notify 2407 // the change 2408 if ((int)mDevice != value) { 2409 uint32_t params = 0; 2410 // check whether speaker is on 2411 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2412 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2413 } 2414 2415 int deviceWithoutSpeaker 2416 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2417 // check if any other device (except speaker) is on 2418 if (value & deviceWithoutSpeaker ) { 2419 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2420 } 2421 2422 if (params != 0) { 2423 addBatteryData(params); 2424 } 2425 } 2426 2427 // forward device change to effects that have requested to be 2428 // aware of attached audio device. 2429 mDevice = (uint32_t)value; 2430 for (size_t i = 0; i < mEffectChains.size(); i++) { 2431 mEffectChains[i]->setDevice_l(mDevice); 2432 } 2433 } 2434 2435 if (status == NO_ERROR) { 2436 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2437 keyValuePair.string()); 2438 if (!mStandby && status == INVALID_OPERATION) { 2439 mOutput->stream->common.standby(&mOutput->stream->common); 2440 mStandby = true; 2441 mBytesWritten = 0; 2442 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2443 keyValuePair.string()); 2444 } 2445 if (status == NO_ERROR && reconfig) { 2446 delete mAudioMixer; 2447 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2448 mAudioMixer = NULL; 2449 readOutputParameters(); 2450 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2451 for (size_t i = 0; i < mTracks.size() ; i++) { 2452 int name = getTrackName_l(); 2453 if (name < 0) break; 2454 mTracks[i]->mName = name; 2455 // limit track sample rate to 2 x new output sample rate 2456 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2457 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2458 } 2459 } 2460 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2461 } 2462 } 2463 2464 mNewParameters.removeAt(0); 2465 2466 mParamStatus = status; 2467 mParamCond.signal(); 2468 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2469 // already timed out waiting for the status and will never signal the condition. 2470 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2471 } 2472 return reconfig; 2473} 2474 2475status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2476{ 2477 const size_t SIZE = 256; 2478 char buffer[SIZE]; 2479 String8 result; 2480 2481 PlaybackThread::dumpInternals(fd, args); 2482 2483 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2484 result.append(buffer); 2485 write(fd, result.string(), result.size()); 2486 return NO_ERROR; 2487} 2488 2489uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2490{ 2491 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2492} 2493 2494uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2495{ 2496 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2497} 2498 2499// ---------------------------------------------------------------------------- 2500AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2501 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2502 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2503 // mLeftVolFloat, mRightVolFloat 2504 // mLeftVolShort, mRightVolShort 2505{ 2506} 2507 2508AudioFlinger::DirectOutputThread::~DirectOutputThread() 2509{ 2510} 2511 2512void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2513{ 2514 // Do not apply volume on compressed audio 2515 if (!audio_is_linear_pcm(mFormat)) { 2516 return; 2517 } 2518 2519 // convert to signed 16 bit before volume calculation 2520 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2521 size_t count = mFrameCount * mChannelCount; 2522 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2523 int16_t *dst = mMixBuffer + count-1; 2524 while(count--) { 2525 *dst-- = (int16_t)(*src--^0x80) << 8; 2526 } 2527 } 2528 2529 size_t frameCount = mFrameCount; 2530 int16_t *out = mMixBuffer; 2531 if (ramp) { 2532 if (mChannelCount == 1) { 2533 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2534 int32_t vlInc = d / (int32_t)frameCount; 2535 int32_t vl = ((int32_t)mLeftVolShort << 16); 2536 do { 2537 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2538 out++; 2539 vl += vlInc; 2540 } while (--frameCount); 2541 2542 } else { 2543 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2544 int32_t vlInc = d / (int32_t)frameCount; 2545 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2546 int32_t vrInc = d / (int32_t)frameCount; 2547 int32_t vl = ((int32_t)mLeftVolShort << 16); 2548 int32_t vr = ((int32_t)mRightVolShort << 16); 2549 do { 2550 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2551 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2552 out += 2; 2553 vl += vlInc; 2554 vr += vrInc; 2555 } while (--frameCount); 2556 } 2557 } else { 2558 if (mChannelCount == 1) { 2559 do { 2560 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2561 out++; 2562 } while (--frameCount); 2563 } else { 2564 do { 2565 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2566 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2567 out += 2; 2568 } while (--frameCount); 2569 } 2570 } 2571 2572 // convert back to unsigned 8 bit after volume calculation 2573 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2574 size_t count = mFrameCount * mChannelCount; 2575 int16_t *src = mMixBuffer; 2576 uint8_t *dst = (uint8_t *)mMixBuffer; 2577 while(count--) { 2578 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2579 } 2580 } 2581 2582 mLeftVolShort = leftVol; 2583 mRightVolShort = rightVol; 2584} 2585 2586bool AudioFlinger::DirectOutputThread::threadLoop() 2587{ 2588 mixer_state mixerStatus = MIXER_IDLE; 2589 sp<Track> trackToRemove; 2590 sp<Track> activeTrack; 2591 nsecs_t standbyTime = systemTime(); 2592 int8_t *curBuf; 2593 size_t mixBufferSize = mFrameCount*mFrameSize; 2594 uint32_t activeSleepTime = activeSleepTimeUs(); 2595 uint32_t idleSleepTime = idleSleepTimeUs(); 2596 uint32_t sleepTime = idleSleepTime; 2597 // use shorter standby delay as on normal output to release 2598 // hardware resources as soon as possible 2599 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2600 2601 acquireWakeLock(); 2602 2603 while (!exitPending()) 2604 { 2605 bool rampVolume; 2606 uint16_t leftVol; 2607 uint16_t rightVol; 2608 Vector< sp<EffectChain> > effectChains; 2609 2610 processConfigEvents(); 2611 2612 mixerStatus = MIXER_IDLE; 2613 2614 { // scope for the mLock 2615 2616 Mutex::Autolock _l(mLock); 2617 2618 if (checkForNewParameters_l()) { 2619 mixBufferSize = mFrameCount*mFrameSize; 2620 activeSleepTime = activeSleepTimeUs(); 2621 idleSleepTime = idleSleepTimeUs(); 2622 standbyDelay = microseconds(activeSleepTime*2); 2623 } 2624 2625 // put audio hardware into standby after short delay 2626 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2627 mSuspended)) { 2628 // wait until we have something to do... 2629 if (!mStandby) { 2630 ALOGV("Audio hardware entering standby, mixer %p", this); 2631 mOutput->stream->common.standby(&mOutput->stream->common); 2632 mStandby = true; 2633 mBytesWritten = 0; 2634 } 2635 2636 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2637 // we're about to wait, flush the binder command buffer 2638 IPCThreadState::self()->flushCommands(); 2639 2640 if (exitPending()) break; 2641 2642 releaseWakeLock_l(); 2643 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2644 mWaitWorkCV.wait(mLock); 2645 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2646 acquireWakeLock_l(); 2647 2648 if (!mMasterMute) { 2649 char value[PROPERTY_VALUE_MAX]; 2650 property_get("ro.audio.silent", value, "0"); 2651 if (atoi(value)) { 2652 ALOGD("Silence is golden"); 2653 setMasterMute(true); 2654 } 2655 } 2656 2657 standbyTime = systemTime() + standbyDelay; 2658 sleepTime = idleSleepTime; 2659 continue; 2660 } 2661 } 2662 2663 effectChains = mEffectChains; 2664 2665 // find out which tracks need to be processed 2666 if (mActiveTracks.size() != 0) { 2667 sp<Track> t = mActiveTracks[0].promote(); 2668 if (t == 0) continue; 2669 2670 Track* const track = t.get(); 2671 audio_track_cblk_t* cblk = track->cblk(); 2672 2673 // The first time a track is added we wait 2674 // for all its buffers to be filled before processing it 2675 if (cblk->framesReady() && track->isReady() && 2676 !track->isPaused() && !track->isTerminated()) 2677 { 2678 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2679 2680 if (track->mFillingUpStatus == Track::FS_FILLED) { 2681 track->mFillingUpStatus = Track::FS_ACTIVE; 2682 mLeftVolFloat = mRightVolFloat = 0; 2683 mLeftVolShort = mRightVolShort = 0; 2684 if (track->mState == TrackBase::RESUMING) { 2685 track->mState = TrackBase::ACTIVE; 2686 rampVolume = true; 2687 } 2688 } else if (cblk->server != 0) { 2689 // If the track is stopped before the first frame was mixed, 2690 // do not apply ramp 2691 rampVolume = true; 2692 } 2693 // compute volume for this track 2694 float left, right; 2695 if (track->isMuted() || mMasterMute || track->isPausing() || 2696 mStreamTypes[track->streamType()].mute) { 2697 left = right = 0; 2698 if (track->isPausing()) { 2699 track->setPaused(); 2700 } 2701 } else { 2702 float typeVolume = mStreamTypes[track->streamType()].volume; 2703 float v = mMasterVolume * typeVolume; 2704 uint32_t vlr = cblk->getVolumeLR(); 2705 float v_clamped = v * (vlr & 0xFFFF); 2706 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2707 left = v_clamped/MAX_GAIN; 2708 v_clamped = v * (vlr >> 16); 2709 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2710 right = v_clamped/MAX_GAIN; 2711 } 2712 2713 if (left != mLeftVolFloat || right != mRightVolFloat) { 2714 mLeftVolFloat = left; 2715 mRightVolFloat = right; 2716 2717 // If audio HAL implements volume control, 2718 // force software volume to nominal value 2719 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2720 left = 1.0f; 2721 right = 1.0f; 2722 } 2723 2724 // Convert volumes from float to 8.24 2725 uint32_t vl = (uint32_t)(left * (1 << 24)); 2726 uint32_t vr = (uint32_t)(right * (1 << 24)); 2727 2728 // Delegate volume control to effect in track effect chain if needed 2729 // only one effect chain can be present on DirectOutputThread, so if 2730 // there is one, the track is connected to it 2731 if (!effectChains.isEmpty()) { 2732 // Do not ramp volume if volume is controlled by effect 2733 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2734 rampVolume = false; 2735 } 2736 } 2737 2738 // Convert volumes from 8.24 to 4.12 format 2739 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2740 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2741 leftVol = (uint16_t)v_clamped; 2742 v_clamped = (vr + (1 << 11)) >> 12; 2743 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2744 rightVol = (uint16_t)v_clamped; 2745 } else { 2746 leftVol = mLeftVolShort; 2747 rightVol = mRightVolShort; 2748 rampVolume = false; 2749 } 2750 2751 // reset retry count 2752 track->mRetryCount = kMaxTrackRetriesDirect; 2753 activeTrack = t; 2754 mixerStatus = MIXER_TRACKS_READY; 2755 } else { 2756 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2757 if (track->isStopped()) { 2758 track->reset(); 2759 } 2760 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2761 // We have consumed all the buffers of this track. 2762 // Remove it from the list of active tracks. 2763 trackToRemove = track; 2764 } else { 2765 // No buffers for this track. Give it a few chances to 2766 // fill a buffer, then remove it from active list. 2767 if (--(track->mRetryCount) <= 0) { 2768 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2769 trackToRemove = track; 2770 } else { 2771 mixerStatus = MIXER_TRACKS_ENABLED; 2772 } 2773 } 2774 } 2775 } 2776 2777 // remove all the tracks that need to be... 2778 if (CC_UNLIKELY(trackToRemove != 0)) { 2779 mActiveTracks.remove(trackToRemove); 2780 if (!effectChains.isEmpty()) { 2781 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2782 trackToRemove->sessionId()); 2783 effectChains[0]->decActiveTrackCnt(); 2784 } 2785 if (trackToRemove->isTerminated()) { 2786 removeTrack_l(trackToRemove); 2787 } 2788 } 2789 2790 lockEffectChains_l(effectChains); 2791 } 2792 2793 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2794 AudioBufferProvider::Buffer buffer; 2795 size_t frameCount = mFrameCount; 2796 curBuf = (int8_t *)mMixBuffer; 2797 // output audio to hardware 2798 while (frameCount) { 2799 buffer.frameCount = frameCount; 2800 activeTrack->getNextBuffer(&buffer); 2801 if (CC_UNLIKELY(buffer.raw == NULL)) { 2802 memset(curBuf, 0, frameCount * mFrameSize); 2803 break; 2804 } 2805 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2806 frameCount -= buffer.frameCount; 2807 curBuf += buffer.frameCount * mFrameSize; 2808 activeTrack->releaseBuffer(&buffer); 2809 } 2810 sleepTime = 0; 2811 standbyTime = systemTime() + standbyDelay; 2812 } else { 2813 if (sleepTime == 0) { 2814 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2815 sleepTime = activeSleepTime; 2816 } else { 2817 sleepTime = idleSleepTime; 2818 } 2819 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2820 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2821 sleepTime = 0; 2822 } 2823 } 2824 2825 if (mSuspended) { 2826 sleepTime = suspendSleepTimeUs(); 2827 } 2828 // sleepTime == 0 means we must write to audio hardware 2829 if (sleepTime == 0) { 2830 if (mixerStatus == MIXER_TRACKS_READY) { 2831 applyVolume(leftVol, rightVol, rampVolume); 2832 } 2833 for (size_t i = 0; i < effectChains.size(); i ++) { 2834 effectChains[i]->process_l(); 2835 } 2836 unlockEffectChains(effectChains); 2837 2838 mLastWriteTime = systemTime(); 2839 mInWrite = true; 2840 mBytesWritten += mixBufferSize; 2841 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2842 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2843 mNumWrites++; 2844 mInWrite = false; 2845 mStandby = false; 2846 } else { 2847 unlockEffectChains(effectChains); 2848 usleep(sleepTime); 2849 } 2850 2851 // finally let go of removed track, without the lock held 2852 // since we can't guarantee the destructors won't acquire that 2853 // same lock. 2854 trackToRemove.clear(); 2855 activeTrack.clear(); 2856 2857 // Effect chains will be actually deleted here if they were removed from 2858 // mEffectChains list during mixing or effects processing 2859 effectChains.clear(); 2860 } 2861 2862 if (!mStandby) { 2863 mOutput->stream->common.standby(&mOutput->stream->common); 2864 } 2865 2866 releaseWakeLock(); 2867 2868 ALOGV("DirectOutputThread %p exiting", this); 2869 return false; 2870} 2871 2872// getTrackName_l() must be called with ThreadBase::mLock held 2873int AudioFlinger::DirectOutputThread::getTrackName_l() 2874{ 2875 return 0; 2876} 2877 2878// deleteTrackName_l() must be called with ThreadBase::mLock held 2879void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2880{ 2881} 2882 2883// checkForNewParameters_l() must be called with ThreadBase::mLock held 2884bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2885{ 2886 bool reconfig = false; 2887 2888 while (!mNewParameters.isEmpty()) { 2889 status_t status = NO_ERROR; 2890 String8 keyValuePair = mNewParameters[0]; 2891 AudioParameter param = AudioParameter(keyValuePair); 2892 int value; 2893 2894 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2895 // do not accept frame count changes if tracks are open as the track buffer 2896 // size depends on frame count and correct behavior would not be garantied 2897 // if frame count is changed after track creation 2898 if (!mTracks.isEmpty()) { 2899 status = INVALID_OPERATION; 2900 } else { 2901 reconfig = true; 2902 } 2903 } 2904 if (status == NO_ERROR) { 2905 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2906 keyValuePair.string()); 2907 if (!mStandby && status == INVALID_OPERATION) { 2908 mOutput->stream->common.standby(&mOutput->stream->common); 2909 mStandby = true; 2910 mBytesWritten = 0; 2911 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2912 keyValuePair.string()); 2913 } 2914 if (status == NO_ERROR && reconfig) { 2915 readOutputParameters(); 2916 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2917 } 2918 } 2919 2920 mNewParameters.removeAt(0); 2921 2922 mParamStatus = status; 2923 mParamCond.signal(); 2924 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2925 // already timed out waiting for the status and will never signal the condition. 2926 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2927 } 2928 return reconfig; 2929} 2930 2931uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2932{ 2933 uint32_t time; 2934 if (audio_is_linear_pcm(mFormat)) { 2935 time = PlaybackThread::activeSleepTimeUs(); 2936 } else { 2937 time = 10000; 2938 } 2939 return time; 2940} 2941 2942uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2943{ 2944 uint32_t time; 2945 if (audio_is_linear_pcm(mFormat)) { 2946 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2947 } else { 2948 time = 10000; 2949 } 2950 return time; 2951} 2952 2953uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2954{ 2955 uint32_t time; 2956 if (audio_is_linear_pcm(mFormat)) { 2957 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2958 } else { 2959 time = 10000; 2960 } 2961 return time; 2962} 2963 2964 2965// ---------------------------------------------------------------------------- 2966 2967AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 2968 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 2969 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 2970 mWaitTimeMs(UINT_MAX) 2971{ 2972 addOutputTrack(mainThread); 2973} 2974 2975AudioFlinger::DuplicatingThread::~DuplicatingThread() 2976{ 2977 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2978 mOutputTracks[i]->destroy(); 2979 } 2980} 2981 2982bool AudioFlinger::DuplicatingThread::threadLoop() 2983{ 2984 Vector< sp<Track> > tracksToRemove; 2985 mixer_state mixerStatus = MIXER_IDLE; 2986 nsecs_t standbyTime = systemTime(); 2987 size_t mixBufferSize = mFrameCount*mFrameSize; 2988 SortedVector< sp<OutputTrack> > outputTracks; 2989 uint32_t writeFrames = 0; 2990 uint32_t activeSleepTime = activeSleepTimeUs(); 2991 uint32_t idleSleepTime = idleSleepTimeUs(); 2992 uint32_t sleepTime = idleSleepTime; 2993 Vector< sp<EffectChain> > effectChains; 2994 2995 acquireWakeLock(); 2996 2997 while (!exitPending()) 2998 { 2999 processConfigEvents(); 3000 3001 mixerStatus = MIXER_IDLE; 3002 { // scope for the mLock 3003 3004 Mutex::Autolock _l(mLock); 3005 3006 if (checkForNewParameters_l()) { 3007 mixBufferSize = mFrameCount*mFrameSize; 3008 updateWaitTime(); 3009 activeSleepTime = activeSleepTimeUs(); 3010 idleSleepTime = idleSleepTimeUs(); 3011 } 3012 3013 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3014 3015 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3016 outputTracks.add(mOutputTracks[i]); 3017 } 3018 3019 // put audio hardware into standby after short delay 3020 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3021 mSuspended)) { 3022 if (!mStandby) { 3023 for (size_t i = 0; i < outputTracks.size(); i++) { 3024 outputTracks[i]->stop(); 3025 } 3026 mStandby = true; 3027 mBytesWritten = 0; 3028 } 3029 3030 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3031 // we're about to wait, flush the binder command buffer 3032 IPCThreadState::self()->flushCommands(); 3033 outputTracks.clear(); 3034 3035 if (exitPending()) break; 3036 3037 releaseWakeLock_l(); 3038 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3039 mWaitWorkCV.wait(mLock); 3040 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3041 acquireWakeLock_l(); 3042 3043 mPrevMixerStatus = MIXER_IDLE; 3044 if (!mMasterMute) { 3045 char value[PROPERTY_VALUE_MAX]; 3046 property_get("ro.audio.silent", value, "0"); 3047 if (atoi(value)) { 3048 ALOGD("Silence is golden"); 3049 setMasterMute(true); 3050 } 3051 } 3052 3053 standbyTime = systemTime() + kStandbyTimeInNsecs; 3054 sleepTime = idleSleepTime; 3055 continue; 3056 } 3057 } 3058 3059 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3060 3061 // prevent any changes in effect chain list and in each effect chain 3062 // during mixing and effect process as the audio buffers could be deleted 3063 // or modified if an effect is created or deleted 3064 lockEffectChains_l(effectChains); 3065 } 3066 3067 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3068 // mix buffers... 3069 if (outputsReady(outputTracks)) { 3070 mAudioMixer->process(); 3071 } else { 3072 memset(mMixBuffer, 0, mixBufferSize); 3073 } 3074 sleepTime = 0; 3075 writeFrames = mFrameCount; 3076 } else { 3077 if (sleepTime == 0) { 3078 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3079 sleepTime = activeSleepTime; 3080 } else { 3081 sleepTime = idleSleepTime; 3082 } 3083 } else if (mBytesWritten != 0) { 3084 // flush remaining overflow buffers in output tracks 3085 for (size_t i = 0; i < outputTracks.size(); i++) { 3086 if (outputTracks[i]->isActive()) { 3087 sleepTime = 0; 3088 writeFrames = 0; 3089 memset(mMixBuffer, 0, mixBufferSize); 3090 break; 3091 } 3092 } 3093 } 3094 } 3095 3096 if (mSuspended) { 3097 sleepTime = suspendSleepTimeUs(); 3098 } 3099 // sleepTime == 0 means we must write to audio hardware 3100 if (sleepTime == 0) { 3101 for (size_t i = 0; i < effectChains.size(); i ++) { 3102 effectChains[i]->process_l(); 3103 } 3104 // enable changes in effect chain 3105 unlockEffectChains(effectChains); 3106 3107 standbyTime = systemTime() + kStandbyTimeInNsecs; 3108 for (size_t i = 0; i < outputTracks.size(); i++) { 3109 outputTracks[i]->write(mMixBuffer, writeFrames); 3110 } 3111 mStandby = false; 3112 mBytesWritten += mixBufferSize; 3113 } else { 3114 // enable changes in effect chain 3115 unlockEffectChains(effectChains); 3116 usleep(sleepTime); 3117 } 3118 3119 // finally let go of all our tracks, without the lock held 3120 // since we can't guarantee the destructors won't acquire that 3121 // same lock. 3122 tracksToRemove.clear(); 3123 outputTracks.clear(); 3124 3125 // Effect chains will be actually deleted here if they were removed from 3126 // mEffectChains list during mixing or effects processing 3127 effectChains.clear(); 3128 } 3129 3130 releaseWakeLock(); 3131 3132 return false; 3133} 3134 3135void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3136{ 3137 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3138 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3139 this, 3140 mSampleRate, 3141 mFormat, 3142 mChannelMask, 3143 frameCount); 3144 if (outputTrack->cblk() != NULL) { 3145 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3146 mOutputTracks.add(outputTrack); 3147 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3148 updateWaitTime(); 3149 } 3150} 3151 3152void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3153{ 3154 Mutex::Autolock _l(mLock); 3155 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3156 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3157 mOutputTracks[i]->destroy(); 3158 mOutputTracks.removeAt(i); 3159 updateWaitTime(); 3160 return; 3161 } 3162 } 3163 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3164} 3165 3166void AudioFlinger::DuplicatingThread::updateWaitTime() 3167{ 3168 mWaitTimeMs = UINT_MAX; 3169 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3170 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3171 if (strong != 0) { 3172 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3173 if (waitTimeMs < mWaitTimeMs) { 3174 mWaitTimeMs = waitTimeMs; 3175 } 3176 } 3177 } 3178} 3179 3180 3181bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3182{ 3183 for (size_t i = 0; i < outputTracks.size(); i++) { 3184 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3185 if (thread == 0) { 3186 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3187 return false; 3188 } 3189 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3190 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3191 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3192 return false; 3193 } 3194 } 3195 return true; 3196} 3197 3198uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3199{ 3200 return (mWaitTimeMs * 1000) / 2; 3201} 3202 3203// ---------------------------------------------------------------------------- 3204 3205// TrackBase constructor must be called with AudioFlinger::mLock held 3206AudioFlinger::ThreadBase::TrackBase::TrackBase( 3207 const wp<ThreadBase>& thread, 3208 const sp<Client>& client, 3209 uint32_t sampleRate, 3210 audio_format_t format, 3211 uint32_t channelMask, 3212 int frameCount, 3213 uint32_t flags, 3214 const sp<IMemory>& sharedBuffer, 3215 int sessionId) 3216 : RefBase(), 3217 mThread(thread), 3218 mClient(client), 3219 mCblk(NULL), 3220 // mBuffer 3221 // mBufferEnd 3222 mFrameCount(0), 3223 mState(IDLE), 3224 mFormat(format), 3225 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3226 mSessionId(sessionId) 3227 // mChannelCount 3228 // mChannelMask 3229{ 3230 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3231 3232 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3233 size_t size = sizeof(audio_track_cblk_t); 3234 uint8_t channelCount = popcount(channelMask); 3235 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3236 if (sharedBuffer == 0) { 3237 size += bufferSize; 3238 } 3239 3240 if (client != NULL) { 3241 mCblkMemory = client->heap()->allocate(size); 3242 if (mCblkMemory != 0) { 3243 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3244 if (mCblk != NULL) { // construct the shared structure in-place. 3245 new(mCblk) audio_track_cblk_t(); 3246 // clear all buffers 3247 mCblk->frameCount = frameCount; 3248 mCblk->sampleRate = sampleRate; 3249 mChannelCount = channelCount; 3250 mChannelMask = channelMask; 3251 if (sharedBuffer == 0) { 3252 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3253 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3254 // Force underrun condition to avoid false underrun callback until first data is 3255 // written to buffer (other flags are cleared) 3256 mCblk->flags = CBLK_UNDERRUN_ON; 3257 } else { 3258 mBuffer = sharedBuffer->pointer(); 3259 } 3260 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3261 } 3262 } else { 3263 ALOGE("not enough memory for AudioTrack size=%u", size); 3264 client->heap()->dump("AudioTrack"); 3265 return; 3266 } 3267 } else { 3268 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3269 // construct the shared structure in-place. 3270 new(mCblk) audio_track_cblk_t(); 3271 // clear all buffers 3272 mCblk->frameCount = frameCount; 3273 mCblk->sampleRate = sampleRate; 3274 mChannelCount = channelCount; 3275 mChannelMask = channelMask; 3276 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3277 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3278 // Force underrun condition to avoid false underrun callback until first data is 3279 // written to buffer (other flags are cleared) 3280 mCblk->flags = CBLK_UNDERRUN_ON; 3281 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3282 } 3283} 3284 3285AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3286{ 3287 if (mCblk != NULL) { 3288 if (mClient == 0) { 3289 delete mCblk; 3290 } else { 3291 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3292 } 3293 } 3294 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3295 if (mClient != 0) { 3296 // Client destructor must run with AudioFlinger mutex locked 3297 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3298 // If the client's reference count drops to zero, the associated destructor 3299 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3300 // relying on the automatic clear() at end of scope. 3301 mClient.clear(); 3302 } 3303} 3304 3305void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3306{ 3307 buffer->raw = NULL; 3308 mFrameCount = buffer->frameCount; 3309 step(); 3310 buffer->frameCount = 0; 3311} 3312 3313bool AudioFlinger::ThreadBase::TrackBase::step() { 3314 bool result; 3315 audio_track_cblk_t* cblk = this->cblk(); 3316 3317 result = cblk->stepServer(mFrameCount); 3318 if (!result) { 3319 ALOGV("stepServer failed acquiring cblk mutex"); 3320 mFlags |= STEPSERVER_FAILED; 3321 } 3322 return result; 3323} 3324 3325void AudioFlinger::ThreadBase::TrackBase::reset() { 3326 audio_track_cblk_t* cblk = this->cblk(); 3327 3328 cblk->user = 0; 3329 cblk->server = 0; 3330 cblk->userBase = 0; 3331 cblk->serverBase = 0; 3332 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3333 ALOGV("TrackBase::reset"); 3334} 3335 3336int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3337 return (int)mCblk->sampleRate; 3338} 3339 3340void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3341 audio_track_cblk_t* cblk = this->cblk(); 3342 size_t frameSize = cblk->frameSize; 3343 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3344 int8_t *bufferEnd = bufferStart + frames * frameSize; 3345 3346 // Check validity of returned pointer in case the track control block would have been corrupted. 3347 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3348 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3349 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3350 server %d, serverBase %d, user %d, userBase %d", 3351 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3352 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3353 return NULL; 3354 } 3355 3356 return bufferStart; 3357} 3358 3359// ---------------------------------------------------------------------------- 3360 3361// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3362AudioFlinger::PlaybackThread::Track::Track( 3363 const wp<ThreadBase>& thread, 3364 const sp<Client>& client, 3365 audio_stream_type_t streamType, 3366 uint32_t sampleRate, 3367 audio_format_t format, 3368 uint32_t channelMask, 3369 int frameCount, 3370 const sp<IMemory>& sharedBuffer, 3371 int sessionId) 3372 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3373 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3374 mAuxEffectId(0), mHasVolumeController(false) 3375{ 3376 if (mCblk != NULL) { 3377 sp<ThreadBase> baseThread = thread.promote(); 3378 if (baseThread != 0) { 3379 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3380 mName = playbackThread->getTrackName_l(); 3381 mMainBuffer = playbackThread->mixBuffer(); 3382 } 3383 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3384 if (mName < 0) { 3385 ALOGE("no more track names available"); 3386 } 3387 mStreamType = streamType; 3388 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3389 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3390 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3391 } 3392} 3393 3394AudioFlinger::PlaybackThread::Track::~Track() 3395{ 3396 ALOGV("PlaybackThread::Track destructor"); 3397 sp<ThreadBase> thread = mThread.promote(); 3398 if (thread != 0) { 3399 Mutex::Autolock _l(thread->mLock); 3400 mState = TERMINATED; 3401 } 3402} 3403 3404void AudioFlinger::PlaybackThread::Track::destroy() 3405{ 3406 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3407 // by removing it from mTracks vector, so there is a risk that this Tracks's 3408 // desctructor is called. As the destructor needs to lock mLock, 3409 // we must acquire a strong reference on this Track before locking mLock 3410 // here so that the destructor is called only when exiting this function. 3411 // On the other hand, as long as Track::destroy() is only called by 3412 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3413 // this Track with its member mTrack. 3414 sp<Track> keep(this); 3415 { // scope for mLock 3416 sp<ThreadBase> thread = mThread.promote(); 3417 if (thread != 0) { 3418 if (!isOutputTrack()) { 3419 if (mState == ACTIVE || mState == RESUMING) { 3420 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3421 3422 // to track the speaker usage 3423 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3424 } 3425 AudioSystem::releaseOutput(thread->id()); 3426 } 3427 Mutex::Autolock _l(thread->mLock); 3428 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3429 playbackThread->destroyTrack_l(this); 3430 } 3431 } 3432} 3433 3434void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3435{ 3436 uint32_t vlr = mCblk->getVolumeLR(); 3437 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", 3438 mName - AudioMixer::TRACK0, 3439 (mClient == 0) ? getpid() : mClient->pid(), 3440 mStreamType, 3441 mFormat, 3442 mChannelMask, 3443 mSessionId, 3444 mFrameCount, 3445 mState, 3446 mMute, 3447 mFillingUpStatus, 3448 mCblk->sampleRate, 3449 vlr & 0xFFFF, 3450 vlr >> 16, 3451 mCblk->server, 3452 mCblk->user, 3453 (int)mMainBuffer, 3454 (int)mAuxBuffer); 3455} 3456 3457status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3458{ 3459 audio_track_cblk_t* cblk = this->cblk(); 3460 uint32_t framesReady; 3461 uint32_t framesReq = buffer->frameCount; 3462 3463 // Check if last stepServer failed, try to step now 3464 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3465 if (!step()) goto getNextBuffer_exit; 3466 ALOGV("stepServer recovered"); 3467 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3468 } 3469 3470 framesReady = cblk->framesReady(); 3471 3472 if (CC_LIKELY(framesReady)) { 3473 uint32_t s = cblk->server; 3474 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3475 3476 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3477 if (framesReq > framesReady) { 3478 framesReq = framesReady; 3479 } 3480 if (s + framesReq > bufferEnd) { 3481 framesReq = bufferEnd - s; 3482 } 3483 3484 buffer->raw = getBuffer(s, framesReq); 3485 if (buffer->raw == NULL) goto getNextBuffer_exit; 3486 3487 buffer->frameCount = framesReq; 3488 return NO_ERROR; 3489 } 3490 3491getNextBuffer_exit: 3492 buffer->raw = NULL; 3493 buffer->frameCount = 0; 3494 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3495 return NOT_ENOUGH_DATA; 3496} 3497 3498bool AudioFlinger::PlaybackThread::Track::isReady() const { 3499 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3500 3501 if (mCblk->framesReady() >= mCblk->frameCount || 3502 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3503 mFillingUpStatus = FS_FILLED; 3504 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3505 return true; 3506 } 3507 return false; 3508} 3509 3510status_t AudioFlinger::PlaybackThread::Track::start() 3511{ 3512 status_t status = NO_ERROR; 3513 ALOGV("start(%d), calling pid %d session %d", 3514 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3515 sp<ThreadBase> thread = mThread.promote(); 3516 if (thread != 0) { 3517 Mutex::Autolock _l(thread->mLock); 3518 track_state state = mState; 3519 // here the track could be either new, or restarted 3520 // in both cases "unstop" the track 3521 if (mState == PAUSED) { 3522 mState = TrackBase::RESUMING; 3523 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3524 } else { 3525 mState = TrackBase::ACTIVE; 3526 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3527 } 3528 3529 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3530 thread->mLock.unlock(); 3531 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3532 thread->mLock.lock(); 3533 3534 // to track the speaker usage 3535 if (status == NO_ERROR) { 3536 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3537 } 3538 } 3539 if (status == NO_ERROR) { 3540 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3541 playbackThread->addTrack_l(this); 3542 } else { 3543 mState = state; 3544 } 3545 } else { 3546 status = BAD_VALUE; 3547 } 3548 return status; 3549} 3550 3551void AudioFlinger::PlaybackThread::Track::stop() 3552{ 3553 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3554 sp<ThreadBase> thread = mThread.promote(); 3555 if (thread != 0) { 3556 Mutex::Autolock _l(thread->mLock); 3557 track_state state = mState; 3558 if (mState > STOPPED) { 3559 mState = STOPPED; 3560 // If the track is not active (PAUSED and buffers full), flush buffers 3561 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3562 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3563 reset(); 3564 } 3565 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3566 } 3567 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3568 thread->mLock.unlock(); 3569 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3570 thread->mLock.lock(); 3571 3572 // to track the speaker usage 3573 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3574 } 3575 } 3576} 3577 3578void AudioFlinger::PlaybackThread::Track::pause() 3579{ 3580 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3581 sp<ThreadBase> thread = mThread.promote(); 3582 if (thread != 0) { 3583 Mutex::Autolock _l(thread->mLock); 3584 if (mState == ACTIVE || mState == RESUMING) { 3585 mState = PAUSING; 3586 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3587 if (!isOutputTrack()) { 3588 thread->mLock.unlock(); 3589 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3590 thread->mLock.lock(); 3591 3592 // to track the speaker usage 3593 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3594 } 3595 } 3596 } 3597} 3598 3599void AudioFlinger::PlaybackThread::Track::flush() 3600{ 3601 ALOGV("flush(%d)", mName); 3602 sp<ThreadBase> thread = mThread.promote(); 3603 if (thread != 0) { 3604 Mutex::Autolock _l(thread->mLock); 3605 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3606 return; 3607 } 3608 // No point remaining in PAUSED state after a flush => go to 3609 // STOPPED state 3610 mState = STOPPED; 3611 3612 // do not reset the track if it is still in the process of being stopped or paused. 3613 // this will be done by prepareTracks_l() when the track is stopped. 3614 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3615 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3616 reset(); 3617 } 3618 } 3619} 3620 3621void AudioFlinger::PlaybackThread::Track::reset() 3622{ 3623 // Do not reset twice to avoid discarding data written just after a flush and before 3624 // the audioflinger thread detects the track is stopped. 3625 if (!mResetDone) { 3626 TrackBase::reset(); 3627 // Force underrun condition to avoid false underrun callback until first data is 3628 // written to buffer 3629 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3630 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3631 mFillingUpStatus = FS_FILLING; 3632 mResetDone = true; 3633 } 3634} 3635 3636void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3637{ 3638 mMute = muted; 3639} 3640 3641status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3642{ 3643 status_t status = DEAD_OBJECT; 3644 sp<ThreadBase> thread = mThread.promote(); 3645 if (thread != 0) { 3646 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3647 status = playbackThread->attachAuxEffect(this, EffectId); 3648 } 3649 return status; 3650} 3651 3652void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3653{ 3654 mAuxEffectId = EffectId; 3655 mAuxBuffer = buffer; 3656} 3657 3658// ---------------------------------------------------------------------------- 3659 3660// RecordTrack constructor must be called with AudioFlinger::mLock held 3661AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3662 const wp<ThreadBase>& thread, 3663 const sp<Client>& client, 3664 uint32_t sampleRate, 3665 audio_format_t format, 3666 uint32_t channelMask, 3667 int frameCount, 3668 uint32_t flags, 3669 int sessionId) 3670 : TrackBase(thread, client, sampleRate, format, 3671 channelMask, frameCount, flags, 0, sessionId), 3672 mOverflow(false) 3673{ 3674 if (mCblk != NULL) { 3675 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3676 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3677 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3678 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3679 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3680 } else { 3681 mCblk->frameSize = sizeof(int8_t); 3682 } 3683 } 3684} 3685 3686AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3687{ 3688 sp<ThreadBase> thread = mThread.promote(); 3689 if (thread != 0) { 3690 AudioSystem::releaseInput(thread->id()); 3691 } 3692} 3693 3694status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3695{ 3696 audio_track_cblk_t* cblk = this->cblk(); 3697 uint32_t framesAvail; 3698 uint32_t framesReq = buffer->frameCount; 3699 3700 // Check if last stepServer failed, try to step now 3701 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3702 if (!step()) goto getNextBuffer_exit; 3703 ALOGV("stepServer recovered"); 3704 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3705 } 3706 3707 framesAvail = cblk->framesAvailable_l(); 3708 3709 if (CC_LIKELY(framesAvail)) { 3710 uint32_t s = cblk->server; 3711 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3712 3713 if (framesReq > framesAvail) { 3714 framesReq = framesAvail; 3715 } 3716 if (s + framesReq > bufferEnd) { 3717 framesReq = bufferEnd - s; 3718 } 3719 3720 buffer->raw = getBuffer(s, framesReq); 3721 if (buffer->raw == NULL) goto getNextBuffer_exit; 3722 3723 buffer->frameCount = framesReq; 3724 return NO_ERROR; 3725 } 3726 3727getNextBuffer_exit: 3728 buffer->raw = NULL; 3729 buffer->frameCount = 0; 3730 return NOT_ENOUGH_DATA; 3731} 3732 3733status_t AudioFlinger::RecordThread::RecordTrack::start() 3734{ 3735 sp<ThreadBase> thread = mThread.promote(); 3736 if (thread != 0) { 3737 RecordThread *recordThread = (RecordThread *)thread.get(); 3738 return recordThread->start(this); 3739 } else { 3740 return BAD_VALUE; 3741 } 3742} 3743 3744void AudioFlinger::RecordThread::RecordTrack::stop() 3745{ 3746 sp<ThreadBase> thread = mThread.promote(); 3747 if (thread != 0) { 3748 RecordThread *recordThread = (RecordThread *)thread.get(); 3749 recordThread->stop(this); 3750 TrackBase::reset(); 3751 // Force overerrun condition to avoid false overrun callback until first data is 3752 // read from buffer 3753 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3754 } 3755} 3756 3757void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3758{ 3759 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3760 (mClient == 0) ? getpid() : mClient->pid(), 3761 mFormat, 3762 mChannelMask, 3763 mSessionId, 3764 mFrameCount, 3765 mState, 3766 mCblk->sampleRate, 3767 mCblk->server, 3768 mCblk->user); 3769} 3770 3771 3772// ---------------------------------------------------------------------------- 3773 3774AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3775 const wp<ThreadBase>& thread, 3776 DuplicatingThread *sourceThread, 3777 uint32_t sampleRate, 3778 audio_format_t format, 3779 uint32_t channelMask, 3780 int frameCount) 3781 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3782 mActive(false), mSourceThread(sourceThread) 3783{ 3784 3785 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3786 if (mCblk != NULL) { 3787 mCblk->flags |= CBLK_DIRECTION_OUT; 3788 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3789 mOutBuffer.frameCount = 0; 3790 playbackThread->mTracks.add(this); 3791 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3792 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3793 mCblk, mBuffer, mCblk->buffers, 3794 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3795 } else { 3796 ALOGW("Error creating output track on thread %p", playbackThread); 3797 } 3798} 3799 3800AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3801{ 3802 clearBufferQueue(); 3803} 3804 3805status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3806{ 3807 status_t status = Track::start(); 3808 if (status != NO_ERROR) { 3809 return status; 3810 } 3811 3812 mActive = true; 3813 mRetryCount = 127; 3814 return status; 3815} 3816 3817void AudioFlinger::PlaybackThread::OutputTrack::stop() 3818{ 3819 Track::stop(); 3820 clearBufferQueue(); 3821 mOutBuffer.frameCount = 0; 3822 mActive = false; 3823} 3824 3825bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3826{ 3827 Buffer *pInBuffer; 3828 Buffer inBuffer; 3829 uint32_t channelCount = mChannelCount; 3830 bool outputBufferFull = false; 3831 inBuffer.frameCount = frames; 3832 inBuffer.i16 = data; 3833 3834 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3835 3836 if (!mActive && frames != 0) { 3837 start(); 3838 sp<ThreadBase> thread = mThread.promote(); 3839 if (thread != 0) { 3840 MixerThread *mixerThread = (MixerThread *)thread.get(); 3841 if (mCblk->frameCount > frames){ 3842 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3843 uint32_t startFrames = (mCblk->frameCount - frames); 3844 pInBuffer = new Buffer; 3845 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3846 pInBuffer->frameCount = startFrames; 3847 pInBuffer->i16 = pInBuffer->mBuffer; 3848 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3849 mBufferQueue.add(pInBuffer); 3850 } else { 3851 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3852 } 3853 } 3854 } 3855 } 3856 3857 while (waitTimeLeftMs) { 3858 // First write pending buffers, then new data 3859 if (mBufferQueue.size()) { 3860 pInBuffer = mBufferQueue.itemAt(0); 3861 } else { 3862 pInBuffer = &inBuffer; 3863 } 3864 3865 if (pInBuffer->frameCount == 0) { 3866 break; 3867 } 3868 3869 if (mOutBuffer.frameCount == 0) { 3870 mOutBuffer.frameCount = pInBuffer->frameCount; 3871 nsecs_t startTime = systemTime(); 3872 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3873 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3874 outputBufferFull = true; 3875 break; 3876 } 3877 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3878 if (waitTimeLeftMs >= waitTimeMs) { 3879 waitTimeLeftMs -= waitTimeMs; 3880 } else { 3881 waitTimeLeftMs = 0; 3882 } 3883 } 3884 3885 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3886 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3887 mCblk->stepUser(outFrames); 3888 pInBuffer->frameCount -= outFrames; 3889 pInBuffer->i16 += outFrames * channelCount; 3890 mOutBuffer.frameCount -= outFrames; 3891 mOutBuffer.i16 += outFrames * channelCount; 3892 3893 if (pInBuffer->frameCount == 0) { 3894 if (mBufferQueue.size()) { 3895 mBufferQueue.removeAt(0); 3896 delete [] pInBuffer->mBuffer; 3897 delete pInBuffer; 3898 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3899 } else { 3900 break; 3901 } 3902 } 3903 } 3904 3905 // If we could not write all frames, allocate a buffer and queue it for next time. 3906 if (inBuffer.frameCount) { 3907 sp<ThreadBase> thread = mThread.promote(); 3908 if (thread != 0 && !thread->standby()) { 3909 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3910 pInBuffer = new Buffer; 3911 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3912 pInBuffer->frameCount = inBuffer.frameCount; 3913 pInBuffer->i16 = pInBuffer->mBuffer; 3914 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3915 mBufferQueue.add(pInBuffer); 3916 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3917 } else { 3918 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3919 } 3920 } 3921 } 3922 3923 // Calling write() with a 0 length buffer, means that no more data will be written: 3924 // If no more buffers are pending, fill output track buffer to make sure it is started 3925 // by output mixer. 3926 if (frames == 0 && mBufferQueue.size() == 0) { 3927 if (mCblk->user < mCblk->frameCount) { 3928 frames = mCblk->frameCount - mCblk->user; 3929 pInBuffer = new Buffer; 3930 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3931 pInBuffer->frameCount = frames; 3932 pInBuffer->i16 = pInBuffer->mBuffer; 3933 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3934 mBufferQueue.add(pInBuffer); 3935 } else if (mActive) { 3936 stop(); 3937 } 3938 } 3939 3940 return outputBufferFull; 3941} 3942 3943status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3944{ 3945 int active; 3946 status_t result; 3947 audio_track_cblk_t* cblk = mCblk; 3948 uint32_t framesReq = buffer->frameCount; 3949 3950// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3951 buffer->frameCount = 0; 3952 3953 uint32_t framesAvail = cblk->framesAvailable(); 3954 3955 3956 if (framesAvail == 0) { 3957 Mutex::Autolock _l(cblk->lock); 3958 goto start_loop_here; 3959 while (framesAvail == 0) { 3960 active = mActive; 3961 if (CC_UNLIKELY(!active)) { 3962 ALOGV("Not active and NO_MORE_BUFFERS"); 3963 return NO_MORE_BUFFERS; 3964 } 3965 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3966 if (result != NO_ERROR) { 3967 return NO_MORE_BUFFERS; 3968 } 3969 // read the server count again 3970 start_loop_here: 3971 framesAvail = cblk->framesAvailable_l(); 3972 } 3973 } 3974 3975// if (framesAvail < framesReq) { 3976// return NO_MORE_BUFFERS; 3977// } 3978 3979 if (framesReq > framesAvail) { 3980 framesReq = framesAvail; 3981 } 3982 3983 uint32_t u = cblk->user; 3984 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3985 3986 if (u + framesReq > bufferEnd) { 3987 framesReq = bufferEnd - u; 3988 } 3989 3990 buffer->frameCount = framesReq; 3991 buffer->raw = (void *)cblk->buffer(u); 3992 return NO_ERROR; 3993} 3994 3995 3996void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3997{ 3998 size_t size = mBufferQueue.size(); 3999 Buffer *pBuffer; 4000 4001 for (size_t i = 0; i < size; i++) { 4002 pBuffer = mBufferQueue.itemAt(i); 4003 delete [] pBuffer->mBuffer; 4004 delete pBuffer; 4005 } 4006 mBufferQueue.clear(); 4007} 4008 4009// ---------------------------------------------------------------------------- 4010 4011AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4012 : RefBase(), 4013 mAudioFlinger(audioFlinger), 4014 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4015 mPid(pid) 4016{ 4017 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4018} 4019 4020// Client destructor must be called with AudioFlinger::mLock held 4021AudioFlinger::Client::~Client() 4022{ 4023 mAudioFlinger->removeClient_l(mPid); 4024} 4025 4026sp<MemoryDealer> AudioFlinger::Client::heap() const 4027{ 4028 return mMemoryDealer; 4029} 4030 4031// ---------------------------------------------------------------------------- 4032 4033AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4034 const sp<IAudioFlingerClient>& client, 4035 pid_t pid) 4036 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4037{ 4038} 4039 4040AudioFlinger::NotificationClient::~NotificationClient() 4041{ 4042} 4043 4044void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4045{ 4046 sp<NotificationClient> keep(this); 4047 { 4048 mAudioFlinger->removeNotificationClient(mPid); 4049 } 4050} 4051 4052// ---------------------------------------------------------------------------- 4053 4054AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4055 : BnAudioTrack(), 4056 mTrack(track) 4057{ 4058} 4059 4060AudioFlinger::TrackHandle::~TrackHandle() { 4061 // just stop the track on deletion, associated resources 4062 // will be freed from the main thread once all pending buffers have 4063 // been played. Unless it's not in the active track list, in which 4064 // case we free everything now... 4065 mTrack->destroy(); 4066} 4067 4068sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4069 return mTrack->getCblk(); 4070} 4071 4072status_t AudioFlinger::TrackHandle::start() { 4073 return mTrack->start(); 4074} 4075 4076void AudioFlinger::TrackHandle::stop() { 4077 mTrack->stop(); 4078} 4079 4080void AudioFlinger::TrackHandle::flush() { 4081 mTrack->flush(); 4082} 4083 4084void AudioFlinger::TrackHandle::mute(bool e) { 4085 mTrack->mute(e); 4086} 4087 4088void AudioFlinger::TrackHandle::pause() { 4089 mTrack->pause(); 4090} 4091 4092status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4093{ 4094 return mTrack->attachAuxEffect(EffectId); 4095} 4096 4097status_t AudioFlinger::TrackHandle::onTransact( 4098 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4099{ 4100 return BnAudioTrack::onTransact(code, data, reply, flags); 4101} 4102 4103// ---------------------------------------------------------------------------- 4104 4105sp<IAudioRecord> AudioFlinger::openRecord( 4106 pid_t pid, 4107 audio_io_handle_t input, 4108 uint32_t sampleRate, 4109 audio_format_t format, 4110 uint32_t channelMask, 4111 int frameCount, 4112 uint32_t flags, 4113 int *sessionId, 4114 status_t *status) 4115{ 4116 sp<RecordThread::RecordTrack> recordTrack; 4117 sp<RecordHandle> recordHandle; 4118 sp<Client> client; 4119 status_t lStatus; 4120 RecordThread *thread; 4121 size_t inFrameCount; 4122 int lSessionId; 4123 4124 // check calling permissions 4125 if (!recordingAllowed()) { 4126 lStatus = PERMISSION_DENIED; 4127 goto Exit; 4128 } 4129 4130 // add client to list 4131 { // scope for mLock 4132 Mutex::Autolock _l(mLock); 4133 thread = checkRecordThread_l(input); 4134 if (thread == NULL) { 4135 lStatus = BAD_VALUE; 4136 goto Exit; 4137 } 4138 4139 client = registerPid_l(pid); 4140 4141 // If no audio session id is provided, create one here 4142 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4143 lSessionId = *sessionId; 4144 } else { 4145 lSessionId = nextUniqueId(); 4146 if (sessionId != NULL) { 4147 *sessionId = lSessionId; 4148 } 4149 } 4150 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4151 recordTrack = thread->createRecordTrack_l(client, 4152 sampleRate, 4153 format, 4154 channelMask, 4155 frameCount, 4156 flags, 4157 lSessionId, 4158 &lStatus); 4159 } 4160 if (lStatus != NO_ERROR) { 4161 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4162 // destructor is called by the TrackBase destructor with mLock held 4163 client.clear(); 4164 recordTrack.clear(); 4165 goto Exit; 4166 } 4167 4168 // return to handle to client 4169 recordHandle = new RecordHandle(recordTrack); 4170 lStatus = NO_ERROR; 4171 4172Exit: 4173 if (status) { 4174 *status = lStatus; 4175 } 4176 return recordHandle; 4177} 4178 4179// ---------------------------------------------------------------------------- 4180 4181AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4182 : BnAudioRecord(), 4183 mRecordTrack(recordTrack) 4184{ 4185} 4186 4187AudioFlinger::RecordHandle::~RecordHandle() { 4188 stop(); 4189} 4190 4191sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4192 return mRecordTrack->getCblk(); 4193} 4194 4195status_t AudioFlinger::RecordHandle::start() { 4196 ALOGV("RecordHandle::start()"); 4197 return mRecordTrack->start(); 4198} 4199 4200void AudioFlinger::RecordHandle::stop() { 4201 ALOGV("RecordHandle::stop()"); 4202 mRecordTrack->stop(); 4203} 4204 4205status_t AudioFlinger::RecordHandle::onTransact( 4206 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4207{ 4208 return BnAudioRecord::onTransact(code, data, reply, flags); 4209} 4210 4211// ---------------------------------------------------------------------------- 4212 4213AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4214 AudioStreamIn *input, 4215 uint32_t sampleRate, 4216 uint32_t channels, 4217 audio_io_handle_t id, 4218 uint32_t device) : 4219 ThreadBase(audioFlinger, id, device, RECORD), 4220 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4221 // mRsmpInIndex and mInputBytes set by readInputParameters() 4222 mReqChannelCount(popcount(channels)), 4223 mReqSampleRate(sampleRate) 4224 // mBytesRead is only meaningful while active, and so is cleared in start() 4225 // (but might be better to also clear here for dump?) 4226{ 4227 snprintf(mName, kNameLength, "AudioIn_%d", id); 4228 4229 readInputParameters(); 4230} 4231 4232 4233AudioFlinger::RecordThread::~RecordThread() 4234{ 4235 delete[] mRsmpInBuffer; 4236 delete mResampler; 4237 delete[] mRsmpOutBuffer; 4238} 4239 4240void AudioFlinger::RecordThread::onFirstRef() 4241{ 4242 run(mName, PRIORITY_URGENT_AUDIO); 4243} 4244 4245status_t AudioFlinger::RecordThread::readyToRun() 4246{ 4247 status_t status = initCheck(); 4248 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4249 return status; 4250} 4251 4252bool AudioFlinger::RecordThread::threadLoop() 4253{ 4254 AudioBufferProvider::Buffer buffer; 4255 sp<RecordTrack> activeTrack; 4256 Vector< sp<EffectChain> > effectChains; 4257 4258 nsecs_t lastWarning = 0; 4259 4260 acquireWakeLock(); 4261 4262 // start recording 4263 while (!exitPending()) { 4264 4265 processConfigEvents(); 4266 4267 { // scope for mLock 4268 Mutex::Autolock _l(mLock); 4269 checkForNewParameters_l(); 4270 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4271 if (!mStandby) { 4272 mInput->stream->common.standby(&mInput->stream->common); 4273 mStandby = true; 4274 } 4275 4276 if (exitPending()) break; 4277 4278 releaseWakeLock_l(); 4279 ALOGV("RecordThread: loop stopping"); 4280 // go to sleep 4281 mWaitWorkCV.wait(mLock); 4282 ALOGV("RecordThread: loop starting"); 4283 acquireWakeLock_l(); 4284 continue; 4285 } 4286 if (mActiveTrack != 0) { 4287 if (mActiveTrack->mState == TrackBase::PAUSING) { 4288 if (!mStandby) { 4289 mInput->stream->common.standby(&mInput->stream->common); 4290 mStandby = true; 4291 } 4292 mActiveTrack.clear(); 4293 mStartStopCond.broadcast(); 4294 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4295 if (mReqChannelCount != mActiveTrack->channelCount()) { 4296 mActiveTrack.clear(); 4297 mStartStopCond.broadcast(); 4298 } else if (mBytesRead != 0) { 4299 // record start succeeds only if first read from audio input 4300 // succeeds 4301 if (mBytesRead > 0) { 4302 mActiveTrack->mState = TrackBase::ACTIVE; 4303 } else { 4304 mActiveTrack.clear(); 4305 } 4306 mStartStopCond.broadcast(); 4307 } 4308 mStandby = false; 4309 } 4310 } 4311 lockEffectChains_l(effectChains); 4312 } 4313 4314 if (mActiveTrack != 0) { 4315 if (mActiveTrack->mState != TrackBase::ACTIVE && 4316 mActiveTrack->mState != TrackBase::RESUMING) { 4317 unlockEffectChains(effectChains); 4318 usleep(kRecordThreadSleepUs); 4319 continue; 4320 } 4321 for (size_t i = 0; i < effectChains.size(); i ++) { 4322 effectChains[i]->process_l(); 4323 } 4324 4325 buffer.frameCount = mFrameCount; 4326 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4327 size_t framesOut = buffer.frameCount; 4328 if (mResampler == NULL) { 4329 // no resampling 4330 while (framesOut) { 4331 size_t framesIn = mFrameCount - mRsmpInIndex; 4332 if (framesIn) { 4333 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4334 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4335 if (framesIn > framesOut) 4336 framesIn = framesOut; 4337 mRsmpInIndex += framesIn; 4338 framesOut -= framesIn; 4339 if ((int)mChannelCount == mReqChannelCount || 4340 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4341 memcpy(dst, src, framesIn * mFrameSize); 4342 } else { 4343 int16_t *src16 = (int16_t *)src; 4344 int16_t *dst16 = (int16_t *)dst; 4345 if (mChannelCount == 1) { 4346 while (framesIn--) { 4347 *dst16++ = *src16; 4348 *dst16++ = *src16++; 4349 } 4350 } else { 4351 while (framesIn--) { 4352 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4353 src16 += 2; 4354 } 4355 } 4356 } 4357 } 4358 if (framesOut && mFrameCount == mRsmpInIndex) { 4359 if (framesOut == mFrameCount && 4360 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4361 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4362 framesOut = 0; 4363 } else { 4364 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4365 mRsmpInIndex = 0; 4366 } 4367 if (mBytesRead < 0) { 4368 ALOGE("Error reading audio input"); 4369 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4370 // Force input into standby so that it tries to 4371 // recover at next read attempt 4372 mInput->stream->common.standby(&mInput->stream->common); 4373 usleep(kRecordThreadSleepUs); 4374 } 4375 mRsmpInIndex = mFrameCount; 4376 framesOut = 0; 4377 buffer.frameCount = 0; 4378 } 4379 } 4380 } 4381 } else { 4382 // resampling 4383 4384 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4385 // alter output frame count as if we were expecting stereo samples 4386 if (mChannelCount == 1 && mReqChannelCount == 1) { 4387 framesOut >>= 1; 4388 } 4389 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4390 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4391 // are 32 bit aligned which should be always true. 4392 if (mChannelCount == 2 && mReqChannelCount == 1) { 4393 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4394 // the resampler always outputs stereo samples: do post stereo to mono conversion 4395 int16_t *src = (int16_t *)mRsmpOutBuffer; 4396 int16_t *dst = buffer.i16; 4397 while (framesOut--) { 4398 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4399 src += 2; 4400 } 4401 } else { 4402 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4403 } 4404 4405 } 4406 mActiveTrack->releaseBuffer(&buffer); 4407 mActiveTrack->overflow(); 4408 } 4409 // client isn't retrieving buffers fast enough 4410 else { 4411 if (!mActiveTrack->setOverflow()) { 4412 nsecs_t now = systemTime(); 4413 if ((now - lastWarning) > kWarningThrottleNs) { 4414 ALOGW("RecordThread: buffer overflow"); 4415 lastWarning = now; 4416 } 4417 } 4418 // Release the processor for a while before asking for a new buffer. 4419 // This will give the application more chance to read from the buffer and 4420 // clear the overflow. 4421 usleep(kRecordThreadSleepUs); 4422 } 4423 } 4424 // enable changes in effect chain 4425 unlockEffectChains(effectChains); 4426 effectChains.clear(); 4427 } 4428 4429 if (!mStandby) { 4430 mInput->stream->common.standby(&mInput->stream->common); 4431 } 4432 mActiveTrack.clear(); 4433 4434 mStartStopCond.broadcast(); 4435 4436 releaseWakeLock(); 4437 4438 ALOGV("RecordThread %p exiting", this); 4439 return false; 4440} 4441 4442 4443sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4444 const sp<AudioFlinger::Client>& client, 4445 uint32_t sampleRate, 4446 audio_format_t format, 4447 int channelMask, 4448 int frameCount, 4449 uint32_t flags, 4450 int sessionId, 4451 status_t *status) 4452{ 4453 sp<RecordTrack> track; 4454 status_t lStatus; 4455 4456 lStatus = initCheck(); 4457 if (lStatus != NO_ERROR) { 4458 ALOGE("Audio driver not initialized."); 4459 goto Exit; 4460 } 4461 4462 { // scope for mLock 4463 Mutex::Autolock _l(mLock); 4464 4465 track = new RecordTrack(this, client, sampleRate, 4466 format, channelMask, frameCount, flags, sessionId); 4467 4468 if (track->getCblk() == 0) { 4469 lStatus = NO_MEMORY; 4470 goto Exit; 4471 } 4472 4473 mTrack = track.get(); 4474 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4475 bool suspend = audio_is_bluetooth_sco_device( 4476 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4477 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4478 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4479 } 4480 lStatus = NO_ERROR; 4481 4482Exit: 4483 if (status) { 4484 *status = lStatus; 4485 } 4486 return track; 4487} 4488 4489status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4490{ 4491 ALOGV("RecordThread::start"); 4492 sp <ThreadBase> strongMe = this; 4493 status_t status = NO_ERROR; 4494 { 4495 AutoMutex lock(mLock); 4496 if (mActiveTrack != 0) { 4497 if (recordTrack != mActiveTrack.get()) { 4498 status = -EBUSY; 4499 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4500 mActiveTrack->mState = TrackBase::ACTIVE; 4501 } 4502 return status; 4503 } 4504 4505 recordTrack->mState = TrackBase::IDLE; 4506 mActiveTrack = recordTrack; 4507 mLock.unlock(); 4508 status_t status = AudioSystem::startInput(mId); 4509 mLock.lock(); 4510 if (status != NO_ERROR) { 4511 mActiveTrack.clear(); 4512 return status; 4513 } 4514 mRsmpInIndex = mFrameCount; 4515 mBytesRead = 0; 4516 if (mResampler != NULL) { 4517 mResampler->reset(); 4518 } 4519 mActiveTrack->mState = TrackBase::RESUMING; 4520 // signal thread to start 4521 ALOGV("Signal record thread"); 4522 mWaitWorkCV.signal(); 4523 // do not wait for mStartStopCond if exiting 4524 if (exitPending()) { 4525 mActiveTrack.clear(); 4526 status = INVALID_OPERATION; 4527 goto startError; 4528 } 4529 mStartStopCond.wait(mLock); 4530 if (mActiveTrack == 0) { 4531 ALOGV("Record failed to start"); 4532 status = BAD_VALUE; 4533 goto startError; 4534 } 4535 ALOGV("Record started OK"); 4536 return status; 4537 } 4538startError: 4539 AudioSystem::stopInput(mId); 4540 return status; 4541} 4542 4543void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4544 ALOGV("RecordThread::stop"); 4545 sp <ThreadBase> strongMe = this; 4546 { 4547 AutoMutex lock(mLock); 4548 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4549 mActiveTrack->mState = TrackBase::PAUSING; 4550 // do not wait for mStartStopCond if exiting 4551 if (exitPending()) { 4552 return; 4553 } 4554 mStartStopCond.wait(mLock); 4555 // if we have been restarted, recordTrack == mActiveTrack.get() here 4556 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4557 mLock.unlock(); 4558 AudioSystem::stopInput(mId); 4559 mLock.lock(); 4560 ALOGV("Record stopped OK"); 4561 } 4562 } 4563 } 4564} 4565 4566status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4567{ 4568 const size_t SIZE = 256; 4569 char buffer[SIZE]; 4570 String8 result; 4571 4572 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4573 result.append(buffer); 4574 4575 if (mActiveTrack != 0) { 4576 result.append("Active Track:\n"); 4577 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4578 mActiveTrack->dump(buffer, SIZE); 4579 result.append(buffer); 4580 4581 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4582 result.append(buffer); 4583 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4584 result.append(buffer); 4585 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4586 result.append(buffer); 4587 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4588 result.append(buffer); 4589 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4590 result.append(buffer); 4591 4592 4593 } else { 4594 result.append("No record client\n"); 4595 } 4596 write(fd, result.string(), result.size()); 4597 4598 dumpBase(fd, args); 4599 dumpEffectChains(fd, args); 4600 4601 return NO_ERROR; 4602} 4603 4604status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4605{ 4606 size_t framesReq = buffer->frameCount; 4607 size_t framesReady = mFrameCount - mRsmpInIndex; 4608 int channelCount; 4609 4610 if (framesReady == 0) { 4611 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4612 if (mBytesRead < 0) { 4613 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4614 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4615 // Force input into standby so that it tries to 4616 // recover at next read attempt 4617 mInput->stream->common.standby(&mInput->stream->common); 4618 usleep(kRecordThreadSleepUs); 4619 } 4620 buffer->raw = NULL; 4621 buffer->frameCount = 0; 4622 return NOT_ENOUGH_DATA; 4623 } 4624 mRsmpInIndex = 0; 4625 framesReady = mFrameCount; 4626 } 4627 4628 if (framesReq > framesReady) { 4629 framesReq = framesReady; 4630 } 4631 4632 if (mChannelCount == 1 && mReqChannelCount == 2) { 4633 channelCount = 1; 4634 } else { 4635 channelCount = 2; 4636 } 4637 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4638 buffer->frameCount = framesReq; 4639 return NO_ERROR; 4640} 4641 4642void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4643{ 4644 mRsmpInIndex += buffer->frameCount; 4645 buffer->frameCount = 0; 4646} 4647 4648bool AudioFlinger::RecordThread::checkForNewParameters_l() 4649{ 4650 bool reconfig = false; 4651 4652 while (!mNewParameters.isEmpty()) { 4653 status_t status = NO_ERROR; 4654 String8 keyValuePair = mNewParameters[0]; 4655 AudioParameter param = AudioParameter(keyValuePair); 4656 int value; 4657 audio_format_t reqFormat = mFormat; 4658 int reqSamplingRate = mReqSampleRate; 4659 int reqChannelCount = mReqChannelCount; 4660 4661 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4662 reqSamplingRate = value; 4663 reconfig = true; 4664 } 4665 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4666 reqFormat = (audio_format_t) value; 4667 reconfig = true; 4668 } 4669 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4670 reqChannelCount = popcount(value); 4671 reconfig = true; 4672 } 4673 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4674 // do not accept frame count changes if tracks are open as the track buffer 4675 // size depends on frame count and correct behavior would not be garantied 4676 // if frame count is changed after track creation 4677 if (mActiveTrack != 0) { 4678 status = INVALID_OPERATION; 4679 } else { 4680 reconfig = true; 4681 } 4682 } 4683 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4684 // forward device change to effects that have requested to be 4685 // aware of attached audio device. 4686 for (size_t i = 0; i < mEffectChains.size(); i++) { 4687 mEffectChains[i]->setDevice_l(value); 4688 } 4689 // store input device and output device but do not forward output device to audio HAL. 4690 // Note that status is ignored by the caller for output device 4691 // (see AudioFlinger::setParameters() 4692 if (value & AUDIO_DEVICE_OUT_ALL) { 4693 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4694 status = BAD_VALUE; 4695 } else { 4696 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4697 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4698 if (mTrack != NULL) { 4699 bool suspend = audio_is_bluetooth_sco_device( 4700 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4701 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4702 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4703 } 4704 } 4705 mDevice |= (uint32_t)value; 4706 } 4707 if (status == NO_ERROR) { 4708 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4709 if (status == INVALID_OPERATION) { 4710 mInput->stream->common.standby(&mInput->stream->common); 4711 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4712 } 4713 if (reconfig) { 4714 if (status == BAD_VALUE && 4715 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4716 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4717 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4718 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4719 (reqChannelCount < 3)) { 4720 status = NO_ERROR; 4721 } 4722 if (status == NO_ERROR) { 4723 readInputParameters(); 4724 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4725 } 4726 } 4727 } 4728 4729 mNewParameters.removeAt(0); 4730 4731 mParamStatus = status; 4732 mParamCond.signal(); 4733 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4734 // already timed out waiting for the status and will never signal the condition. 4735 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4736 } 4737 return reconfig; 4738} 4739 4740String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4741{ 4742 char *s; 4743 String8 out_s8 = String8(); 4744 4745 Mutex::Autolock _l(mLock); 4746 if (initCheck() != NO_ERROR) { 4747 return out_s8; 4748 } 4749 4750 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4751 out_s8 = String8(s); 4752 free(s); 4753 return out_s8; 4754} 4755 4756void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4757 AudioSystem::OutputDescriptor desc; 4758 void *param2 = NULL; 4759 4760 switch (event) { 4761 case AudioSystem::INPUT_OPENED: 4762 case AudioSystem::INPUT_CONFIG_CHANGED: 4763 desc.channels = mChannelMask; 4764 desc.samplingRate = mSampleRate; 4765 desc.format = mFormat; 4766 desc.frameCount = mFrameCount; 4767 desc.latency = 0; 4768 param2 = &desc; 4769 break; 4770 4771 case AudioSystem::INPUT_CLOSED: 4772 default: 4773 break; 4774 } 4775 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4776} 4777 4778void AudioFlinger::RecordThread::readInputParameters() 4779{ 4780 delete mRsmpInBuffer; 4781 // mRsmpInBuffer is always assigned a new[] below 4782 delete mRsmpOutBuffer; 4783 mRsmpOutBuffer = NULL; 4784 delete mResampler; 4785 mResampler = NULL; 4786 4787 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4788 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4789 mChannelCount = (uint16_t)popcount(mChannelMask); 4790 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4791 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4792 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4793 mFrameCount = mInputBytes / mFrameSize; 4794 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4795 4796 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4797 { 4798 int channelCount; 4799 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4800 // stereo to mono post process as the resampler always outputs stereo. 4801 if (mChannelCount == 1 && mReqChannelCount == 2) { 4802 channelCount = 1; 4803 } else { 4804 channelCount = 2; 4805 } 4806 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4807 mResampler->setSampleRate(mSampleRate); 4808 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4809 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4810 4811 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4812 if (mChannelCount == 1 && mReqChannelCount == 1) { 4813 mFrameCount >>= 1; 4814 } 4815 4816 } 4817 mRsmpInIndex = mFrameCount; 4818} 4819 4820unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4821{ 4822 Mutex::Autolock _l(mLock); 4823 if (initCheck() != NO_ERROR) { 4824 return 0; 4825 } 4826 4827 return mInput->stream->get_input_frames_lost(mInput->stream); 4828} 4829 4830uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4831{ 4832 Mutex::Autolock _l(mLock); 4833 uint32_t result = 0; 4834 if (getEffectChain_l(sessionId) != 0) { 4835 result = EFFECT_SESSION; 4836 } 4837 4838 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4839 result |= TRACK_SESSION; 4840 } 4841 4842 return result; 4843} 4844 4845AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4846{ 4847 Mutex::Autolock _l(mLock); 4848 return mTrack; 4849} 4850 4851AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4852{ 4853 Mutex::Autolock _l(mLock); 4854 return mInput; 4855} 4856 4857AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4858{ 4859 Mutex::Autolock _l(mLock); 4860 AudioStreamIn *input = mInput; 4861 mInput = NULL; 4862 return input; 4863} 4864 4865// this method must always be called either with ThreadBase mLock held or inside the thread loop 4866audio_stream_t* AudioFlinger::RecordThread::stream() 4867{ 4868 if (mInput == NULL) { 4869 return NULL; 4870 } 4871 return &mInput->stream->common; 4872} 4873 4874 4875// ---------------------------------------------------------------------------- 4876 4877audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 4878 uint32_t *pSamplingRate, 4879 audio_format_t *pFormat, 4880 uint32_t *pChannels, 4881 uint32_t *pLatencyMs, 4882 uint32_t flags) 4883{ 4884 status_t status; 4885 PlaybackThread *thread = NULL; 4886 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4887 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4888 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4889 uint32_t channels = pChannels ? *pChannels : 0; 4890 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4891 audio_stream_out_t *outStream; 4892 audio_hw_device_t *outHwDev; 4893 4894 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4895 pDevices ? *pDevices : 0, 4896 samplingRate, 4897 format, 4898 channels, 4899 flags); 4900 4901 if (pDevices == NULL || *pDevices == 0) { 4902 return 0; 4903 } 4904 4905 Mutex::Autolock _l(mLock); 4906 4907 outHwDev = findSuitableHwDev_l(*pDevices); 4908 if (outHwDev == NULL) 4909 return 0; 4910 4911 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4912 &channels, &samplingRate, &outStream); 4913 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4914 outStream, 4915 samplingRate, 4916 format, 4917 channels, 4918 status); 4919 4920 mHardwareStatus = AUDIO_HW_IDLE; 4921 if (outStream != NULL) { 4922 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4923 audio_io_handle_t id = nextUniqueId(); 4924 4925 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4926 (format != AUDIO_FORMAT_PCM_16_BIT) || 4927 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4928 thread = new DirectOutputThread(this, output, id, *pDevices); 4929 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4930 } else { 4931 thread = new MixerThread(this, output, id, *pDevices); 4932 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4933 } 4934 mPlaybackThreads.add(id, thread); 4935 4936 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 4937 if (pFormat != NULL) *pFormat = format; 4938 if (pChannels != NULL) *pChannels = channels; 4939 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 4940 4941 // notify client processes of the new output creation 4942 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4943 return id; 4944 } 4945 4946 return 0; 4947} 4948 4949audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 4950 audio_io_handle_t output2) 4951{ 4952 Mutex::Autolock _l(mLock); 4953 MixerThread *thread1 = checkMixerThread_l(output1); 4954 MixerThread *thread2 = checkMixerThread_l(output2); 4955 4956 if (thread1 == NULL || thread2 == NULL) { 4957 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4958 return 0; 4959 } 4960 4961 audio_io_handle_t id = nextUniqueId(); 4962 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4963 thread->addOutputTrack(thread2); 4964 mPlaybackThreads.add(id, thread); 4965 // notify client processes of the new output creation 4966 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4967 return id; 4968} 4969 4970status_t AudioFlinger::closeOutput(audio_io_handle_t output) 4971{ 4972 // keep strong reference on the playback thread so that 4973 // it is not destroyed while exit() is executed 4974 sp <PlaybackThread> thread; 4975 { 4976 Mutex::Autolock _l(mLock); 4977 thread = checkPlaybackThread_l(output); 4978 if (thread == NULL) { 4979 return BAD_VALUE; 4980 } 4981 4982 ALOGV("closeOutput() %d", output); 4983 4984 if (thread->type() == ThreadBase::MIXER) { 4985 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4986 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4987 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4988 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4989 } 4990 } 4991 } 4992 void *param2 = NULL; 4993 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4994 mPlaybackThreads.removeItem(output); 4995 } 4996 thread->exit(); 4997 // The thread entity (active unit of execution) is no longer running here, 4998 // but the ThreadBase container still exists. 4999 5000 if (thread->type() != ThreadBase::DUPLICATING) { 5001 AudioStreamOut *out = thread->clearOutput(); 5002 assert(out != NULL); 5003 // from now on thread->mOutput is NULL 5004 out->hwDev->close_output_stream(out->hwDev, out->stream); 5005 delete out; 5006 } 5007 return NO_ERROR; 5008} 5009 5010status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5011{ 5012 Mutex::Autolock _l(mLock); 5013 PlaybackThread *thread = checkPlaybackThread_l(output); 5014 5015 if (thread == NULL) { 5016 return BAD_VALUE; 5017 } 5018 5019 ALOGV("suspendOutput() %d", output); 5020 thread->suspend(); 5021 5022 return NO_ERROR; 5023} 5024 5025status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5026{ 5027 Mutex::Autolock _l(mLock); 5028 PlaybackThread *thread = checkPlaybackThread_l(output); 5029 5030 if (thread == NULL) { 5031 return BAD_VALUE; 5032 } 5033 5034 ALOGV("restoreOutput() %d", output); 5035 5036 thread->restore(); 5037 5038 return NO_ERROR; 5039} 5040 5041audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5042 uint32_t *pSamplingRate, 5043 audio_format_t *pFormat, 5044 uint32_t *pChannels, 5045 audio_in_acoustics_t acoustics) 5046{ 5047 status_t status; 5048 RecordThread *thread = NULL; 5049 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5050 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5051 uint32_t channels = pChannels ? *pChannels : 0; 5052 uint32_t reqSamplingRate = samplingRate; 5053 audio_format_t reqFormat = format; 5054 uint32_t reqChannels = channels; 5055 audio_stream_in_t *inStream; 5056 audio_hw_device_t *inHwDev; 5057 5058 if (pDevices == NULL || *pDevices == 0) { 5059 return 0; 5060 } 5061 5062 Mutex::Autolock _l(mLock); 5063 5064 inHwDev = findSuitableHwDev_l(*pDevices); 5065 if (inHwDev == NULL) 5066 return 0; 5067 5068 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5069 &channels, &samplingRate, 5070 acoustics, 5071 &inStream); 5072 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5073 inStream, 5074 samplingRate, 5075 format, 5076 channels, 5077 acoustics, 5078 status); 5079 5080 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5081 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5082 // or stereo to mono conversions on 16 bit PCM inputs. 5083 if (inStream == NULL && status == BAD_VALUE && 5084 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5085 (samplingRate <= 2 * reqSamplingRate) && 5086 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5087 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5088 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5089 &channels, &samplingRate, 5090 acoustics, 5091 &inStream); 5092 } 5093 5094 if (inStream != NULL) { 5095 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5096 5097 audio_io_handle_t id = nextUniqueId(); 5098 // Start record thread 5099 // RecorThread require both input and output device indication to forward to audio 5100 // pre processing modules 5101 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5102 thread = new RecordThread(this, 5103 input, 5104 reqSamplingRate, 5105 reqChannels, 5106 id, 5107 device); 5108 mRecordThreads.add(id, thread); 5109 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5110 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5111 if (pFormat != NULL) *pFormat = format; 5112 if (pChannels != NULL) *pChannels = reqChannels; 5113 5114 input->stream->common.standby(&input->stream->common); 5115 5116 // notify client processes of the new input creation 5117 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5118 return id; 5119 } 5120 5121 return 0; 5122} 5123 5124status_t AudioFlinger::closeInput(audio_io_handle_t input) 5125{ 5126 // keep strong reference on the record thread so that 5127 // it is not destroyed while exit() is executed 5128 sp <RecordThread> thread; 5129 { 5130 Mutex::Autolock _l(mLock); 5131 thread = checkRecordThread_l(input); 5132 if (thread == NULL) { 5133 return BAD_VALUE; 5134 } 5135 5136 ALOGV("closeInput() %d", input); 5137 void *param2 = NULL; 5138 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5139 mRecordThreads.removeItem(input); 5140 } 5141 thread->exit(); 5142 // The thread entity (active unit of execution) is no longer running here, 5143 // but the ThreadBase container still exists. 5144 5145 AudioStreamIn *in = thread->clearInput(); 5146 assert(in != NULL); 5147 // from now on thread->mInput is NULL 5148 in->hwDev->close_input_stream(in->hwDev, in->stream); 5149 delete in; 5150 5151 return NO_ERROR; 5152} 5153 5154status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5155{ 5156 Mutex::Autolock _l(mLock); 5157 MixerThread *dstThread = checkMixerThread_l(output); 5158 if (dstThread == NULL) { 5159 ALOGW("setStreamOutput() bad output id %d", output); 5160 return BAD_VALUE; 5161 } 5162 5163 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5164 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5165 5166 dstThread->setStreamValid(stream, true); 5167 5168 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5169 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5170 if (thread != dstThread && 5171 thread->type() != ThreadBase::DIRECT) { 5172 MixerThread *srcThread = (MixerThread *)thread; 5173 srcThread->setStreamValid(stream, false); 5174 srcThread->invalidateTracks(stream); 5175 } 5176 } 5177 5178 return NO_ERROR; 5179} 5180 5181 5182int AudioFlinger::newAudioSessionId() 5183{ 5184 return nextUniqueId(); 5185} 5186 5187void AudioFlinger::acquireAudioSessionId(int audioSession) 5188{ 5189 Mutex::Autolock _l(mLock); 5190 pid_t caller = IPCThreadState::self()->getCallingPid(); 5191 ALOGV("acquiring %d from %d", audioSession, caller); 5192 int num = mAudioSessionRefs.size(); 5193 for (int i = 0; i< num; i++) { 5194 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5195 if (ref->sessionid == audioSession && ref->pid == caller) { 5196 ref->cnt++; 5197 ALOGV(" incremented refcount to %d", ref->cnt); 5198 return; 5199 } 5200 } 5201 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5202 ALOGV(" added new entry for %d", audioSession); 5203} 5204 5205void AudioFlinger::releaseAudioSessionId(int audioSession) 5206{ 5207 Mutex::Autolock _l(mLock); 5208 pid_t caller = IPCThreadState::self()->getCallingPid(); 5209 ALOGV("releasing %d from %d", audioSession, caller); 5210 int num = mAudioSessionRefs.size(); 5211 for (int i = 0; i< num; i++) { 5212 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5213 if (ref->sessionid == audioSession && ref->pid == caller) { 5214 ref->cnt--; 5215 ALOGV(" decremented refcount to %d", ref->cnt); 5216 if (ref->cnt == 0) { 5217 mAudioSessionRefs.removeAt(i); 5218 delete ref; 5219 purgeStaleEffects_l(); 5220 } 5221 return; 5222 } 5223 } 5224 ALOGW("session id %d not found for pid %d", audioSession, caller); 5225} 5226 5227void AudioFlinger::purgeStaleEffects_l() { 5228 5229 ALOGV("purging stale effects"); 5230 5231 Vector< sp<EffectChain> > chains; 5232 5233 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5234 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5235 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5236 sp<EffectChain> ec = t->mEffectChains[j]; 5237 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5238 chains.push(ec); 5239 } 5240 } 5241 } 5242 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5243 sp<RecordThread> t = mRecordThreads.valueAt(i); 5244 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5245 sp<EffectChain> ec = t->mEffectChains[j]; 5246 chains.push(ec); 5247 } 5248 } 5249 5250 for (size_t i = 0; i < chains.size(); i++) { 5251 sp<EffectChain> ec = chains[i]; 5252 int sessionid = ec->sessionId(); 5253 sp<ThreadBase> t = ec->mThread.promote(); 5254 if (t == 0) { 5255 continue; 5256 } 5257 size_t numsessionrefs = mAudioSessionRefs.size(); 5258 bool found = false; 5259 for (size_t k = 0; k < numsessionrefs; k++) { 5260 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5261 if (ref->sessionid == sessionid) { 5262 ALOGV(" session %d still exists for %d with %d refs", 5263 sessionid, ref->pid, ref->cnt); 5264 found = true; 5265 break; 5266 } 5267 } 5268 if (!found) { 5269 // remove all effects from the chain 5270 while (ec->mEffects.size()) { 5271 sp<EffectModule> effect = ec->mEffects[0]; 5272 effect->unPin(); 5273 Mutex::Autolock _l (t->mLock); 5274 t->removeEffect_l(effect); 5275 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5276 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5277 if (handle != 0) { 5278 handle->mEffect.clear(); 5279 if (handle->mHasControl && handle->mEnabled) { 5280 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5281 } 5282 } 5283 } 5284 AudioSystem::unregisterEffect(effect->id()); 5285 } 5286 } 5287 } 5288 return; 5289} 5290 5291// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5292AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5293{ 5294 PlaybackThread *thread = NULL; 5295 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5296 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5297 } 5298 return thread; 5299} 5300 5301// checkMixerThread_l() must be called with AudioFlinger::mLock held 5302AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5303{ 5304 PlaybackThread *thread = checkPlaybackThread_l(output); 5305 if (thread != NULL) { 5306 if (thread->type() == ThreadBase::DIRECT) { 5307 thread = NULL; 5308 } 5309 } 5310 return (MixerThread *)thread; 5311} 5312 5313// checkRecordThread_l() must be called with AudioFlinger::mLock held 5314AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5315{ 5316 RecordThread *thread = NULL; 5317 if (mRecordThreads.indexOfKey(input) >= 0) { 5318 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5319 } 5320 return thread; 5321} 5322 5323uint32_t AudioFlinger::nextUniqueId() 5324{ 5325 return android_atomic_inc(&mNextUniqueId); 5326} 5327 5328AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5329{ 5330 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5331 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5332 AudioStreamOut *output = thread->getOutput(); 5333 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5334 return thread; 5335 } 5336 } 5337 return NULL; 5338} 5339 5340uint32_t AudioFlinger::primaryOutputDevice_l() 5341{ 5342 PlaybackThread *thread = primaryPlaybackThread_l(); 5343 5344 if (thread == NULL) { 5345 return 0; 5346 } 5347 5348 return thread->device(); 5349} 5350 5351 5352// ---------------------------------------------------------------------------- 5353// Effect management 5354// ---------------------------------------------------------------------------- 5355 5356 5357status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5358{ 5359 Mutex::Autolock _l(mLock); 5360 return EffectQueryNumberEffects(numEffects); 5361} 5362 5363status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5364{ 5365 Mutex::Autolock _l(mLock); 5366 return EffectQueryEffect(index, descriptor); 5367} 5368 5369status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5370 effect_descriptor_t *descriptor) const 5371{ 5372 Mutex::Autolock _l(mLock); 5373 return EffectGetDescriptor(pUuid, descriptor); 5374} 5375 5376 5377sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5378 effect_descriptor_t *pDesc, 5379 const sp<IEffectClient>& effectClient, 5380 int32_t priority, 5381 audio_io_handle_t io, 5382 int sessionId, 5383 status_t *status, 5384 int *id, 5385 int *enabled) 5386{ 5387 status_t lStatus = NO_ERROR; 5388 sp<EffectHandle> handle; 5389 effect_descriptor_t desc; 5390 5391 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5392 pid, effectClient.get(), priority, sessionId, io); 5393 5394 if (pDesc == NULL) { 5395 lStatus = BAD_VALUE; 5396 goto Exit; 5397 } 5398 5399 // check audio settings permission for global effects 5400 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5401 lStatus = PERMISSION_DENIED; 5402 goto Exit; 5403 } 5404 5405 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5406 // that can only be created by audio policy manager (running in same process) 5407 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5408 lStatus = PERMISSION_DENIED; 5409 goto Exit; 5410 } 5411 5412 if (io == 0) { 5413 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5414 // output must be specified by AudioPolicyManager when using session 5415 // AUDIO_SESSION_OUTPUT_STAGE 5416 lStatus = BAD_VALUE; 5417 goto Exit; 5418 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5419 // if the output returned by getOutputForEffect() is removed before we lock the 5420 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5421 // and we will exit safely 5422 io = AudioSystem::getOutputForEffect(&desc); 5423 } 5424 } 5425 5426 { 5427 Mutex::Autolock _l(mLock); 5428 5429 5430 if (!EffectIsNullUuid(&pDesc->uuid)) { 5431 // if uuid is specified, request effect descriptor 5432 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5433 if (lStatus < 0) { 5434 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5435 goto Exit; 5436 } 5437 } else { 5438 // if uuid is not specified, look for an available implementation 5439 // of the required type in effect factory 5440 if (EffectIsNullUuid(&pDesc->type)) { 5441 ALOGW("createEffect() no effect type"); 5442 lStatus = BAD_VALUE; 5443 goto Exit; 5444 } 5445 uint32_t numEffects = 0; 5446 effect_descriptor_t d; 5447 d.flags = 0; // prevent compiler warning 5448 bool found = false; 5449 5450 lStatus = EffectQueryNumberEffects(&numEffects); 5451 if (lStatus < 0) { 5452 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5453 goto Exit; 5454 } 5455 for (uint32_t i = 0; i < numEffects; i++) { 5456 lStatus = EffectQueryEffect(i, &desc); 5457 if (lStatus < 0) { 5458 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5459 continue; 5460 } 5461 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5462 // If matching type found save effect descriptor. If the session is 5463 // 0 and the effect is not auxiliary, continue enumeration in case 5464 // an auxiliary version of this effect type is available 5465 found = true; 5466 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5467 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5468 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5469 break; 5470 } 5471 } 5472 } 5473 if (!found) { 5474 lStatus = BAD_VALUE; 5475 ALOGW("createEffect() effect not found"); 5476 goto Exit; 5477 } 5478 // For same effect type, chose auxiliary version over insert version if 5479 // connect to output mix (Compliance to OpenSL ES) 5480 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5481 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5482 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5483 } 5484 } 5485 5486 // Do not allow auxiliary effects on a session different from 0 (output mix) 5487 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5488 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5489 lStatus = INVALID_OPERATION; 5490 goto Exit; 5491 } 5492 5493 // check recording permission for visualizer 5494 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5495 !recordingAllowed()) { 5496 lStatus = PERMISSION_DENIED; 5497 goto Exit; 5498 } 5499 5500 // return effect descriptor 5501 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5502 5503 // If output is not specified try to find a matching audio session ID in one of the 5504 // output threads. 5505 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5506 // because of code checking output when entering the function. 5507 // Note: io is never 0 when creating an effect on an input 5508 if (io == 0) { 5509 // look for the thread where the specified audio session is present 5510 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5511 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5512 io = mPlaybackThreads.keyAt(i); 5513 break; 5514 } 5515 } 5516 if (io == 0) { 5517 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5518 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5519 io = mRecordThreads.keyAt(i); 5520 break; 5521 } 5522 } 5523 } 5524 // If no output thread contains the requested session ID, default to 5525 // first output. The effect chain will be moved to the correct output 5526 // thread when a track with the same session ID is created 5527 if (io == 0 && mPlaybackThreads.size()) { 5528 io = mPlaybackThreads.keyAt(0); 5529 } 5530 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5531 } 5532 ThreadBase *thread = checkRecordThread_l(io); 5533 if (thread == NULL) { 5534 thread = checkPlaybackThread_l(io); 5535 if (thread == NULL) { 5536 ALOGE("createEffect() unknown output thread"); 5537 lStatus = BAD_VALUE; 5538 goto Exit; 5539 } 5540 } 5541 5542 sp<Client> client = registerPid_l(pid); 5543 5544 // create effect on selected output thread 5545 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5546 &desc, enabled, &lStatus); 5547 if (handle != 0 && id != NULL) { 5548 *id = handle->id(); 5549 } 5550 } 5551 5552Exit: 5553 if(status) { 5554 *status = lStatus; 5555 } 5556 return handle; 5557} 5558 5559status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 5560 audio_io_handle_t dstOutput) 5561{ 5562 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5563 sessionId, srcOutput, dstOutput); 5564 Mutex::Autolock _l(mLock); 5565 if (srcOutput == dstOutput) { 5566 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5567 return NO_ERROR; 5568 } 5569 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5570 if (srcThread == NULL) { 5571 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5572 return BAD_VALUE; 5573 } 5574 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5575 if (dstThread == NULL) { 5576 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5577 return BAD_VALUE; 5578 } 5579 5580 Mutex::Autolock _dl(dstThread->mLock); 5581 Mutex::Autolock _sl(srcThread->mLock); 5582 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5583 5584 return NO_ERROR; 5585} 5586 5587// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5588status_t AudioFlinger::moveEffectChain_l(int sessionId, 5589 AudioFlinger::PlaybackThread *srcThread, 5590 AudioFlinger::PlaybackThread *dstThread, 5591 bool reRegister) 5592{ 5593 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5594 sessionId, srcThread, dstThread); 5595 5596 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5597 if (chain == 0) { 5598 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5599 sessionId, srcThread); 5600 return INVALID_OPERATION; 5601 } 5602 5603 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5604 // so that a new chain is created with correct parameters when first effect is added. This is 5605 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5606 // removed. 5607 srcThread->removeEffectChain_l(chain); 5608 5609 // transfer all effects one by one so that new effect chain is created on new thread with 5610 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5611 audio_io_handle_t dstOutput = dstThread->id(); 5612 sp<EffectChain> dstChain; 5613 uint32_t strategy = 0; // prevent compiler warning 5614 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5615 while (effect != 0) { 5616 srcThread->removeEffect_l(effect); 5617 dstThread->addEffect_l(effect); 5618 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5619 if (effect->state() == EffectModule::ACTIVE || 5620 effect->state() == EffectModule::STOPPING) { 5621 effect->start(); 5622 } 5623 // if the move request is not received from audio policy manager, the effect must be 5624 // re-registered with the new strategy and output 5625 if (dstChain == 0) { 5626 dstChain = effect->chain().promote(); 5627 if (dstChain == 0) { 5628 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5629 srcThread->addEffect_l(effect); 5630 return NO_INIT; 5631 } 5632 strategy = dstChain->strategy(); 5633 } 5634 if (reRegister) { 5635 AudioSystem::unregisterEffect(effect->id()); 5636 AudioSystem::registerEffect(&effect->desc(), 5637 dstOutput, 5638 strategy, 5639 sessionId, 5640 effect->id()); 5641 } 5642 effect = chain->getEffectFromId_l(0); 5643 } 5644 5645 return NO_ERROR; 5646} 5647 5648 5649// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5650sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5651 const sp<AudioFlinger::Client>& client, 5652 const sp<IEffectClient>& effectClient, 5653 int32_t priority, 5654 int sessionId, 5655 effect_descriptor_t *desc, 5656 int *enabled, 5657 status_t *status 5658 ) 5659{ 5660 sp<EffectModule> effect; 5661 sp<EffectHandle> handle; 5662 status_t lStatus; 5663 sp<EffectChain> chain; 5664 bool chainCreated = false; 5665 bool effectCreated = false; 5666 bool effectRegistered = false; 5667 5668 lStatus = initCheck(); 5669 if (lStatus != NO_ERROR) { 5670 ALOGW("createEffect_l() Audio driver not initialized."); 5671 goto Exit; 5672 } 5673 5674 // Do not allow effects with session ID 0 on direct output or duplicating threads 5675 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5676 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5677 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5678 desc->name, sessionId); 5679 lStatus = BAD_VALUE; 5680 goto Exit; 5681 } 5682 // Only Pre processor effects are allowed on input threads and only on input threads 5683 if ((mType == RECORD && 5684 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5685 (mType != RECORD && 5686 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5687 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5688 desc->name, desc->flags, mType); 5689 lStatus = BAD_VALUE; 5690 goto Exit; 5691 } 5692 5693 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5694 5695 { // scope for mLock 5696 Mutex::Autolock _l(mLock); 5697 5698 // check for existing effect chain with the requested audio session 5699 chain = getEffectChain_l(sessionId); 5700 if (chain == 0) { 5701 // create a new chain for this session 5702 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5703 chain = new EffectChain(this, sessionId); 5704 addEffectChain_l(chain); 5705 chain->setStrategy(getStrategyForSession_l(sessionId)); 5706 chainCreated = true; 5707 } else { 5708 effect = chain->getEffectFromDesc_l(desc); 5709 } 5710 5711 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5712 5713 if (effect == 0) { 5714 int id = mAudioFlinger->nextUniqueId(); 5715 // Check CPU and memory usage 5716 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5717 if (lStatus != NO_ERROR) { 5718 goto Exit; 5719 } 5720 effectRegistered = true; 5721 // create a new effect module if none present in the chain 5722 effect = new EffectModule(this, chain, desc, id, sessionId); 5723 lStatus = effect->status(); 5724 if (lStatus != NO_ERROR) { 5725 goto Exit; 5726 } 5727 lStatus = chain->addEffect_l(effect); 5728 if (lStatus != NO_ERROR) { 5729 goto Exit; 5730 } 5731 effectCreated = true; 5732 5733 effect->setDevice(mDevice); 5734 effect->setMode(mAudioFlinger->getMode()); 5735 } 5736 // create effect handle and connect it to effect module 5737 handle = new EffectHandle(effect, client, effectClient, priority); 5738 lStatus = effect->addHandle(handle); 5739 if (enabled != NULL) { 5740 *enabled = (int)effect->isEnabled(); 5741 } 5742 } 5743 5744Exit: 5745 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5746 Mutex::Autolock _l(mLock); 5747 if (effectCreated) { 5748 chain->removeEffect_l(effect); 5749 } 5750 if (effectRegistered) { 5751 AudioSystem::unregisterEffect(effect->id()); 5752 } 5753 if (chainCreated) { 5754 removeEffectChain_l(chain); 5755 } 5756 handle.clear(); 5757 } 5758 5759 if(status) { 5760 *status = lStatus; 5761 } 5762 return handle; 5763} 5764 5765sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5766{ 5767 sp<EffectChain> chain = getEffectChain_l(sessionId); 5768 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 5769} 5770 5771// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5772// PlaybackThread::mLock held 5773status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5774{ 5775 // check for existing effect chain with the requested audio session 5776 int sessionId = effect->sessionId(); 5777 sp<EffectChain> chain = getEffectChain_l(sessionId); 5778 bool chainCreated = false; 5779 5780 if (chain == 0) { 5781 // create a new chain for this session 5782 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5783 chain = new EffectChain(this, sessionId); 5784 addEffectChain_l(chain); 5785 chain->setStrategy(getStrategyForSession_l(sessionId)); 5786 chainCreated = true; 5787 } 5788 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5789 5790 if (chain->getEffectFromId_l(effect->id()) != 0) { 5791 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5792 this, effect->desc().name, chain.get()); 5793 return BAD_VALUE; 5794 } 5795 5796 status_t status = chain->addEffect_l(effect); 5797 if (status != NO_ERROR) { 5798 if (chainCreated) { 5799 removeEffectChain_l(chain); 5800 } 5801 return status; 5802 } 5803 5804 effect->setDevice(mDevice); 5805 effect->setMode(mAudioFlinger->getMode()); 5806 return NO_ERROR; 5807} 5808 5809void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5810 5811 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5812 effect_descriptor_t desc = effect->desc(); 5813 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5814 detachAuxEffect_l(effect->id()); 5815 } 5816 5817 sp<EffectChain> chain = effect->chain().promote(); 5818 if (chain != 0) { 5819 // remove effect chain if removing last effect 5820 if (chain->removeEffect_l(effect) == 0) { 5821 removeEffectChain_l(chain); 5822 } 5823 } else { 5824 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5825 } 5826} 5827 5828void AudioFlinger::ThreadBase::lockEffectChains_l( 5829 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5830{ 5831 effectChains = mEffectChains; 5832 for (size_t i = 0; i < mEffectChains.size(); i++) { 5833 mEffectChains[i]->lock(); 5834 } 5835} 5836 5837void AudioFlinger::ThreadBase::unlockEffectChains( 5838 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5839{ 5840 for (size_t i = 0; i < effectChains.size(); i++) { 5841 effectChains[i]->unlock(); 5842 } 5843} 5844 5845sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5846{ 5847 Mutex::Autolock _l(mLock); 5848 return getEffectChain_l(sessionId); 5849} 5850 5851sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5852{ 5853 size_t size = mEffectChains.size(); 5854 for (size_t i = 0; i < size; i++) { 5855 if (mEffectChains[i]->sessionId() == sessionId) { 5856 return mEffectChains[i]; 5857 } 5858 } 5859 return 0; 5860} 5861 5862void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5863{ 5864 Mutex::Autolock _l(mLock); 5865 size_t size = mEffectChains.size(); 5866 for (size_t i = 0; i < size; i++) { 5867 mEffectChains[i]->setMode_l(mode); 5868 } 5869} 5870 5871void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5872 const wp<EffectHandle>& handle, 5873 bool unpinIfLast) { 5874 5875 Mutex::Autolock _l(mLock); 5876 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5877 // delete the effect module if removing last handle on it 5878 if (effect->removeHandle(handle) == 0) { 5879 if (!effect->isPinned() || unpinIfLast) { 5880 removeEffect_l(effect); 5881 AudioSystem::unregisterEffect(effect->id()); 5882 } 5883 } 5884} 5885 5886status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5887{ 5888 int session = chain->sessionId(); 5889 int16_t *buffer = mMixBuffer; 5890 bool ownsBuffer = false; 5891 5892 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5893 if (session > 0) { 5894 // Only one effect chain can be present in direct output thread and it uses 5895 // the mix buffer as input 5896 if (mType != DIRECT) { 5897 size_t numSamples = mFrameCount * mChannelCount; 5898 buffer = new int16_t[numSamples]; 5899 memset(buffer, 0, numSamples * sizeof(int16_t)); 5900 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5901 ownsBuffer = true; 5902 } 5903 5904 // Attach all tracks with same session ID to this chain. 5905 for (size_t i = 0; i < mTracks.size(); ++i) { 5906 sp<Track> track = mTracks[i]; 5907 if (session == track->sessionId()) { 5908 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5909 track->setMainBuffer(buffer); 5910 chain->incTrackCnt(); 5911 } 5912 } 5913 5914 // indicate all active tracks in the chain 5915 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5916 sp<Track> track = mActiveTracks[i].promote(); 5917 if (track == 0) continue; 5918 if (session == track->sessionId()) { 5919 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5920 chain->incActiveTrackCnt(); 5921 } 5922 } 5923 } 5924 5925 chain->setInBuffer(buffer, ownsBuffer); 5926 chain->setOutBuffer(mMixBuffer); 5927 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5928 // chains list in order to be processed last as it contains output stage effects 5929 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5930 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5931 // after track specific effects and before output stage 5932 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5933 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5934 // Effect chain for other sessions are inserted at beginning of effect 5935 // chains list to be processed before output mix effects. Relative order between other 5936 // sessions is not important 5937 size_t size = mEffectChains.size(); 5938 size_t i = 0; 5939 for (i = 0; i < size; i++) { 5940 if (mEffectChains[i]->sessionId() < session) break; 5941 } 5942 mEffectChains.insertAt(chain, i); 5943 checkSuspendOnAddEffectChain_l(chain); 5944 5945 return NO_ERROR; 5946} 5947 5948size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5949{ 5950 int session = chain->sessionId(); 5951 5952 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5953 5954 for (size_t i = 0; i < mEffectChains.size(); i++) { 5955 if (chain == mEffectChains[i]) { 5956 mEffectChains.removeAt(i); 5957 // detach all active tracks from the chain 5958 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5959 sp<Track> track = mActiveTracks[i].promote(); 5960 if (track == 0) continue; 5961 if (session == track->sessionId()) { 5962 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5963 chain.get(), session); 5964 chain->decActiveTrackCnt(); 5965 } 5966 } 5967 5968 // detach all tracks with same session ID from this chain 5969 for (size_t i = 0; i < mTracks.size(); ++i) { 5970 sp<Track> track = mTracks[i]; 5971 if (session == track->sessionId()) { 5972 track->setMainBuffer(mMixBuffer); 5973 chain->decTrackCnt(); 5974 } 5975 } 5976 break; 5977 } 5978 } 5979 return mEffectChains.size(); 5980} 5981 5982status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5983 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5984{ 5985 Mutex::Autolock _l(mLock); 5986 return attachAuxEffect_l(track, EffectId); 5987} 5988 5989status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5990 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5991{ 5992 status_t status = NO_ERROR; 5993 5994 if (EffectId == 0) { 5995 track->setAuxBuffer(0, NULL); 5996 } else { 5997 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 5998 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 5999 if (effect != 0) { 6000 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6001 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6002 } else { 6003 status = INVALID_OPERATION; 6004 } 6005 } else { 6006 status = BAD_VALUE; 6007 } 6008 } 6009 return status; 6010} 6011 6012void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6013{ 6014 for (size_t i = 0; i < mTracks.size(); ++i) { 6015 sp<Track> track = mTracks[i]; 6016 if (track->auxEffectId() == effectId) { 6017 attachAuxEffect_l(track, 0); 6018 } 6019 } 6020} 6021 6022status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6023{ 6024 // only one chain per input thread 6025 if (mEffectChains.size() != 0) { 6026 return INVALID_OPERATION; 6027 } 6028 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6029 6030 chain->setInBuffer(NULL); 6031 chain->setOutBuffer(NULL); 6032 6033 checkSuspendOnAddEffectChain_l(chain); 6034 6035 mEffectChains.add(chain); 6036 6037 return NO_ERROR; 6038} 6039 6040size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6041{ 6042 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6043 ALOGW_IF(mEffectChains.size() != 1, 6044 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6045 chain.get(), mEffectChains.size(), this); 6046 if (mEffectChains.size() == 1) { 6047 mEffectChains.removeAt(0); 6048 } 6049 return 0; 6050} 6051 6052// ---------------------------------------------------------------------------- 6053// EffectModule implementation 6054// ---------------------------------------------------------------------------- 6055 6056#undef LOG_TAG 6057#define LOG_TAG "AudioFlinger::EffectModule" 6058 6059AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6060 const wp<AudioFlinger::EffectChain>& chain, 6061 effect_descriptor_t *desc, 6062 int id, 6063 int sessionId) 6064 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6065 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6066{ 6067 ALOGV("Constructor %p", this); 6068 int lStatus; 6069 sp<ThreadBase> thread = mThread.promote(); 6070 if (thread == 0) { 6071 return; 6072 } 6073 6074 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6075 6076 // create effect engine from effect factory 6077 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6078 6079 if (mStatus != NO_ERROR) { 6080 return; 6081 } 6082 lStatus = init(); 6083 if (lStatus < 0) { 6084 mStatus = lStatus; 6085 goto Error; 6086 } 6087 6088 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6089 mPinned = true; 6090 } 6091 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6092 return; 6093Error: 6094 EffectRelease(mEffectInterface); 6095 mEffectInterface = NULL; 6096 ALOGV("Constructor Error %d", mStatus); 6097} 6098 6099AudioFlinger::EffectModule::~EffectModule() 6100{ 6101 ALOGV("Destructor %p", this); 6102 if (mEffectInterface != NULL) { 6103 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6104 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6105 sp<ThreadBase> thread = mThread.promote(); 6106 if (thread != 0) { 6107 audio_stream_t *stream = thread->stream(); 6108 if (stream != NULL) { 6109 stream->remove_audio_effect(stream, mEffectInterface); 6110 } 6111 } 6112 } 6113 // release effect engine 6114 EffectRelease(mEffectInterface); 6115 } 6116} 6117 6118status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6119{ 6120 status_t status; 6121 6122 Mutex::Autolock _l(mLock); 6123 // First handle in mHandles has highest priority and controls the effect module 6124 int priority = handle->priority(); 6125 size_t size = mHandles.size(); 6126 sp<EffectHandle> h; 6127 size_t i; 6128 for (i = 0; i < size; i++) { 6129 h = mHandles[i].promote(); 6130 if (h == 0) continue; 6131 if (h->priority() <= priority) break; 6132 } 6133 // if inserted in first place, move effect control from previous owner to this handle 6134 if (i == 0) { 6135 bool enabled = false; 6136 if (h != 0) { 6137 enabled = h->enabled(); 6138 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6139 } 6140 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6141 status = NO_ERROR; 6142 } else { 6143 status = ALREADY_EXISTS; 6144 } 6145 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6146 mHandles.insertAt(handle, i); 6147 return status; 6148} 6149 6150size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6151{ 6152 Mutex::Autolock _l(mLock); 6153 size_t size = mHandles.size(); 6154 size_t i; 6155 for (i = 0; i < size; i++) { 6156 if (mHandles[i] == handle) break; 6157 } 6158 if (i == size) { 6159 return size; 6160 } 6161 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6162 6163 bool enabled = false; 6164 EffectHandle *hdl = handle.unsafe_get(); 6165 if (hdl != NULL) { 6166 ALOGV("removeHandle() unsafe_get OK"); 6167 enabled = hdl->enabled(); 6168 } 6169 mHandles.removeAt(i); 6170 size = mHandles.size(); 6171 // if removed from first place, move effect control from this handle to next in line 6172 if (i == 0 && size != 0) { 6173 sp<EffectHandle> h = mHandles[0].promote(); 6174 if (h != 0) { 6175 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6176 } 6177 } 6178 6179 // Prevent calls to process() and other functions on effect interface from now on. 6180 // The effect engine will be released by the destructor when the last strong reference on 6181 // this object is released which can happen after next process is called. 6182 if (size == 0 && !mPinned) { 6183 mState = DESTROYED; 6184 } 6185 6186 return size; 6187} 6188 6189sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6190{ 6191 Mutex::Autolock _l(mLock); 6192 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6193} 6194 6195void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6196{ 6197 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6198 // keep a strong reference on this EffectModule to avoid calling the 6199 // destructor before we exit 6200 sp<EffectModule> keep(this); 6201 { 6202 sp<ThreadBase> thread = mThread.promote(); 6203 if (thread != 0) { 6204 thread->disconnectEffect(keep, handle, unpinIfLast); 6205 } 6206 } 6207} 6208 6209void AudioFlinger::EffectModule::updateState() { 6210 Mutex::Autolock _l(mLock); 6211 6212 switch (mState) { 6213 case RESTART: 6214 reset_l(); 6215 // FALL THROUGH 6216 6217 case STARTING: 6218 // clear auxiliary effect input buffer for next accumulation 6219 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6220 memset(mConfig.inputCfg.buffer.raw, 6221 0, 6222 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6223 } 6224 start_l(); 6225 mState = ACTIVE; 6226 break; 6227 case STOPPING: 6228 stop_l(); 6229 mDisableWaitCnt = mMaxDisableWaitCnt; 6230 mState = STOPPED; 6231 break; 6232 case STOPPED: 6233 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6234 // turn off sequence. 6235 if (--mDisableWaitCnt == 0) { 6236 reset_l(); 6237 mState = IDLE; 6238 } 6239 break; 6240 default: //IDLE , ACTIVE, DESTROYED 6241 break; 6242 } 6243} 6244 6245void AudioFlinger::EffectModule::process() 6246{ 6247 Mutex::Autolock _l(mLock); 6248 6249 if (mState == DESTROYED || mEffectInterface == NULL || 6250 mConfig.inputCfg.buffer.raw == NULL || 6251 mConfig.outputCfg.buffer.raw == NULL) { 6252 return; 6253 } 6254 6255 if (isProcessEnabled()) { 6256 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6257 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6258 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6259 mConfig.inputCfg.buffer.s32, 6260 mConfig.inputCfg.buffer.frameCount/2); 6261 } 6262 6263 // do the actual processing in the effect engine 6264 int ret = (*mEffectInterface)->process(mEffectInterface, 6265 &mConfig.inputCfg.buffer, 6266 &mConfig.outputCfg.buffer); 6267 6268 // force transition to IDLE state when engine is ready 6269 if (mState == STOPPED && ret == -ENODATA) { 6270 mDisableWaitCnt = 1; 6271 } 6272 6273 // clear auxiliary effect input buffer for next accumulation 6274 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6275 memset(mConfig.inputCfg.buffer.raw, 0, 6276 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6277 } 6278 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6279 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6280 // If an insert effect is idle and input buffer is different from output buffer, 6281 // accumulate input onto output 6282 sp<EffectChain> chain = mChain.promote(); 6283 if (chain != 0 && chain->activeTrackCnt() != 0) { 6284 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6285 int16_t *in = mConfig.inputCfg.buffer.s16; 6286 int16_t *out = mConfig.outputCfg.buffer.s16; 6287 for (size_t i = 0; i < frameCnt; i++) { 6288 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6289 } 6290 } 6291 } 6292} 6293 6294void AudioFlinger::EffectModule::reset_l() 6295{ 6296 if (mEffectInterface == NULL) { 6297 return; 6298 } 6299 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6300} 6301 6302status_t AudioFlinger::EffectModule::configure() 6303{ 6304 uint32_t channels; 6305 if (mEffectInterface == NULL) { 6306 return NO_INIT; 6307 } 6308 6309 sp<ThreadBase> thread = mThread.promote(); 6310 if (thread == 0) { 6311 return DEAD_OBJECT; 6312 } 6313 6314 // TODO: handle configuration of effects replacing track process 6315 if (thread->channelCount() == 1) { 6316 channels = AUDIO_CHANNEL_OUT_MONO; 6317 } else { 6318 channels = AUDIO_CHANNEL_OUT_STEREO; 6319 } 6320 6321 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6322 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6323 } else { 6324 mConfig.inputCfg.channels = channels; 6325 } 6326 mConfig.outputCfg.channels = channels; 6327 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6328 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6329 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6330 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6331 mConfig.inputCfg.bufferProvider.cookie = NULL; 6332 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6333 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6334 mConfig.outputCfg.bufferProvider.cookie = NULL; 6335 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6336 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6337 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6338 // Insert effect: 6339 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6340 // always overwrites output buffer: input buffer == output buffer 6341 // - in other sessions: 6342 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6343 // other effect: overwrites output buffer: input buffer == output buffer 6344 // Auxiliary effect: 6345 // accumulates in output buffer: input buffer != output buffer 6346 // Therefore: accumulate <=> input buffer != output buffer 6347 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6348 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6349 } else { 6350 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6351 } 6352 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6353 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6354 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6355 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6356 6357 ALOGV("configure() %p thread %p buffer %p framecount %d", 6358 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6359 6360 status_t cmdStatus; 6361 uint32_t size = sizeof(int); 6362 status_t status = (*mEffectInterface)->command(mEffectInterface, 6363 EFFECT_CMD_SET_CONFIG, 6364 sizeof(effect_config_t), 6365 &mConfig, 6366 &size, 6367 &cmdStatus); 6368 if (status == 0) { 6369 status = cmdStatus; 6370 } 6371 6372 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6373 (1000 * mConfig.outputCfg.buffer.frameCount); 6374 6375 return status; 6376} 6377 6378status_t AudioFlinger::EffectModule::init() 6379{ 6380 Mutex::Autolock _l(mLock); 6381 if (mEffectInterface == NULL) { 6382 return NO_INIT; 6383 } 6384 status_t cmdStatus; 6385 uint32_t size = sizeof(status_t); 6386 status_t status = (*mEffectInterface)->command(mEffectInterface, 6387 EFFECT_CMD_INIT, 6388 0, 6389 NULL, 6390 &size, 6391 &cmdStatus); 6392 if (status == 0) { 6393 status = cmdStatus; 6394 } 6395 return status; 6396} 6397 6398status_t AudioFlinger::EffectModule::start() 6399{ 6400 Mutex::Autolock _l(mLock); 6401 return start_l(); 6402} 6403 6404status_t AudioFlinger::EffectModule::start_l() 6405{ 6406 if (mEffectInterface == NULL) { 6407 return NO_INIT; 6408 } 6409 status_t cmdStatus; 6410 uint32_t size = sizeof(status_t); 6411 status_t status = (*mEffectInterface)->command(mEffectInterface, 6412 EFFECT_CMD_ENABLE, 6413 0, 6414 NULL, 6415 &size, 6416 &cmdStatus); 6417 if (status == 0) { 6418 status = cmdStatus; 6419 } 6420 if (status == 0 && 6421 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6422 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6423 sp<ThreadBase> thread = mThread.promote(); 6424 if (thread != 0) { 6425 audio_stream_t *stream = thread->stream(); 6426 if (stream != NULL) { 6427 stream->add_audio_effect(stream, mEffectInterface); 6428 } 6429 } 6430 } 6431 return status; 6432} 6433 6434status_t AudioFlinger::EffectModule::stop() 6435{ 6436 Mutex::Autolock _l(mLock); 6437 return stop_l(); 6438} 6439 6440status_t AudioFlinger::EffectModule::stop_l() 6441{ 6442 if (mEffectInterface == NULL) { 6443 return NO_INIT; 6444 } 6445 status_t cmdStatus; 6446 uint32_t size = sizeof(status_t); 6447 status_t status = (*mEffectInterface)->command(mEffectInterface, 6448 EFFECT_CMD_DISABLE, 6449 0, 6450 NULL, 6451 &size, 6452 &cmdStatus); 6453 if (status == 0) { 6454 status = cmdStatus; 6455 } 6456 if (status == 0 && 6457 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6458 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6459 sp<ThreadBase> thread = mThread.promote(); 6460 if (thread != 0) { 6461 audio_stream_t *stream = thread->stream(); 6462 if (stream != NULL) { 6463 stream->remove_audio_effect(stream, mEffectInterface); 6464 } 6465 } 6466 } 6467 return status; 6468} 6469 6470status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6471 uint32_t cmdSize, 6472 void *pCmdData, 6473 uint32_t *replySize, 6474 void *pReplyData) 6475{ 6476 Mutex::Autolock _l(mLock); 6477// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6478 6479 if (mState == DESTROYED || mEffectInterface == NULL) { 6480 return NO_INIT; 6481 } 6482 status_t status = (*mEffectInterface)->command(mEffectInterface, 6483 cmdCode, 6484 cmdSize, 6485 pCmdData, 6486 replySize, 6487 pReplyData); 6488 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6489 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6490 for (size_t i = 1; i < mHandles.size(); i++) { 6491 sp<EffectHandle> h = mHandles[i].promote(); 6492 if (h != 0) { 6493 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6494 } 6495 } 6496 } 6497 return status; 6498} 6499 6500status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6501{ 6502 6503 Mutex::Autolock _l(mLock); 6504 ALOGV("setEnabled %p enabled %d", this, enabled); 6505 6506 if (enabled != isEnabled()) { 6507 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6508 if (enabled && status != NO_ERROR) { 6509 return status; 6510 } 6511 6512 switch (mState) { 6513 // going from disabled to enabled 6514 case IDLE: 6515 mState = STARTING; 6516 break; 6517 case STOPPED: 6518 mState = RESTART; 6519 break; 6520 case STOPPING: 6521 mState = ACTIVE; 6522 break; 6523 6524 // going from enabled to disabled 6525 case RESTART: 6526 mState = STOPPED; 6527 break; 6528 case STARTING: 6529 mState = IDLE; 6530 break; 6531 case ACTIVE: 6532 mState = STOPPING; 6533 break; 6534 case DESTROYED: 6535 return NO_ERROR; // simply ignore as we are being destroyed 6536 } 6537 for (size_t i = 1; i < mHandles.size(); i++) { 6538 sp<EffectHandle> h = mHandles[i].promote(); 6539 if (h != 0) { 6540 h->setEnabled(enabled); 6541 } 6542 } 6543 } 6544 return NO_ERROR; 6545} 6546 6547bool AudioFlinger::EffectModule::isEnabled() const 6548{ 6549 switch (mState) { 6550 case RESTART: 6551 case STARTING: 6552 case ACTIVE: 6553 return true; 6554 case IDLE: 6555 case STOPPING: 6556 case STOPPED: 6557 case DESTROYED: 6558 default: 6559 return false; 6560 } 6561} 6562 6563bool AudioFlinger::EffectModule::isProcessEnabled() const 6564{ 6565 switch (mState) { 6566 case RESTART: 6567 case ACTIVE: 6568 case STOPPING: 6569 case STOPPED: 6570 return true; 6571 case IDLE: 6572 case STARTING: 6573 case DESTROYED: 6574 default: 6575 return false; 6576 } 6577} 6578 6579status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6580{ 6581 Mutex::Autolock _l(mLock); 6582 status_t status = NO_ERROR; 6583 6584 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6585 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6586 if (isProcessEnabled() && 6587 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6588 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6589 status_t cmdStatus; 6590 uint32_t volume[2]; 6591 uint32_t *pVolume = NULL; 6592 uint32_t size = sizeof(volume); 6593 volume[0] = *left; 6594 volume[1] = *right; 6595 if (controller) { 6596 pVolume = volume; 6597 } 6598 status = (*mEffectInterface)->command(mEffectInterface, 6599 EFFECT_CMD_SET_VOLUME, 6600 size, 6601 volume, 6602 &size, 6603 pVolume); 6604 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6605 *left = volume[0]; 6606 *right = volume[1]; 6607 } 6608 } 6609 return status; 6610} 6611 6612status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6613{ 6614 Mutex::Autolock _l(mLock); 6615 status_t status = NO_ERROR; 6616 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6617 // audio pre processing modules on RecordThread can receive both output and 6618 // input device indication in the same call 6619 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6620 if (dev) { 6621 status_t cmdStatus; 6622 uint32_t size = sizeof(status_t); 6623 6624 status = (*mEffectInterface)->command(mEffectInterface, 6625 EFFECT_CMD_SET_DEVICE, 6626 sizeof(uint32_t), 6627 &dev, 6628 &size, 6629 &cmdStatus); 6630 if (status == NO_ERROR) { 6631 status = cmdStatus; 6632 } 6633 } 6634 dev = device & AUDIO_DEVICE_IN_ALL; 6635 if (dev) { 6636 status_t cmdStatus; 6637 uint32_t size = sizeof(status_t); 6638 6639 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6640 EFFECT_CMD_SET_INPUT_DEVICE, 6641 sizeof(uint32_t), 6642 &dev, 6643 &size, 6644 &cmdStatus); 6645 if (status2 == NO_ERROR) { 6646 status2 = cmdStatus; 6647 } 6648 if (status == NO_ERROR) { 6649 status = status2; 6650 } 6651 } 6652 } 6653 return status; 6654} 6655 6656status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6657{ 6658 Mutex::Autolock _l(mLock); 6659 status_t status = NO_ERROR; 6660 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6661 status_t cmdStatus; 6662 uint32_t size = sizeof(status_t); 6663 status = (*mEffectInterface)->command(mEffectInterface, 6664 EFFECT_CMD_SET_AUDIO_MODE, 6665 sizeof(audio_mode_t), 6666 &mode, 6667 &size, 6668 &cmdStatus); 6669 if (status == NO_ERROR) { 6670 status = cmdStatus; 6671 } 6672 } 6673 return status; 6674} 6675 6676void AudioFlinger::EffectModule::setSuspended(bool suspended) 6677{ 6678 Mutex::Autolock _l(mLock); 6679 mSuspended = suspended; 6680} 6681 6682bool AudioFlinger::EffectModule::suspended() const 6683{ 6684 Mutex::Autolock _l(mLock); 6685 return mSuspended; 6686} 6687 6688status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6689{ 6690 const size_t SIZE = 256; 6691 char buffer[SIZE]; 6692 String8 result; 6693 6694 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6695 result.append(buffer); 6696 6697 bool locked = tryLock(mLock); 6698 // failed to lock - AudioFlinger is probably deadlocked 6699 if (!locked) { 6700 result.append("\t\tCould not lock Fx mutex:\n"); 6701 } 6702 6703 result.append("\t\tSession Status State Engine:\n"); 6704 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6705 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6706 result.append(buffer); 6707 6708 result.append("\t\tDescriptor:\n"); 6709 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6710 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6711 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6712 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6713 result.append(buffer); 6714 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6715 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6716 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6717 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6718 result.append(buffer); 6719 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6720 mDescriptor.apiVersion, 6721 mDescriptor.flags); 6722 result.append(buffer); 6723 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6724 mDescriptor.name); 6725 result.append(buffer); 6726 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6727 mDescriptor.implementor); 6728 result.append(buffer); 6729 6730 result.append("\t\t- Input configuration:\n"); 6731 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6732 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6733 (uint32_t)mConfig.inputCfg.buffer.raw, 6734 mConfig.inputCfg.buffer.frameCount, 6735 mConfig.inputCfg.samplingRate, 6736 mConfig.inputCfg.channels, 6737 mConfig.inputCfg.format); 6738 result.append(buffer); 6739 6740 result.append("\t\t- Output configuration:\n"); 6741 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6742 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6743 (uint32_t)mConfig.outputCfg.buffer.raw, 6744 mConfig.outputCfg.buffer.frameCount, 6745 mConfig.outputCfg.samplingRate, 6746 mConfig.outputCfg.channels, 6747 mConfig.outputCfg.format); 6748 result.append(buffer); 6749 6750 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6751 result.append(buffer); 6752 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6753 for (size_t i = 0; i < mHandles.size(); ++i) { 6754 sp<EffectHandle> handle = mHandles[i].promote(); 6755 if (handle != 0) { 6756 handle->dump(buffer, SIZE); 6757 result.append(buffer); 6758 } 6759 } 6760 6761 result.append("\n"); 6762 6763 write(fd, result.string(), result.length()); 6764 6765 if (locked) { 6766 mLock.unlock(); 6767 } 6768 6769 return NO_ERROR; 6770} 6771 6772// ---------------------------------------------------------------------------- 6773// EffectHandle implementation 6774// ---------------------------------------------------------------------------- 6775 6776#undef LOG_TAG 6777#define LOG_TAG "AudioFlinger::EffectHandle" 6778 6779AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6780 const sp<AudioFlinger::Client>& client, 6781 const sp<IEffectClient>& effectClient, 6782 int32_t priority) 6783 : BnEffect(), 6784 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6785 mPriority(priority), mHasControl(false), mEnabled(false) 6786{ 6787 ALOGV("constructor %p", this); 6788 6789 if (client == 0) { 6790 return; 6791 } 6792 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6793 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6794 if (mCblkMemory != 0) { 6795 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6796 6797 if (mCblk != NULL) { 6798 new(mCblk) effect_param_cblk_t(); 6799 mBuffer = (uint8_t *)mCblk + bufOffset; 6800 } 6801 } else { 6802 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6803 return; 6804 } 6805} 6806 6807AudioFlinger::EffectHandle::~EffectHandle() 6808{ 6809 ALOGV("Destructor %p", this); 6810 disconnect(false); 6811 ALOGV("Destructor DONE %p", this); 6812} 6813 6814status_t AudioFlinger::EffectHandle::enable() 6815{ 6816 ALOGV("enable %p", this); 6817 if (!mHasControl) return INVALID_OPERATION; 6818 if (mEffect == 0) return DEAD_OBJECT; 6819 6820 if (mEnabled) { 6821 return NO_ERROR; 6822 } 6823 6824 mEnabled = true; 6825 6826 sp<ThreadBase> thread = mEffect->thread().promote(); 6827 if (thread != 0) { 6828 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6829 } 6830 6831 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6832 if (mEffect->suspended()) { 6833 return NO_ERROR; 6834 } 6835 6836 status_t status = mEffect->setEnabled(true); 6837 if (status != NO_ERROR) { 6838 if (thread != 0) { 6839 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6840 } 6841 mEnabled = false; 6842 } 6843 return status; 6844} 6845 6846status_t AudioFlinger::EffectHandle::disable() 6847{ 6848 ALOGV("disable %p", this); 6849 if (!mHasControl) return INVALID_OPERATION; 6850 if (mEffect == 0) return DEAD_OBJECT; 6851 6852 if (!mEnabled) { 6853 return NO_ERROR; 6854 } 6855 mEnabled = false; 6856 6857 if (mEffect->suspended()) { 6858 return NO_ERROR; 6859 } 6860 6861 status_t status = mEffect->setEnabled(false); 6862 6863 sp<ThreadBase> thread = mEffect->thread().promote(); 6864 if (thread != 0) { 6865 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6866 } 6867 6868 return status; 6869} 6870 6871void AudioFlinger::EffectHandle::disconnect() 6872{ 6873 disconnect(true); 6874} 6875 6876void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 6877{ 6878 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 6879 if (mEffect == 0) { 6880 return; 6881 } 6882 mEffect->disconnect(this, unpinIfLast); 6883 6884 if (mHasControl && mEnabled) { 6885 sp<ThreadBase> thread = mEffect->thread().promote(); 6886 if (thread != 0) { 6887 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6888 } 6889 } 6890 6891 // release sp on module => module destructor can be called now 6892 mEffect.clear(); 6893 if (mClient != 0) { 6894 if (mCblk != NULL) { 6895 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 6896 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6897 } 6898 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 6899 // Client destructor must run with AudioFlinger mutex locked 6900 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6901 mClient.clear(); 6902 } 6903} 6904 6905status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6906 uint32_t cmdSize, 6907 void *pCmdData, 6908 uint32_t *replySize, 6909 void *pReplyData) 6910{ 6911// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6912// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6913 6914 // only get parameter command is permitted for applications not controlling the effect 6915 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6916 return INVALID_OPERATION; 6917 } 6918 if (mEffect == 0) return DEAD_OBJECT; 6919 if (mClient == 0) return INVALID_OPERATION; 6920 6921 // handle commands that are not forwarded transparently to effect engine 6922 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6923 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6924 // no risk to block the whole media server process or mixer threads is we are stuck here 6925 Mutex::Autolock _l(mCblk->lock); 6926 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6927 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6928 mCblk->serverIndex = 0; 6929 mCblk->clientIndex = 0; 6930 return BAD_VALUE; 6931 } 6932 status_t status = NO_ERROR; 6933 while (mCblk->serverIndex < mCblk->clientIndex) { 6934 int reply; 6935 uint32_t rsize = sizeof(int); 6936 int *p = (int *)(mBuffer + mCblk->serverIndex); 6937 int size = *p++; 6938 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6939 ALOGW("command(): invalid parameter block size"); 6940 break; 6941 } 6942 effect_param_t *param = (effect_param_t *)p; 6943 if (param->psize == 0 || param->vsize == 0) { 6944 ALOGW("command(): null parameter or value size"); 6945 mCblk->serverIndex += size; 6946 continue; 6947 } 6948 uint32_t psize = sizeof(effect_param_t) + 6949 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6950 param->vsize; 6951 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6952 psize, 6953 p, 6954 &rsize, 6955 &reply); 6956 // stop at first error encountered 6957 if (ret != NO_ERROR) { 6958 status = ret; 6959 *(int *)pReplyData = reply; 6960 break; 6961 } else if (reply != NO_ERROR) { 6962 *(int *)pReplyData = reply; 6963 break; 6964 } 6965 mCblk->serverIndex += size; 6966 } 6967 mCblk->serverIndex = 0; 6968 mCblk->clientIndex = 0; 6969 return status; 6970 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6971 *(int *)pReplyData = NO_ERROR; 6972 return enable(); 6973 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6974 *(int *)pReplyData = NO_ERROR; 6975 return disable(); 6976 } 6977 6978 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6979} 6980 6981void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 6982{ 6983 ALOGV("setControl %p control %d", this, hasControl); 6984 6985 mHasControl = hasControl; 6986 mEnabled = enabled; 6987 6988 if (signal && mEffectClient != 0) { 6989 mEffectClient->controlStatusChanged(hasControl); 6990 } 6991} 6992 6993void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6994 uint32_t cmdSize, 6995 void *pCmdData, 6996 uint32_t replySize, 6997 void *pReplyData) 6998{ 6999 if (mEffectClient != 0) { 7000 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7001 } 7002} 7003 7004 7005 7006void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7007{ 7008 if (mEffectClient != 0) { 7009 mEffectClient->enableStatusChanged(enabled); 7010 } 7011} 7012 7013status_t AudioFlinger::EffectHandle::onTransact( 7014 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7015{ 7016 return BnEffect::onTransact(code, data, reply, flags); 7017} 7018 7019 7020void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7021{ 7022 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7023 7024 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7025 (mClient == 0) ? getpid() : mClient->pid(), 7026 mPriority, 7027 mHasControl, 7028 !locked, 7029 mCblk ? mCblk->clientIndex : 0, 7030 mCblk ? mCblk->serverIndex : 0 7031 ); 7032 7033 if (locked) { 7034 mCblk->lock.unlock(); 7035 } 7036} 7037 7038#undef LOG_TAG 7039#define LOG_TAG "AudioFlinger::EffectChain" 7040 7041AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7042 int sessionId) 7043 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7044 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7045 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7046{ 7047 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7048 sp<ThreadBase> thread = mThread.promote(); 7049 if (thread == 0) { 7050 return; 7051 } 7052 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7053 thread->frameCount(); 7054} 7055 7056AudioFlinger::EffectChain::~EffectChain() 7057{ 7058 if (mOwnInBuffer) { 7059 delete mInBuffer; 7060 } 7061 7062} 7063 7064// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7065sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7066{ 7067 size_t size = mEffects.size(); 7068 7069 for (size_t i = 0; i < size; i++) { 7070 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7071 return mEffects[i]; 7072 } 7073 } 7074 return 0; 7075} 7076 7077// getEffectFromId_l() must be called with ThreadBase::mLock held 7078sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7079{ 7080 size_t size = mEffects.size(); 7081 7082 for (size_t i = 0; i < size; i++) { 7083 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7084 if (id == 0 || mEffects[i]->id() == id) { 7085 return mEffects[i]; 7086 } 7087 } 7088 return 0; 7089} 7090 7091// getEffectFromType_l() must be called with ThreadBase::mLock held 7092sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7093 const effect_uuid_t *type) 7094{ 7095 size_t size = mEffects.size(); 7096 7097 for (size_t i = 0; i < size; i++) { 7098 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7099 return mEffects[i]; 7100 } 7101 } 7102 return 0; 7103} 7104 7105// Must be called with EffectChain::mLock locked 7106void AudioFlinger::EffectChain::process_l() 7107{ 7108 sp<ThreadBase> thread = mThread.promote(); 7109 if (thread == 0) { 7110 ALOGW("process_l(): cannot promote mixer thread"); 7111 return; 7112 } 7113 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7114 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7115 // always process effects unless no more tracks are on the session and the effect tail 7116 // has been rendered 7117 bool doProcess = true; 7118 if (!isGlobalSession) { 7119 bool tracksOnSession = (trackCnt() != 0); 7120 7121 if (!tracksOnSession && mTailBufferCount == 0) { 7122 doProcess = false; 7123 } 7124 7125 if (activeTrackCnt() == 0) { 7126 // if no track is active and the effect tail has not been rendered, 7127 // the input buffer must be cleared here as the mixer process will not do it 7128 if (tracksOnSession || mTailBufferCount > 0) { 7129 size_t numSamples = thread->frameCount() * thread->channelCount(); 7130 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7131 if (mTailBufferCount > 0) { 7132 mTailBufferCount--; 7133 } 7134 } 7135 } 7136 } 7137 7138 size_t size = mEffects.size(); 7139 if (doProcess) { 7140 for (size_t i = 0; i < size; i++) { 7141 mEffects[i]->process(); 7142 } 7143 } 7144 for (size_t i = 0; i < size; i++) { 7145 mEffects[i]->updateState(); 7146 } 7147} 7148 7149// addEffect_l() must be called with PlaybackThread::mLock held 7150status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7151{ 7152 effect_descriptor_t desc = effect->desc(); 7153 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7154 7155 Mutex::Autolock _l(mLock); 7156 effect->setChain(this); 7157 sp<ThreadBase> thread = mThread.promote(); 7158 if (thread == 0) { 7159 return NO_INIT; 7160 } 7161 effect->setThread(thread); 7162 7163 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7164 // Auxiliary effects are inserted at the beginning of mEffects vector as 7165 // they are processed first and accumulated in chain input buffer 7166 mEffects.insertAt(effect, 0); 7167 7168 // the input buffer for auxiliary effect contains mono samples in 7169 // 32 bit format. This is to avoid saturation in AudoMixer 7170 // accumulation stage. Saturation is done in EffectModule::process() before 7171 // calling the process in effect engine 7172 size_t numSamples = thread->frameCount(); 7173 int32_t *buffer = new int32_t[numSamples]; 7174 memset(buffer, 0, numSamples * sizeof(int32_t)); 7175 effect->setInBuffer((int16_t *)buffer); 7176 // auxiliary effects output samples to chain input buffer for further processing 7177 // by insert effects 7178 effect->setOutBuffer(mInBuffer); 7179 } else { 7180 // Insert effects are inserted at the end of mEffects vector as they are processed 7181 // after track and auxiliary effects. 7182 // Insert effect order as a function of indicated preference: 7183 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7184 // another effect is present 7185 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7186 // last effect claiming first position 7187 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7188 // first effect claiming last position 7189 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7190 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7191 // already present 7192 7193 int size = (int)mEffects.size(); 7194 int idx_insert = size; 7195 int idx_insert_first = -1; 7196 int idx_insert_last = -1; 7197 7198 for (int i = 0; i < size; i++) { 7199 effect_descriptor_t d = mEffects[i]->desc(); 7200 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7201 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7202 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7203 // check invalid effect chaining combinations 7204 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7205 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7206 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7207 return INVALID_OPERATION; 7208 } 7209 // remember position of first insert effect and by default 7210 // select this as insert position for new effect 7211 if (idx_insert == size) { 7212 idx_insert = i; 7213 } 7214 // remember position of last insert effect claiming 7215 // first position 7216 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7217 idx_insert_first = i; 7218 } 7219 // remember position of first insert effect claiming 7220 // last position 7221 if (iPref == EFFECT_FLAG_INSERT_LAST && 7222 idx_insert_last == -1) { 7223 idx_insert_last = i; 7224 } 7225 } 7226 } 7227 7228 // modify idx_insert from first position if needed 7229 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7230 if (idx_insert_last != -1) { 7231 idx_insert = idx_insert_last; 7232 } else { 7233 idx_insert = size; 7234 } 7235 } else { 7236 if (idx_insert_first != -1) { 7237 idx_insert = idx_insert_first + 1; 7238 } 7239 } 7240 7241 // always read samples from chain input buffer 7242 effect->setInBuffer(mInBuffer); 7243 7244 // if last effect in the chain, output samples to chain 7245 // output buffer, otherwise to chain input buffer 7246 if (idx_insert == size) { 7247 if (idx_insert != 0) { 7248 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7249 mEffects[idx_insert-1]->configure(); 7250 } 7251 effect->setOutBuffer(mOutBuffer); 7252 } else { 7253 effect->setOutBuffer(mInBuffer); 7254 } 7255 mEffects.insertAt(effect, idx_insert); 7256 7257 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7258 } 7259 effect->configure(); 7260 return NO_ERROR; 7261} 7262 7263// removeEffect_l() must be called with PlaybackThread::mLock held 7264size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7265{ 7266 Mutex::Autolock _l(mLock); 7267 int size = (int)mEffects.size(); 7268 int i; 7269 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7270 7271 for (i = 0; i < size; i++) { 7272 if (effect == mEffects[i]) { 7273 // calling stop here will remove pre-processing effect from the audio HAL. 7274 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7275 // the middle of a read from audio HAL 7276 if (mEffects[i]->state() == EffectModule::ACTIVE || 7277 mEffects[i]->state() == EffectModule::STOPPING) { 7278 mEffects[i]->stop(); 7279 } 7280 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7281 delete[] effect->inBuffer(); 7282 } else { 7283 if (i == size - 1 && i != 0) { 7284 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7285 mEffects[i - 1]->configure(); 7286 } 7287 } 7288 mEffects.removeAt(i); 7289 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7290 break; 7291 } 7292 } 7293 7294 return mEffects.size(); 7295} 7296 7297// setDevice_l() must be called with PlaybackThread::mLock held 7298void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7299{ 7300 size_t size = mEffects.size(); 7301 for (size_t i = 0; i < size; i++) { 7302 mEffects[i]->setDevice(device); 7303 } 7304} 7305 7306// setMode_l() must be called with PlaybackThread::mLock held 7307void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7308{ 7309 size_t size = mEffects.size(); 7310 for (size_t i = 0; i < size; i++) { 7311 mEffects[i]->setMode(mode); 7312 } 7313} 7314 7315// setVolume_l() must be called with PlaybackThread::mLock held 7316bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7317{ 7318 uint32_t newLeft = *left; 7319 uint32_t newRight = *right; 7320 bool hasControl = false; 7321 int ctrlIdx = -1; 7322 size_t size = mEffects.size(); 7323 7324 // first update volume controller 7325 for (size_t i = size; i > 0; i--) { 7326 if (mEffects[i - 1]->isProcessEnabled() && 7327 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7328 ctrlIdx = i - 1; 7329 hasControl = true; 7330 break; 7331 } 7332 } 7333 7334 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7335 if (hasControl) { 7336 *left = mNewLeftVolume; 7337 *right = mNewRightVolume; 7338 } 7339 return hasControl; 7340 } 7341 7342 mVolumeCtrlIdx = ctrlIdx; 7343 mLeftVolume = newLeft; 7344 mRightVolume = newRight; 7345 7346 // second get volume update from volume controller 7347 if (ctrlIdx >= 0) { 7348 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7349 mNewLeftVolume = newLeft; 7350 mNewRightVolume = newRight; 7351 } 7352 // then indicate volume to all other effects in chain. 7353 // Pass altered volume to effects before volume controller 7354 // and requested volume to effects after controller 7355 uint32_t lVol = newLeft; 7356 uint32_t rVol = newRight; 7357 7358 for (size_t i = 0; i < size; i++) { 7359 if ((int)i == ctrlIdx) continue; 7360 // this also works for ctrlIdx == -1 when there is no volume controller 7361 if ((int)i > ctrlIdx) { 7362 lVol = *left; 7363 rVol = *right; 7364 } 7365 mEffects[i]->setVolume(&lVol, &rVol, false); 7366 } 7367 *left = newLeft; 7368 *right = newRight; 7369 7370 return hasControl; 7371} 7372 7373status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7374{ 7375 const size_t SIZE = 256; 7376 char buffer[SIZE]; 7377 String8 result; 7378 7379 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7380 result.append(buffer); 7381 7382 bool locked = tryLock(mLock); 7383 // failed to lock - AudioFlinger is probably deadlocked 7384 if (!locked) { 7385 result.append("\tCould not lock mutex:\n"); 7386 } 7387 7388 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7389 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7390 mEffects.size(), 7391 (uint32_t)mInBuffer, 7392 (uint32_t)mOutBuffer, 7393 mActiveTrackCnt); 7394 result.append(buffer); 7395 write(fd, result.string(), result.size()); 7396 7397 for (size_t i = 0; i < mEffects.size(); ++i) { 7398 sp<EffectModule> effect = mEffects[i]; 7399 if (effect != 0) { 7400 effect->dump(fd, args); 7401 } 7402 } 7403 7404 if (locked) { 7405 mLock.unlock(); 7406 } 7407 7408 return NO_ERROR; 7409} 7410 7411// must be called with ThreadBase::mLock held 7412void AudioFlinger::EffectChain::setEffectSuspended_l( 7413 const effect_uuid_t *type, bool suspend) 7414{ 7415 sp<SuspendedEffectDesc> desc; 7416 // use effect type UUID timelow as key as there is no real risk of identical 7417 // timeLow fields among effect type UUIDs. 7418 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7419 if (suspend) { 7420 if (index >= 0) { 7421 desc = mSuspendedEffects.valueAt(index); 7422 } else { 7423 desc = new SuspendedEffectDesc(); 7424 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7425 mSuspendedEffects.add(type->timeLow, desc); 7426 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7427 } 7428 if (desc->mRefCount++ == 0) { 7429 sp<EffectModule> effect = getEffectIfEnabled(type); 7430 if (effect != 0) { 7431 desc->mEffect = effect; 7432 effect->setSuspended(true); 7433 effect->setEnabled(false); 7434 } 7435 } 7436 } else { 7437 if (index < 0) { 7438 return; 7439 } 7440 desc = mSuspendedEffects.valueAt(index); 7441 if (desc->mRefCount <= 0) { 7442 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7443 desc->mRefCount = 1; 7444 } 7445 if (--desc->mRefCount == 0) { 7446 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7447 if (desc->mEffect != 0) { 7448 sp<EffectModule> effect = desc->mEffect.promote(); 7449 if (effect != 0) { 7450 effect->setSuspended(false); 7451 sp<EffectHandle> handle = effect->controlHandle(); 7452 if (handle != 0) { 7453 effect->setEnabled(handle->enabled()); 7454 } 7455 } 7456 desc->mEffect.clear(); 7457 } 7458 mSuspendedEffects.removeItemsAt(index); 7459 } 7460 } 7461} 7462 7463// must be called with ThreadBase::mLock held 7464void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7465{ 7466 sp<SuspendedEffectDesc> desc; 7467 7468 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7469 if (suspend) { 7470 if (index >= 0) { 7471 desc = mSuspendedEffects.valueAt(index); 7472 } else { 7473 desc = new SuspendedEffectDesc(); 7474 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7475 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7476 } 7477 if (desc->mRefCount++ == 0) { 7478 Vector< sp<EffectModule> > effects; 7479 getSuspendEligibleEffects(effects); 7480 for (size_t i = 0; i < effects.size(); i++) { 7481 setEffectSuspended_l(&effects[i]->desc().type, true); 7482 } 7483 } 7484 } else { 7485 if (index < 0) { 7486 return; 7487 } 7488 desc = mSuspendedEffects.valueAt(index); 7489 if (desc->mRefCount <= 0) { 7490 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7491 desc->mRefCount = 1; 7492 } 7493 if (--desc->mRefCount == 0) { 7494 Vector<const effect_uuid_t *> types; 7495 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7496 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7497 continue; 7498 } 7499 types.add(&mSuspendedEffects.valueAt(i)->mType); 7500 } 7501 for (size_t i = 0; i < types.size(); i++) { 7502 setEffectSuspended_l(types[i], false); 7503 } 7504 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7505 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7506 } 7507 } 7508} 7509 7510 7511// The volume effect is used for automated tests only 7512#ifndef OPENSL_ES_H_ 7513static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7514 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7515const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7516#endif //OPENSL_ES_H_ 7517 7518bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7519{ 7520 // auxiliary effects and visualizer are never suspended on output mix 7521 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7522 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7523 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7524 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7525 return false; 7526 } 7527 return true; 7528} 7529 7530void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 7531{ 7532 effects.clear(); 7533 for (size_t i = 0; i < mEffects.size(); i++) { 7534 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 7535 effects.add(mEffects[i]); 7536 } 7537 } 7538} 7539 7540sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7541 const effect_uuid_t *type) 7542{ 7543 sp<EffectModule> effect = getEffectFromType_l(type); 7544 return effect != 0 && effect->isEnabled() ? effect : 0; 7545} 7546 7547void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7548 bool enabled) 7549{ 7550 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7551 if (enabled) { 7552 if (index < 0) { 7553 // if the effect is not suspend check if all effects are suspended 7554 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7555 if (index < 0) { 7556 return; 7557 } 7558 if (!isEffectEligibleForSuspend(effect->desc())) { 7559 return; 7560 } 7561 setEffectSuspended_l(&effect->desc().type, enabled); 7562 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7563 if (index < 0) { 7564 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7565 return; 7566 } 7567 } 7568 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7569 effect->desc().type.timeLow); 7570 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7571 // if effect is requested to suspended but was not yet enabled, supend it now. 7572 if (desc->mEffect == 0) { 7573 desc->mEffect = effect; 7574 effect->setEnabled(false); 7575 effect->setSuspended(true); 7576 } 7577 } else { 7578 if (index < 0) { 7579 return; 7580 } 7581 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7582 effect->desc().type.timeLow); 7583 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7584 desc->mEffect.clear(); 7585 effect->setSuspended(false); 7586 } 7587} 7588 7589#undef LOG_TAG 7590#define LOG_TAG "AudioFlinger" 7591 7592// ---------------------------------------------------------------------------- 7593 7594status_t AudioFlinger::onTransact( 7595 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7596{ 7597 return BnAudioFlinger::onTransact(code, data, reply, flags); 7598} 7599 7600}; // namespace android 7601