AudioFlinger.cpp revision 05632a5fa4b88ca474294887fc92a9fcdf0e2352
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/AudioTrack.h> 41#include <media/AudioRecord.h> 42#include <media/IMediaPlayerService.h> 43#include <media/IMediaDeathNotifier.h> 44 45#include <private/media/AudioTrackShared.h> 46#include <private/media/AudioEffectShared.h> 47 48#include <system/audio.h> 49#include <hardware/audio.h> 50 51#include "AudioMixer.h" 52#include "AudioFlinger.h" 53 54#include <media/EffectsFactoryApi.h> 55#include <audio_effects/effect_visualizer.h> 56#include <audio_effects/effect_ns.h> 57#include <audio_effects/effect_aec.h> 58 59#include <audio_utils/primitives.h> 60 61#include <cpustats/ThreadCpuUsage.h> 62#include <powermanager/PowerManager.h> 63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 64 65// ---------------------------------------------------------------------------- 66 67 68namespace android { 69 70static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 71static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 72 73//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 74static const float MAX_GAIN = 4096.0f; 75static const float MAX_GAIN_INT = 0x1000; 76 77// retry counts for buffer fill timeout 78// 50 * ~20msecs = 1 second 79static const int8_t kMaxTrackRetries = 50; 80static const int8_t kMaxTrackStartupRetries = 50; 81// allow less retry attempts on direct output thread. 82// direct outputs can be a scarce resource in audio hardware and should 83// be released as quickly as possible. 84static const int8_t kMaxTrackRetriesDirect = 2; 85 86static const int kDumpLockRetries = 50; 87static const int kDumpLockSleepUs = 20000; 88 89// don't warn about blocked writes or record buffer overflows more often than this 90static const nsecs_t kWarningThrottleNs = seconds(5); 91 92// RecordThread loop sleep time upon application overrun or audio HAL read error 93static const int kRecordThreadSleepUs = 5000; 94 95// maximum time to wait for setParameters to complete 96static const nsecs_t kSetParametersTimeoutNs = seconds(2); 97 98// minimum sleep time for the mixer thread loop when tracks are active but in underrun 99static const uint32_t kMinThreadSleepTimeUs = 5000; 100// maximum divider applied to the active sleep time in the mixer thread loop 101static const uint32_t kMaxThreadSleepTimeShift = 2; 102 103 104// ---------------------------------------------------------------------------- 105 106static bool recordingAllowed() { 107 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 108 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 109 if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO"); 110 return ok; 111} 112 113static bool settingsAllowed() { 114 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 115 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 116 if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 117 return ok; 118} 119 120// To collect the amplifier usage 121static void addBatteryData(uint32_t params) { 122 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 123 if (service == NULL) { 124 // it already logged 125 return; 126 } 127 128 service->addBatteryData(params); 129} 130 131static int load_audio_interface(const char *if_name, const hw_module_t **mod, 132 audio_hw_device_t **dev) 133{ 134 int rc; 135 136 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 137 if (rc) 138 goto out; 139 140 rc = audio_hw_device_open(*mod, dev); 141 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 142 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 143 if (rc) 144 goto out; 145 146 return 0; 147 148out: 149 *mod = NULL; 150 *dev = NULL; 151 return rc; 152} 153 154static const char * const audio_interfaces[] = { 155 "primary", 156 "a2dp", 157 "usb", 158}; 159#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 160 161// ---------------------------------------------------------------------------- 162 163AudioFlinger::AudioFlinger() 164 : BnAudioFlinger(), 165 mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 166 mBtNrecIsOff(false) 167{ 168} 169 170void AudioFlinger::onFirstRef() 171{ 172 int rc = 0; 173 174 Mutex::Autolock _l(mLock); 175 176 /* TODO: move all this work into an Init() function */ 177 mHardwareStatus = AUDIO_HW_IDLE; 178 179 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 180 const hw_module_t *mod; 181 audio_hw_device_t *dev; 182 183 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 184 if (rc) 185 continue; 186 187 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 188 mod->name, mod->id); 189 mAudioHwDevs.push(dev); 190 191 if (!mPrimaryHardwareDev) { 192 mPrimaryHardwareDev = dev; 193 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 194 mod->name, mod->id, audio_interfaces[i]); 195 } 196 } 197 198 mHardwareStatus = AUDIO_HW_INIT; 199 200 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 201 ALOGE("Primary audio interface not found"); 202 return; 203 } 204 205 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 206 audio_hw_device_t *dev = mAudioHwDevs[i]; 207 208 mHardwareStatus = AUDIO_HW_INIT; 209 rc = dev->init_check(dev); 210 if (rc == 0) { 211 AutoMutex lock(mHardwareLock); 212 213 mMode = AUDIO_MODE_NORMAL; 214 mHardwareStatus = AUDIO_HW_SET_MODE; 215 dev->set_mode(dev, mMode); 216 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 217 dev->set_master_volume(dev, 1.0f); 218 mHardwareStatus = AUDIO_HW_IDLE; 219 } 220 } 221} 222 223status_t AudioFlinger::initCheck() const 224{ 225 Mutex::Autolock _l(mLock); 226 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 227 return NO_INIT; 228 return NO_ERROR; 229} 230 231AudioFlinger::~AudioFlinger() 232{ 233 int num_devs = mAudioHwDevs.size(); 234 235 while (!mRecordThreads.isEmpty()) { 236 // closeInput() will remove first entry from mRecordThreads 237 closeInput(mRecordThreads.keyAt(0)); 238 } 239 while (!mPlaybackThreads.isEmpty()) { 240 // closeOutput() will remove first entry from mPlaybackThreads 241 closeOutput(mPlaybackThreads.keyAt(0)); 242 } 243 244 for (int i = 0; i < num_devs; i++) { 245 audio_hw_device_t *dev = mAudioHwDevs[i]; 246 audio_hw_device_close(dev); 247 } 248 mAudioHwDevs.clear(); 249} 250 251audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 252{ 253 /* first matching HW device is returned */ 254 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 255 audio_hw_device_t *dev = mAudioHwDevs[i]; 256 if ((dev->get_supported_devices(dev) & devices) == devices) 257 return dev; 258 } 259 return NULL; 260} 261 262status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 263{ 264 const size_t SIZE = 256; 265 char buffer[SIZE]; 266 String8 result; 267 268 result.append("Clients:\n"); 269 for (size_t i = 0; i < mClients.size(); ++i) { 270 wp<Client> wClient = mClients.valueAt(i); 271 if (wClient != 0) { 272 sp<Client> client = wClient.promote(); 273 if (client != 0) { 274 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 275 result.append(buffer); 276 } 277 } 278 } 279 280 result.append("Global session refs:\n"); 281 result.append(" session pid cnt\n"); 282 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 283 AudioSessionRef *r = mAudioSessionRefs[i]; 284 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 285 result.append(buffer); 286 } 287 write(fd, result.string(), result.size()); 288 return NO_ERROR; 289} 290 291 292status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 293{ 294 const size_t SIZE = 256; 295 char buffer[SIZE]; 296 String8 result; 297 hardware_call_state hardwareStatus = mHardwareStatus; 298 299 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 300 result.append(buffer); 301 write(fd, result.string(), result.size()); 302 return NO_ERROR; 303} 304 305status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 306{ 307 const size_t SIZE = 256; 308 char buffer[SIZE]; 309 String8 result; 310 snprintf(buffer, SIZE, "Permission Denial: " 311 "can't dump AudioFlinger from pid=%d, uid=%d\n", 312 IPCThreadState::self()->getCallingPid(), 313 IPCThreadState::self()->getCallingUid()); 314 result.append(buffer); 315 write(fd, result.string(), result.size()); 316 return NO_ERROR; 317} 318 319static bool tryLock(Mutex& mutex) 320{ 321 bool locked = false; 322 for (int i = 0; i < kDumpLockRetries; ++i) { 323 if (mutex.tryLock() == NO_ERROR) { 324 locked = true; 325 break; 326 } 327 usleep(kDumpLockSleepUs); 328 } 329 return locked; 330} 331 332status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 333{ 334 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 335 dumpPermissionDenial(fd, args); 336 } else { 337 // get state of hardware lock 338 bool hardwareLocked = tryLock(mHardwareLock); 339 if (!hardwareLocked) { 340 String8 result(kHardwareLockedString); 341 write(fd, result.string(), result.size()); 342 } else { 343 mHardwareLock.unlock(); 344 } 345 346 bool locked = tryLock(mLock); 347 348 // failed to lock - AudioFlinger is probably deadlocked 349 if (!locked) { 350 String8 result(kDeadlockedString); 351 write(fd, result.string(), result.size()); 352 } 353 354 dumpClients(fd, args); 355 dumpInternals(fd, args); 356 357 // dump playback threads 358 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 359 mPlaybackThreads.valueAt(i)->dump(fd, args); 360 } 361 362 // dump record threads 363 for (size_t i = 0; i < mRecordThreads.size(); i++) { 364 mRecordThreads.valueAt(i)->dump(fd, args); 365 } 366 367 // dump all hardware devs 368 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 369 audio_hw_device_t *dev = mAudioHwDevs[i]; 370 dev->dump(dev, fd); 371 } 372 if (locked) mLock.unlock(); 373 } 374 return NO_ERROR; 375} 376 377 378// IAudioFlinger interface 379 380 381sp<IAudioTrack> AudioFlinger::createTrack( 382 pid_t pid, 383 int streamType, 384 uint32_t sampleRate, 385 uint32_t format, 386 uint32_t channelMask, 387 int frameCount, 388 uint32_t flags, 389 const sp<IMemory>& sharedBuffer, 390 int output, 391 int *sessionId, 392 status_t *status) 393{ 394 sp<PlaybackThread::Track> track; 395 sp<TrackHandle> trackHandle; 396 sp<Client> client; 397 wp<Client> wclient; 398 status_t lStatus; 399 int lSessionId; 400 401 if (streamType >= AUDIO_STREAM_CNT) { 402 ALOGE("createTrack() invalid stream type %d", streamType); 403 lStatus = BAD_VALUE; 404 goto Exit; 405 } 406 407 { 408 Mutex::Autolock _l(mLock); 409 PlaybackThread *thread = checkPlaybackThread_l(output); 410 PlaybackThread *effectThread = NULL; 411 if (thread == NULL) { 412 ALOGE("unknown output thread"); 413 lStatus = BAD_VALUE; 414 goto Exit; 415 } 416 417 wclient = mClients.valueFor(pid); 418 419 if (wclient != NULL) { 420 client = wclient.promote(); 421 } else { 422 client = new Client(this, pid); 423 mClients.add(pid, client); 424 } 425 426 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 427 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 428 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 429 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 430 if (mPlaybackThreads.keyAt(i) != output) { 431 // prevent same audio session on different output threads 432 uint32_t sessions = t->hasAudioSession(*sessionId); 433 if (sessions & PlaybackThread::TRACK_SESSION) { 434 ALOGE("createTrack() session ID %d already in use", *sessionId); 435 lStatus = BAD_VALUE; 436 goto Exit; 437 } 438 // check if an effect with same session ID is waiting for a track to be created 439 if (sessions & PlaybackThread::EFFECT_SESSION) { 440 effectThread = t.get(); 441 } 442 } 443 } 444 lSessionId = *sessionId; 445 } else { 446 // if no audio session id is provided, create one here 447 lSessionId = nextUniqueId(); 448 if (sessionId != NULL) { 449 *sessionId = lSessionId; 450 } 451 } 452 ALOGV("createTrack() lSessionId: %d", lSessionId); 453 454 track = thread->createTrack_l(client, streamType, sampleRate, format, 455 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 456 457 // move effect chain to this output thread if an effect on same session was waiting 458 // for a track to be created 459 if (lStatus == NO_ERROR && effectThread != NULL) { 460 Mutex::Autolock _dl(thread->mLock); 461 Mutex::Autolock _sl(effectThread->mLock); 462 moveEffectChain_l(lSessionId, effectThread, thread, true); 463 } 464 } 465 if (lStatus == NO_ERROR) { 466 trackHandle = new TrackHandle(track); 467 } else { 468 // remove local strong reference to Client before deleting the Track so that the Client 469 // destructor is called by the TrackBase destructor with mLock held 470 client.clear(); 471 track.clear(); 472 } 473 474Exit: 475 if(status) { 476 *status = lStatus; 477 } 478 return trackHandle; 479} 480 481uint32_t AudioFlinger::sampleRate(int output) const 482{ 483 Mutex::Autolock _l(mLock); 484 PlaybackThread *thread = checkPlaybackThread_l(output); 485 if (thread == NULL) { 486 ALOGW("sampleRate() unknown thread %d", output); 487 return 0; 488 } 489 return thread->sampleRate(); 490} 491 492int AudioFlinger::channelCount(int output) const 493{ 494 Mutex::Autolock _l(mLock); 495 PlaybackThread *thread = checkPlaybackThread_l(output); 496 if (thread == NULL) { 497 ALOGW("channelCount() unknown thread %d", output); 498 return 0; 499 } 500 return thread->channelCount(); 501} 502 503uint32_t AudioFlinger::format(int output) const 504{ 505 Mutex::Autolock _l(mLock); 506 PlaybackThread *thread = checkPlaybackThread_l(output); 507 if (thread == NULL) { 508 ALOGW("format() unknown thread %d", output); 509 return 0; 510 } 511 return thread->format(); 512} 513 514size_t AudioFlinger::frameCount(int output) const 515{ 516 Mutex::Autolock _l(mLock); 517 PlaybackThread *thread = checkPlaybackThread_l(output); 518 if (thread == NULL) { 519 ALOGW("frameCount() unknown thread %d", output); 520 return 0; 521 } 522 return thread->frameCount(); 523} 524 525uint32_t AudioFlinger::latency(int output) const 526{ 527 Mutex::Autolock _l(mLock); 528 PlaybackThread *thread = checkPlaybackThread_l(output); 529 if (thread == NULL) { 530 ALOGW("latency() unknown thread %d", output); 531 return 0; 532 } 533 return thread->latency(); 534} 535 536status_t AudioFlinger::setMasterVolume(float value) 537{ 538 status_t ret = initCheck(); 539 if (ret != NO_ERROR) { 540 return ret; 541 } 542 543 // check calling permissions 544 if (!settingsAllowed()) { 545 return PERMISSION_DENIED; 546 } 547 548 // when hw supports master volume, don't scale in sw mixer 549 { // scope for the lock 550 AutoMutex lock(mHardwareLock); 551 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 552 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 553 value = 1.0f; 554 } 555 mHardwareStatus = AUDIO_HW_IDLE; 556 } 557 558 Mutex::Autolock _l(mLock); 559 mMasterVolume = value; 560 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 561 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 562 563 return NO_ERROR; 564} 565 566status_t AudioFlinger::setMode(int mode) 567{ 568 status_t ret = initCheck(); 569 if (ret != NO_ERROR) { 570 return ret; 571 } 572 573 // check calling permissions 574 if (!settingsAllowed()) { 575 return PERMISSION_DENIED; 576 } 577 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 578 ALOGW("Illegal value: setMode(%d)", mode); 579 return BAD_VALUE; 580 } 581 582 { // scope for the lock 583 AutoMutex lock(mHardwareLock); 584 mHardwareStatus = AUDIO_HW_SET_MODE; 585 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 586 mHardwareStatus = AUDIO_HW_IDLE; 587 } 588 589 if (NO_ERROR == ret) { 590 Mutex::Autolock _l(mLock); 591 mMode = mode; 592 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 593 mPlaybackThreads.valueAt(i)->setMode(mode); 594 } 595 596 return ret; 597} 598 599status_t AudioFlinger::setMicMute(bool state) 600{ 601 status_t ret = initCheck(); 602 if (ret != NO_ERROR) { 603 return ret; 604 } 605 606 // check calling permissions 607 if (!settingsAllowed()) { 608 return PERMISSION_DENIED; 609 } 610 611 AutoMutex lock(mHardwareLock); 612 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 613 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 614 mHardwareStatus = AUDIO_HW_IDLE; 615 return ret; 616} 617 618bool AudioFlinger::getMicMute() const 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return false; 623 } 624 625 bool state = AUDIO_MODE_INVALID; 626 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 627 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 628 mHardwareStatus = AUDIO_HW_IDLE; 629 return state; 630} 631 632status_t AudioFlinger::setMasterMute(bool muted) 633{ 634 // check calling permissions 635 if (!settingsAllowed()) { 636 return PERMISSION_DENIED; 637 } 638 639 Mutex::Autolock _l(mLock); 640 mMasterMute = muted; 641 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 642 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 643 644 return NO_ERROR; 645} 646 647float AudioFlinger::masterVolume() const 648{ 649 return mMasterVolume; 650} 651 652bool AudioFlinger::masterMute() const 653{ 654 return mMasterMute; 655} 656 657status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 658{ 659 // check calling permissions 660 if (!settingsAllowed()) { 661 return PERMISSION_DENIED; 662 } 663 664 if (stream < 0 || 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(int stream, bool muted) 692{ 693 // check calling permissions 694 if (!settingsAllowed()) { 695 return PERMISSION_DENIED; 696 } 697 698 if (stream < 0 || 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(int stream, int output) const 713{ 714 if (stream < 0 || 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(int stream) const 734{ 735 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 736 return true; 737 } 738 739 return mStreamTypes[stream].mute; 740} 741 742status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 743{ 744 status_t result; 745 746 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 747 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 748 // check calling permissions 749 if (!settingsAllowed()) { 750 return PERMISSION_DENIED; 751 } 752 753 // ioHandle == 0 means the parameters are global to the audio hardware interface 754 if (ioHandle == 0) { 755 AutoMutex lock(mHardwareLock); 756 mHardwareStatus = AUDIO_SET_PARAMETER; 757 status_t final_result = NO_ERROR; 758 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 759 audio_hw_device_t *dev = mAudioHwDevs[i]; 760 result = dev->set_parameters(dev, keyValuePairs.string()); 761 final_result = result ?: final_result; 762 } 763 mHardwareStatus = AUDIO_HW_IDLE; 764 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 765 AudioParameter param = AudioParameter(keyValuePairs); 766 String8 value; 767 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 768 Mutex::Autolock _l(mLock); 769 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 770 if (mBtNrecIsOff != btNrecIsOff) { 771 for (size_t i = 0; i < mRecordThreads.size(); i++) { 772 sp<RecordThread> thread = mRecordThreads.valueAt(i); 773 RecordThread::RecordTrack *track = thread->track(); 774 if (track != NULL) { 775 audio_devices_t device = (audio_devices_t)( 776 thread->device() & AUDIO_DEVICE_IN_ALL); 777 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 778 thread->setEffectSuspended(FX_IID_AEC, 779 suspend, 780 track->sessionId()); 781 thread->setEffectSuspended(FX_IID_NS, 782 suspend, 783 track->sessionId()); 784 } 785 } 786 mBtNrecIsOff = btNrecIsOff; 787 } 788 } 789 return final_result; 790 } 791 792 // hold a strong ref on thread in case closeOutput() or closeInput() is called 793 // and the thread is exited once the lock is released 794 sp<ThreadBase> thread; 795 { 796 Mutex::Autolock _l(mLock); 797 thread = checkPlaybackThread_l(ioHandle); 798 if (thread == NULL) { 799 thread = checkRecordThread_l(ioHandle); 800 } else if (thread.get() == 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 != NULL) { 812 result = thread->setParameters(keyValuePairs); 813 return result; 814 } 815 return BAD_VALUE; 816} 817 818String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 819{ 820// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 821// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 822 823 if (ioHandle == 0) { 824 String8 out_s8; 825 826 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 827 audio_hw_device_t *dev = mAudioHwDevs[i]; 828 char *s = dev->get_parameters(dev, keys.string()); 829 out_s8 += String8(s); 830 free(s); 831 } 832 return out_s8; 833 } 834 835 Mutex::Autolock _l(mLock); 836 837 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 838 if (playbackThread != NULL) { 839 return playbackThread->getParameters(keys); 840 } 841 RecordThread *recordThread = checkRecordThread_l(ioHandle); 842 if (recordThread != NULL) { 843 return recordThread->getParameters(keys); 844 } 845 return String8(""); 846} 847 848size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 849{ 850 status_t ret = initCheck(); 851 if (ret != NO_ERROR) { 852 return 0; 853 } 854 855 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 856} 857 858unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 859{ 860 if (ioHandle == 0) { 861 return 0; 862 } 863 864 Mutex::Autolock _l(mLock); 865 866 RecordThread *recordThread = checkRecordThread_l(ioHandle); 867 if (recordThread != NULL) { 868 return recordThread->getInputFramesLost(); 869 } 870 return 0; 871} 872 873status_t AudioFlinger::setVoiceVolume(float value) 874{ 875 status_t ret = initCheck(); 876 if (ret != NO_ERROR) { 877 return ret; 878 } 879 880 // check calling permissions 881 if (!settingsAllowed()) { 882 return PERMISSION_DENIED; 883 } 884 885 AutoMutex lock(mHardwareLock); 886 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 887 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 888 mHardwareStatus = AUDIO_HW_IDLE; 889 890 return ret; 891} 892 893status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 894{ 895 status_t status; 896 897 Mutex::Autolock _l(mLock); 898 899 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 900 if (playbackThread != NULL) { 901 return playbackThread->getRenderPosition(halFrames, dspFrames); 902 } 903 904 return BAD_VALUE; 905} 906 907void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 908{ 909 910 Mutex::Autolock _l(mLock); 911 912 int pid = IPCThreadState::self()->getCallingPid(); 913 if (mNotificationClients.indexOfKey(pid) < 0) { 914 sp<NotificationClient> notificationClient = new NotificationClient(this, 915 client, 916 pid); 917 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 918 919 mNotificationClients.add(pid, notificationClient); 920 921 sp<IBinder> binder = client->asBinder(); 922 binder->linkToDeath(notificationClient); 923 924 // the config change is always sent from playback or record threads to avoid deadlock 925 // with AudioSystem::gLock 926 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 927 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 928 } 929 930 for (size_t i = 0; i < mRecordThreads.size(); i++) { 931 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 932 } 933 } 934} 935 936void AudioFlinger::removeNotificationClient(pid_t pid) 937{ 938 Mutex::Autolock _l(mLock); 939 940 int index = mNotificationClients.indexOfKey(pid); 941 if (index >= 0) { 942 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 943 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 944 mNotificationClients.removeItem(pid); 945 } 946 947 ALOGV("%d died, releasing its sessions", pid); 948 int num = mAudioSessionRefs.size(); 949 bool removed = false; 950 for (int i = 0; i< num; i++) { 951 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 952 ALOGV(" pid %d @ %d", ref->pid, i); 953 if (ref->pid == pid) { 954 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 955 mAudioSessionRefs.removeAt(i); 956 delete ref; 957 removed = true; 958 i--; 959 num--; 960 } 961 } 962 if (removed) { 963 purgeStaleEffects_l(); 964 } 965} 966 967// audioConfigChanged_l() must be called with AudioFlinger::mLock held 968void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 969{ 970 size_t size = mNotificationClients.size(); 971 for (size_t i = 0; i < size; i++) { 972 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 973 } 974} 975 976// removeClient_l() must be called with AudioFlinger::mLock held 977void AudioFlinger::removeClient_l(pid_t pid) 978{ 979 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 980 mClients.removeItem(pid); 981} 982 983 984// ---------------------------------------------------------------------------- 985 986AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 987 : Thread(false), 988 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 989 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), 990 mDevice(device) 991{ 992 mDeathRecipient = new PMDeathRecipient(this); 993} 994 995AudioFlinger::ThreadBase::~ThreadBase() 996{ 997 mParamCond.broadcast(); 998 // do not lock the mutex in destructor 999 releaseWakeLock_l(); 1000 if (mPowerManager != 0) { 1001 sp<IBinder> binder = mPowerManager->asBinder(); 1002 binder->unlinkToDeath(mDeathRecipient); 1003 } 1004} 1005 1006void AudioFlinger::ThreadBase::exit() 1007{ 1008 // keep a strong ref on ourself so that we won't get 1009 // destroyed in the middle of requestExitAndWait() 1010 sp <ThreadBase> strongMe = this; 1011 1012 ALOGV("ThreadBase::exit"); 1013 { 1014 AutoMutex lock(mLock); 1015 mExiting = true; 1016 requestExit(); 1017 mWaitWorkCV.signal(); 1018 } 1019 requestExitAndWait(); 1020} 1021 1022uint32_t AudioFlinger::ThreadBase::sampleRate() const 1023{ 1024 return mSampleRate; 1025} 1026 1027int AudioFlinger::ThreadBase::channelCount() const 1028{ 1029 return (int)mChannelCount; 1030} 1031 1032uint32_t AudioFlinger::ThreadBase::format() const 1033{ 1034 return mFormat; 1035} 1036 1037size_t AudioFlinger::ThreadBase::frameCount() const 1038{ 1039 return mFrameCount; 1040} 1041 1042status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1043{ 1044 status_t status; 1045 1046 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1047 Mutex::Autolock _l(mLock); 1048 1049 mNewParameters.add(keyValuePairs); 1050 mWaitWorkCV.signal(); 1051 // wait condition with timeout in case the thread loop has exited 1052 // before the request could be processed 1053 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1054 status = mParamStatus; 1055 mWaitWorkCV.signal(); 1056 } else { 1057 status = TIMED_OUT; 1058 } 1059 return status; 1060} 1061 1062void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1063{ 1064 Mutex::Autolock _l(mLock); 1065 sendConfigEvent_l(event, param); 1066} 1067 1068// sendConfigEvent_l() must be called with ThreadBase::mLock held 1069void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1070{ 1071 ConfigEvent configEvent; 1072 configEvent.mEvent = event; 1073 configEvent.mParam = param; 1074 mConfigEvents.add(configEvent); 1075 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1076 mWaitWorkCV.signal(); 1077} 1078 1079void AudioFlinger::ThreadBase::processConfigEvents() 1080{ 1081 mLock.lock(); 1082 while(!mConfigEvents.isEmpty()) { 1083 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1084 ConfigEvent configEvent = mConfigEvents[0]; 1085 mConfigEvents.removeAt(0); 1086 // release mLock before locking AudioFlinger mLock: lock order is always 1087 // AudioFlinger then ThreadBase to avoid cross deadlock 1088 mLock.unlock(); 1089 mAudioFlinger->mLock.lock(); 1090 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1091 mAudioFlinger->mLock.unlock(); 1092 mLock.lock(); 1093 } 1094 mLock.unlock(); 1095} 1096 1097status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1098{ 1099 const size_t SIZE = 256; 1100 char buffer[SIZE]; 1101 String8 result; 1102 1103 bool locked = tryLock(mLock); 1104 if (!locked) { 1105 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1106 write(fd, buffer, strlen(buffer)); 1107 } 1108 1109 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1110 result.append(buffer); 1111 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1112 result.append(buffer); 1113 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1114 result.append(buffer); 1115 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1116 result.append(buffer); 1117 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 1122 result.append(buffer); 1123 1124 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1125 result.append(buffer); 1126 result.append(" Index Command"); 1127 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1128 snprintf(buffer, SIZE, "\n %02d ", i); 1129 result.append(buffer); 1130 result.append(mNewParameters[i]); 1131 } 1132 1133 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1134 result.append(buffer); 1135 snprintf(buffer, SIZE, " Index event param\n"); 1136 result.append(buffer); 1137 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1138 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1139 result.append(buffer); 1140 } 1141 result.append("\n"); 1142 1143 write(fd, result.string(), result.size()); 1144 1145 if (locked) { 1146 mLock.unlock(); 1147 } 1148 return NO_ERROR; 1149} 1150 1151status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1152{ 1153 const size_t SIZE = 256; 1154 char buffer[SIZE]; 1155 String8 result; 1156 1157 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1158 write(fd, buffer, strlen(buffer)); 1159 1160 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1161 sp<EffectChain> chain = mEffectChains[i]; 1162 if (chain != 0) { 1163 chain->dump(fd, args); 1164 } 1165 } 1166 return NO_ERROR; 1167} 1168 1169void AudioFlinger::ThreadBase::acquireWakeLock() 1170{ 1171 Mutex::Autolock _l(mLock); 1172 acquireWakeLock_l(); 1173} 1174 1175void AudioFlinger::ThreadBase::acquireWakeLock_l() 1176{ 1177 if (mPowerManager == 0) { 1178 // use checkService() to avoid blocking if power service is not up yet 1179 sp<IBinder> binder = 1180 defaultServiceManager()->checkService(String16("power")); 1181 if (binder == 0) { 1182 ALOGW("Thread %s cannot connect to the power manager service", mName); 1183 } else { 1184 mPowerManager = interface_cast<IPowerManager>(binder); 1185 binder->linkToDeath(mDeathRecipient); 1186 } 1187 } 1188 if (mPowerManager != 0) { 1189 sp<IBinder> binder = new BBinder(); 1190 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1191 binder, 1192 String16(mName)); 1193 if (status == NO_ERROR) { 1194 mWakeLockToken = binder; 1195 } 1196 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1197 } 1198} 1199 1200void AudioFlinger::ThreadBase::releaseWakeLock() 1201{ 1202 Mutex::Autolock _l(mLock); 1203 releaseWakeLock_l(); 1204} 1205 1206void AudioFlinger::ThreadBase::releaseWakeLock_l() 1207{ 1208 if (mWakeLockToken != 0) { 1209 ALOGV("releaseWakeLock_l() %s", mName); 1210 if (mPowerManager != 0) { 1211 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1212 } 1213 mWakeLockToken.clear(); 1214 } 1215} 1216 1217void AudioFlinger::ThreadBase::clearPowerManager() 1218{ 1219 Mutex::Autolock _l(mLock); 1220 releaseWakeLock_l(); 1221 mPowerManager.clear(); 1222} 1223 1224void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1225{ 1226 sp<ThreadBase> thread = mThread.promote(); 1227 if (thread != 0) { 1228 thread->clearPowerManager(); 1229 } 1230 ALOGW("power manager service died !!!"); 1231} 1232 1233void AudioFlinger::ThreadBase::setEffectSuspended( 1234 const effect_uuid_t *type, bool suspend, int sessionId) 1235{ 1236 Mutex::Autolock _l(mLock); 1237 setEffectSuspended_l(type, suspend, sessionId); 1238} 1239 1240void AudioFlinger::ThreadBase::setEffectSuspended_l( 1241 const effect_uuid_t *type, bool suspend, int sessionId) 1242{ 1243 sp<EffectChain> chain; 1244 chain = getEffectChain_l(sessionId); 1245 if (chain != 0) { 1246 if (type != NULL) { 1247 chain->setEffectSuspended_l(type, suspend); 1248 } else { 1249 chain->setEffectSuspendedAll_l(suspend); 1250 } 1251 } 1252 1253 updateSuspendedSessions_l(type, suspend, sessionId); 1254} 1255 1256void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1257{ 1258 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1259 if (index < 0) { 1260 return; 1261 } 1262 1263 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1264 mSuspendedSessions.editValueAt(index); 1265 1266 for (size_t i = 0; i < sessionEffects.size(); i++) { 1267 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1268 for (int j = 0; j < desc->mRefCount; j++) { 1269 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1270 chain->setEffectSuspendedAll_l(true); 1271 } else { 1272 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1273 desc->mType.timeLow); 1274 chain->setEffectSuspended_l(&desc->mType, true); 1275 } 1276 } 1277 } 1278} 1279 1280void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1281 bool suspend, 1282 int sessionId) 1283{ 1284 int index = mSuspendedSessions.indexOfKey(sessionId); 1285 1286 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1287 1288 if (suspend) { 1289 if (index >= 0) { 1290 sessionEffects = mSuspendedSessions.editValueAt(index); 1291 } else { 1292 mSuspendedSessions.add(sessionId, sessionEffects); 1293 } 1294 } else { 1295 if (index < 0) { 1296 return; 1297 } 1298 sessionEffects = mSuspendedSessions.editValueAt(index); 1299 } 1300 1301 1302 int key = EffectChain::kKeyForSuspendAll; 1303 if (type != NULL) { 1304 key = type->timeLow; 1305 } 1306 index = sessionEffects.indexOfKey(key); 1307 1308 sp <SuspendedSessionDesc> desc; 1309 if (suspend) { 1310 if (index >= 0) { 1311 desc = sessionEffects.valueAt(index); 1312 } else { 1313 desc = new SuspendedSessionDesc(); 1314 if (type != NULL) { 1315 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1316 } 1317 sessionEffects.add(key, desc); 1318 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1319 } 1320 desc->mRefCount++; 1321 } else { 1322 if (index < 0) { 1323 return; 1324 } 1325 desc = sessionEffects.valueAt(index); 1326 if (--desc->mRefCount == 0) { 1327 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1328 sessionEffects.removeItemsAt(index); 1329 if (sessionEffects.isEmpty()) { 1330 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1331 sessionId); 1332 mSuspendedSessions.removeItem(sessionId); 1333 } 1334 } 1335 } 1336 if (!sessionEffects.isEmpty()) { 1337 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1338 } 1339} 1340 1341void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1342 bool enabled, 1343 int sessionId) 1344{ 1345 Mutex::Autolock _l(mLock); 1346 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1347} 1348 1349void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1350 bool enabled, 1351 int sessionId) 1352{ 1353 if (mType != RECORD) { 1354 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1355 // another session. This gives the priority to well behaved effect control panels 1356 // and applications not using global effects. 1357 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1358 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1359 } 1360 } 1361 1362 sp<EffectChain> chain = getEffectChain_l(sessionId); 1363 if (chain != 0) { 1364 chain->checkSuspendOnEffectEnabled(effect, enabled); 1365 } 1366} 1367 1368// ---------------------------------------------------------------------------- 1369 1370AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1371 AudioStreamOut* output, 1372 int id, 1373 uint32_t device) 1374 : ThreadBase(audioFlinger, id, device), 1375 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1376 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1377{ 1378 snprintf(mName, kNameLength, "AudioOut_%d", id); 1379 1380 readOutputParameters(); 1381 1382 mMasterVolume = mAudioFlinger->masterVolume(); 1383 mMasterMute = mAudioFlinger->masterMute(); 1384 1385 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 1386 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1387 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1388 mStreamTypes[stream].valid = true; 1389 } 1390} 1391 1392AudioFlinger::PlaybackThread::~PlaybackThread() 1393{ 1394 delete [] mMixBuffer; 1395} 1396 1397status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1398{ 1399 dumpInternals(fd, args); 1400 dumpTracks(fd, args); 1401 dumpEffectChains(fd, args); 1402 return NO_ERROR; 1403} 1404 1405status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1406{ 1407 const size_t SIZE = 256; 1408 char buffer[SIZE]; 1409 String8 result; 1410 1411 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1412 result.append(buffer); 1413 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1414 for (size_t i = 0; i < mTracks.size(); ++i) { 1415 sp<Track> track = mTracks[i]; 1416 if (track != 0) { 1417 track->dump(buffer, SIZE); 1418 result.append(buffer); 1419 } 1420 } 1421 1422 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1423 result.append(buffer); 1424 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1425 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1426 wp<Track> wTrack = mActiveTracks[i]; 1427 if (wTrack != 0) { 1428 sp<Track> track = wTrack.promote(); 1429 if (track != 0) { 1430 track->dump(buffer, SIZE); 1431 result.append(buffer); 1432 } 1433 } 1434 } 1435 write(fd, result.string(), result.size()); 1436 return NO_ERROR; 1437} 1438 1439status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1440{ 1441 const size_t SIZE = 256; 1442 char buffer[SIZE]; 1443 String8 result; 1444 1445 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1446 result.append(buffer); 1447 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1448 result.append(buffer); 1449 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1450 result.append(buffer); 1451 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1452 result.append(buffer); 1453 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1454 result.append(buffer); 1455 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1456 result.append(buffer); 1457 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1458 result.append(buffer); 1459 write(fd, result.string(), result.size()); 1460 1461 dumpBase(fd, args); 1462 1463 return NO_ERROR; 1464} 1465 1466// Thread virtuals 1467status_t AudioFlinger::PlaybackThread::readyToRun() 1468{ 1469 status_t status = initCheck(); 1470 if (status == NO_ERROR) { 1471 ALOGI("AudioFlinger's thread %p ready to run", this); 1472 } else { 1473 ALOGE("No working audio driver found."); 1474 } 1475 return status; 1476} 1477 1478void AudioFlinger::PlaybackThread::onFirstRef() 1479{ 1480 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1481} 1482 1483// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1484sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1485 const sp<AudioFlinger::Client>& client, 1486 int streamType, 1487 uint32_t sampleRate, 1488 uint32_t format, 1489 uint32_t channelMask, 1490 int frameCount, 1491 const sp<IMemory>& sharedBuffer, 1492 int sessionId, 1493 status_t *status) 1494{ 1495 sp<Track> track; 1496 status_t lStatus; 1497 1498 if (mType == DIRECT) { 1499 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1500 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1501 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1502 "for output %p with format %d", 1503 sampleRate, format, channelMask, mOutput, mFormat); 1504 lStatus = BAD_VALUE; 1505 goto Exit; 1506 } 1507 } 1508 } else { 1509 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1510 if (sampleRate > mSampleRate*2) { 1511 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1512 lStatus = BAD_VALUE; 1513 goto Exit; 1514 } 1515 } 1516 1517 lStatus = initCheck(); 1518 if (lStatus != NO_ERROR) { 1519 ALOGE("Audio driver not initialized."); 1520 goto Exit; 1521 } 1522 1523 { // scope for mLock 1524 Mutex::Autolock _l(mLock); 1525 1526 // all tracks in same audio session must share the same routing strategy otherwise 1527 // conflicts will happen when tracks are moved from one output to another by audio policy 1528 // manager 1529 uint32_t strategy = 1530 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1531 for (size_t i = 0; i < mTracks.size(); ++i) { 1532 sp<Track> t = mTracks[i]; 1533 if (t != 0) { 1534 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1535 if (sessionId == t->sessionId() && strategy != actual) { 1536 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1537 strategy, actual); 1538 lStatus = BAD_VALUE; 1539 goto Exit; 1540 } 1541 } 1542 } 1543 1544 track = new Track(this, client, streamType, sampleRate, format, 1545 channelMask, frameCount, sharedBuffer, sessionId); 1546 if (track->getCblk() == NULL || track->name() < 0) { 1547 lStatus = NO_MEMORY; 1548 goto Exit; 1549 } 1550 mTracks.add(track); 1551 1552 sp<EffectChain> chain = getEffectChain_l(sessionId); 1553 if (chain != 0) { 1554 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1555 track->setMainBuffer(chain->inBuffer()); 1556 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1557 chain->incTrackCnt(); 1558 } 1559 1560 // invalidate track immediately if the stream type was moved to another thread since 1561 // createTrack() was called by the client process. 1562 if (!mStreamTypes[streamType].valid) { 1563 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1564 this, streamType); 1565 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1566 } 1567 } 1568 lStatus = NO_ERROR; 1569 1570Exit: 1571 if(status) { 1572 *status = lStatus; 1573 } 1574 return track; 1575} 1576 1577uint32_t AudioFlinger::PlaybackThread::latency() const 1578{ 1579 Mutex::Autolock _l(mLock); 1580 if (initCheck() == NO_ERROR) { 1581 return mOutput->stream->get_latency(mOutput->stream); 1582 } else { 1583 return 0; 1584 } 1585} 1586 1587status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1588{ 1589 mMasterVolume = value; 1590 return NO_ERROR; 1591} 1592 1593status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1594{ 1595 mMasterMute = muted; 1596 return NO_ERROR; 1597} 1598 1599float AudioFlinger::PlaybackThread::masterVolume() const 1600{ 1601 return mMasterVolume; 1602} 1603 1604bool AudioFlinger::PlaybackThread::masterMute() const 1605{ 1606 return mMasterMute; 1607} 1608 1609status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1610{ 1611 mStreamTypes[stream].volume = value; 1612 return NO_ERROR; 1613} 1614 1615status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1616{ 1617 mStreamTypes[stream].mute = muted; 1618 return NO_ERROR; 1619} 1620 1621float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1622{ 1623 return mStreamTypes[stream].volume; 1624} 1625 1626bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1627{ 1628 return mStreamTypes[stream].mute; 1629} 1630 1631// addTrack_l() must be called with ThreadBase::mLock held 1632status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1633{ 1634 status_t status = ALREADY_EXISTS; 1635 1636 // set retry count for buffer fill 1637 track->mRetryCount = kMaxTrackStartupRetries; 1638 if (mActiveTracks.indexOf(track) < 0) { 1639 // the track is newly added, make sure it fills up all its 1640 // buffers before playing. This is to ensure the client will 1641 // effectively get the latency it requested. 1642 track->mFillingUpStatus = Track::FS_FILLING; 1643 track->mResetDone = false; 1644 mActiveTracks.add(track); 1645 if (track->mainBuffer() != mMixBuffer) { 1646 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1647 if (chain != 0) { 1648 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1649 chain->incActiveTrackCnt(); 1650 } 1651 } 1652 1653 status = NO_ERROR; 1654 } 1655 1656 ALOGV("mWaitWorkCV.broadcast"); 1657 mWaitWorkCV.broadcast(); 1658 1659 return status; 1660} 1661 1662// destroyTrack_l() must be called with ThreadBase::mLock held 1663void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1664{ 1665 track->mState = TrackBase::TERMINATED; 1666 if (mActiveTracks.indexOf(track) < 0) { 1667 removeTrack_l(track); 1668 } 1669} 1670 1671void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1672{ 1673 mTracks.remove(track); 1674 deleteTrackName_l(track->name()); 1675 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1676 if (chain != 0) { 1677 chain->decTrackCnt(); 1678 } 1679} 1680 1681String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1682{ 1683 String8 out_s8 = String8(""); 1684 char *s; 1685 1686 Mutex::Autolock _l(mLock); 1687 if (initCheck() != NO_ERROR) { 1688 return out_s8; 1689 } 1690 1691 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1692 out_s8 = String8(s); 1693 free(s); 1694 return out_s8; 1695} 1696 1697// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1698void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1699 AudioSystem::OutputDescriptor desc; 1700 void *param2 = 0; 1701 1702 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1703 1704 switch (event) { 1705 case AudioSystem::OUTPUT_OPENED: 1706 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1707 desc.channels = mChannelMask; 1708 desc.samplingRate = mSampleRate; 1709 desc.format = mFormat; 1710 desc.frameCount = mFrameCount; 1711 desc.latency = latency(); 1712 param2 = &desc; 1713 break; 1714 1715 case AudioSystem::STREAM_CONFIG_CHANGED: 1716 param2 = ¶m; 1717 case AudioSystem::OUTPUT_CLOSED: 1718 default: 1719 break; 1720 } 1721 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1722} 1723 1724void AudioFlinger::PlaybackThread::readOutputParameters() 1725{ 1726 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1727 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1728 mChannelCount = (uint16_t)popcount(mChannelMask); 1729 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1730 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); 1731 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1732 1733 // FIXME - Current mixer implementation only supports stereo output: Always 1734 // Allocate a stereo buffer even if HW output is mono. 1735 if (mMixBuffer != NULL) delete[] mMixBuffer; 1736 mMixBuffer = new int16_t[mFrameCount * 2]; 1737 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1738 1739 // force reconfiguration of effect chains and engines to take new buffer size and audio 1740 // parameters into account 1741 // Note that mLock is not held when readOutputParameters() is called from the constructor 1742 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1743 // matter. 1744 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1745 Vector< sp<EffectChain> > effectChains = mEffectChains; 1746 for (size_t i = 0; i < effectChains.size(); i ++) { 1747 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1748 } 1749} 1750 1751status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1752{ 1753 if (halFrames == 0 || dspFrames == 0) { 1754 return BAD_VALUE; 1755 } 1756 Mutex::Autolock _l(mLock); 1757 if (initCheck() != NO_ERROR) { 1758 return INVALID_OPERATION; 1759 } 1760 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1761 1762 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1763} 1764 1765uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1766{ 1767 Mutex::Autolock _l(mLock); 1768 uint32_t result = 0; 1769 if (getEffectChain_l(sessionId) != 0) { 1770 result = EFFECT_SESSION; 1771 } 1772 1773 for (size_t i = 0; i < mTracks.size(); ++i) { 1774 sp<Track> track = mTracks[i]; 1775 if (sessionId == track->sessionId() && 1776 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1777 result |= TRACK_SESSION; 1778 break; 1779 } 1780 } 1781 1782 return result; 1783} 1784 1785uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1786{ 1787 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1788 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1789 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1790 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1791 } 1792 for (size_t i = 0; i < mTracks.size(); i++) { 1793 sp<Track> track = mTracks[i]; 1794 if (sessionId == track->sessionId() && 1795 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1796 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1797 } 1798 } 1799 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1800} 1801 1802 1803AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() 1804{ 1805 Mutex::Autolock _l(mLock); 1806 return mOutput; 1807} 1808 1809AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1810{ 1811 Mutex::Autolock _l(mLock); 1812 AudioStreamOut *output = mOutput; 1813 mOutput = NULL; 1814 return output; 1815} 1816 1817// this method must always be called either with ThreadBase mLock held or inside the thread loop 1818audio_stream_t* AudioFlinger::PlaybackThread::stream() 1819{ 1820 if (mOutput == NULL) { 1821 return NULL; 1822 } 1823 return &mOutput->stream->common; 1824} 1825 1826uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1827{ 1828 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1829 // decoding and transfer time. So sleeping for half of the latency would likely cause 1830 // underruns 1831 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1832 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1833 } else { 1834 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1835 } 1836} 1837 1838// ---------------------------------------------------------------------------- 1839 1840AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1841 : PlaybackThread(audioFlinger, output, id, device), 1842 mAudioMixer(NULL) 1843{ 1844 mType = ThreadBase::MIXER; 1845 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1846 1847 // FIXME - Current mixer implementation only supports stereo output 1848 if (mChannelCount == 1) { 1849 ALOGE("Invalid audio hardware channel count"); 1850 } 1851} 1852 1853AudioFlinger::MixerThread::~MixerThread() 1854{ 1855 delete mAudioMixer; 1856} 1857 1858bool AudioFlinger::MixerThread::threadLoop() 1859{ 1860 Vector< sp<Track> > tracksToRemove; 1861 uint32_t mixerStatus = MIXER_IDLE; 1862 nsecs_t standbyTime = systemTime(); 1863 size_t mixBufferSize = mFrameCount * mFrameSize; 1864 // FIXME: Relaxed timing because of a certain device that can't meet latency 1865 // Should be reduced to 2x after the vendor fixes the driver issue 1866 // increase threshold again due to low power audio mode. The way this warning threshold is 1867 // calculated and its usefulness should be reconsidered anyway. 1868 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1869 nsecs_t lastWarning = 0; 1870 bool longStandbyExit = false; 1871 uint32_t activeSleepTime = activeSleepTimeUs(); 1872 uint32_t idleSleepTime = idleSleepTimeUs(); 1873 uint32_t sleepTime = idleSleepTime; 1874 uint32_t sleepTimeShift = 0; 1875 Vector< sp<EffectChain> > effectChains; 1876#ifdef DEBUG_CPU_USAGE 1877 ThreadCpuUsage cpu; 1878 const CentralTendencyStatistics& stats = cpu.statistics(); 1879#endif 1880 1881 acquireWakeLock(); 1882 1883 while (!exitPending()) 1884 { 1885#ifdef DEBUG_CPU_USAGE 1886 cpu.sampleAndEnable(); 1887 unsigned n = stats.n(); 1888 // cpu.elapsed() is expensive, so don't call it every loop 1889 if ((n & 127) == 1) { 1890 long long elapsed = cpu.elapsed(); 1891 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1892 double perLoop = elapsed / (double) n; 1893 double perLoop100 = perLoop * 0.01; 1894 double mean = stats.mean(); 1895 double stddev = stats.stddev(); 1896 double minimum = stats.minimum(); 1897 double maximum = stats.maximum(); 1898 cpu.resetStatistics(); 1899 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", 1900 elapsed * .000000001, n, perLoop * .000001, 1901 mean * .001, 1902 stddev * .001, 1903 minimum * .001, 1904 maximum * .001, 1905 mean / perLoop100, 1906 stddev / perLoop100, 1907 minimum / perLoop100, 1908 maximum / perLoop100); 1909 } 1910 } 1911#endif 1912 processConfigEvents(); 1913 1914 mixerStatus = MIXER_IDLE; 1915 { // scope for mLock 1916 1917 Mutex::Autolock _l(mLock); 1918 1919 if (checkForNewParameters_l()) { 1920 mixBufferSize = mFrameCount * mFrameSize; 1921 // FIXME: Relaxed timing because of a certain device that can't meet latency 1922 // Should be reduced to 2x after the vendor fixes the driver issue 1923 // increase threshold again due to low power audio mode. The way this warning 1924 // threshold is calculated and its usefulness should be reconsidered anyway. 1925 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1926 activeSleepTime = activeSleepTimeUs(); 1927 idleSleepTime = idleSleepTimeUs(); 1928 } 1929 1930 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1931 1932 // put audio hardware into standby after short delay 1933 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1934 mSuspended)) { 1935 if (!mStandby) { 1936 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1937 mOutput->stream->common.standby(&mOutput->stream->common); 1938 mStandby = true; 1939 mBytesWritten = 0; 1940 } 1941 1942 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1943 // we're about to wait, flush the binder command buffer 1944 IPCThreadState::self()->flushCommands(); 1945 1946 if (exitPending()) break; 1947 1948 releaseWakeLock_l(); 1949 // wait until we have something to do... 1950 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1951 mWaitWorkCV.wait(mLock); 1952 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1953 acquireWakeLock_l(); 1954 1955 if (mMasterMute == false) { 1956 char value[PROPERTY_VALUE_MAX]; 1957 property_get("ro.audio.silent", value, "0"); 1958 if (atoi(value)) { 1959 ALOGD("Silence is golden"); 1960 setMasterMute(true); 1961 } 1962 } 1963 1964 standbyTime = systemTime() + kStandbyTimeInNsecs; 1965 sleepTime = idleSleepTime; 1966 sleepTimeShift = 0; 1967 continue; 1968 } 1969 } 1970 1971 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1972 1973 // prevent any changes in effect chain list and in each effect chain 1974 // during mixing and effect process as the audio buffers could be deleted 1975 // or modified if an effect is created or deleted 1976 lockEffectChains_l(effectChains); 1977 } 1978 1979 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1980 // mix buffers... 1981 mAudioMixer->process(); 1982 sleepTime = 0; 1983 // increase sleep time progressively when application underrun condition clears 1984 if (sleepTimeShift > 0) { 1985 sleepTimeShift--; 1986 } 1987 standbyTime = systemTime() + kStandbyTimeInNsecs; 1988 //TODO: delay standby when effects have a tail 1989 } else { 1990 // If no tracks are ready, sleep once for the duration of an output 1991 // buffer size, then write 0s to the output 1992 if (sleepTime == 0) { 1993 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1994 sleepTime = activeSleepTime >> sleepTimeShift; 1995 if (sleepTime < kMinThreadSleepTimeUs) { 1996 sleepTime = kMinThreadSleepTimeUs; 1997 } 1998 // reduce sleep time in case of consecutive application underruns to avoid 1999 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2000 // duration we would end up writing less data than needed by the audio HAL if 2001 // the condition persists. 2002 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2003 sleepTimeShift++; 2004 } 2005 } else { 2006 sleepTime = idleSleepTime; 2007 } 2008 } else if (mBytesWritten != 0 || 2009 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2010 memset (mMixBuffer, 0, mixBufferSize); 2011 sleepTime = 0; 2012 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2013 } 2014 // TODO add standby time extension fct of effect tail 2015 } 2016 2017 if (mSuspended) { 2018 sleepTime = suspendSleepTimeUs(); 2019 } 2020 // sleepTime == 0 means we must write to audio hardware 2021 if (sleepTime == 0) { 2022 for (size_t i = 0; i < effectChains.size(); i ++) { 2023 effectChains[i]->process_l(); 2024 } 2025 // enable changes in effect chain 2026 unlockEffectChains(effectChains); 2027 mLastWriteTime = systemTime(); 2028 mInWrite = true; 2029 mBytesWritten += mixBufferSize; 2030 2031 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2032 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2033 mNumWrites++; 2034 mInWrite = false; 2035 nsecs_t now = systemTime(); 2036 nsecs_t delta = now - mLastWriteTime; 2037 if (!mStandby && delta > maxPeriod) { 2038 mNumDelayedWrites++; 2039 if ((now - lastWarning) > kWarningThrottleNs) { 2040 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2041 ns2ms(delta), mNumDelayedWrites, this); 2042 lastWarning = now; 2043 } 2044 if (mStandby) { 2045 longStandbyExit = true; 2046 } 2047 } 2048 mStandby = false; 2049 } else { 2050 // enable changes in effect chain 2051 unlockEffectChains(effectChains); 2052 usleep(sleepTime); 2053 } 2054 2055 // finally let go of all our tracks, without the lock held 2056 // since we can't guarantee the destructors won't acquire that 2057 // same lock. 2058 tracksToRemove.clear(); 2059 2060 // Effect chains will be actually deleted here if they were removed from 2061 // mEffectChains list during mixing or effects processing 2062 effectChains.clear(); 2063 } 2064 2065 if (!mStandby) { 2066 mOutput->stream->common.standby(&mOutput->stream->common); 2067 } 2068 2069 releaseWakeLock(); 2070 2071 ALOGV("MixerThread %p exiting", this); 2072 return false; 2073} 2074 2075// prepareTracks_l() must be called with ThreadBase::mLock held 2076uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2077{ 2078 2079 uint32_t mixerStatus = MIXER_IDLE; 2080 // find out which tracks need to be processed 2081 size_t count = activeTracks.size(); 2082 size_t mixedTracks = 0; 2083 size_t tracksWithEffect = 0; 2084 2085 float masterVolume = mMasterVolume; 2086 bool masterMute = mMasterMute; 2087 2088 if (masterMute) { 2089 masterVolume = 0; 2090 } 2091 // Delegate master volume control to effect in output mix effect chain if needed 2092 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2093 if (chain != 0) { 2094 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2095 chain->setVolume_l(&v, &v); 2096 masterVolume = (float)((v + (1 << 23)) >> 24); 2097 chain.clear(); 2098 } 2099 2100 for (size_t i=0 ; i<count ; i++) { 2101 sp<Track> t = activeTracks[i].promote(); 2102 if (t == 0) continue; 2103 2104 // this const just means the local variable doesn't change 2105 Track* const track = t.get(); 2106 audio_track_cblk_t* cblk = track->cblk(); 2107 2108 // The first time a track is added we wait 2109 // for all its buffers to be filled before processing it 2110 int name = track->name(); 2111 // make sure that we have enough frames to mix one full buffer. 2112 // enforce this condition only once to enable draining the buffer in case the client 2113 // app does not call stop() and relies on underrun to stop: 2114 // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed 2115 // during last round 2116 uint32_t minFrames = 1; 2117 if (!track->isStopped() && !track->isPausing() && 2118 (track->mRetryCount >= kMaxTrackRetries)) { 2119 if (t->sampleRate() == (int)mSampleRate) { 2120 minFrames = mFrameCount; 2121 } else { 2122 // +1 for rounding and +1 for additional sample needed for interpolation 2123 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2124 // add frames already consumed but not yet released by the resampler 2125 // because cblk->framesReady() will include these frames 2126 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2127 // the minimum track buffer size is normally twice the number of frames necessary 2128 // to fill one buffer and the resampler should not leave more than one buffer worth 2129 // of unreleased frames after each pass, but just in case... 2130 ALOG_ASSERT(minFrames <= cblk->frameCount); 2131 } 2132 } 2133 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2134 !track->isPaused() && !track->isTerminated()) 2135 { 2136 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2137 2138 mixedTracks++; 2139 2140 // track->mainBuffer() != mMixBuffer means there is an effect chain 2141 // connected to the track 2142 chain.clear(); 2143 if (track->mainBuffer() != mMixBuffer) { 2144 chain = getEffectChain_l(track->sessionId()); 2145 // Delegate volume control to effect in track effect chain if needed 2146 if (chain != 0) { 2147 tracksWithEffect++; 2148 } else { 2149 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2150 name, track->sessionId()); 2151 } 2152 } 2153 2154 2155 int param = AudioMixer::VOLUME; 2156 if (track->mFillingUpStatus == Track::FS_FILLED) { 2157 // no ramp for the first volume setting 2158 track->mFillingUpStatus = Track::FS_ACTIVE; 2159 if (track->mState == TrackBase::RESUMING) { 2160 track->mState = TrackBase::ACTIVE; 2161 param = AudioMixer::RAMP_VOLUME; 2162 } 2163 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2164 } else if (cblk->server != 0) { 2165 // If the track is stopped before the first frame was mixed, 2166 // do not apply ramp 2167 param = AudioMixer::RAMP_VOLUME; 2168 } 2169 2170 // compute volume for this track 2171 uint32_t vl, vr, va; 2172 if (track->isMuted() || track->isPausing() || 2173 mStreamTypes[track->type()].mute) { 2174 vl = vr = va = 0; 2175 if (track->isPausing()) { 2176 track->setPaused(); 2177 } 2178 } else { 2179 2180 // read original volumes with volume control 2181 float typeVolume = mStreamTypes[track->type()].volume; 2182 float v = masterVolume * typeVolume; 2183 vl = (uint32_t)(v * cblk->volume[0]) << 12; 2184 vr = (uint32_t)(v * cblk->volume[1]) << 12; 2185 2186 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2187 // send level comes from shared memory and so may be corrupt 2188 if (sendLevel >= 0x1000) { 2189 ALOGV("Track send level out of range: %04X", sendLevel); 2190 sendLevel = 0x1000; 2191 } 2192 va = (uint32_t)(v * sendLevel); 2193 } 2194 // Delegate volume control to effect in track effect chain if needed 2195 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2196 // Do not ramp volume if volume is controlled by effect 2197 param = AudioMixer::VOLUME; 2198 track->mHasVolumeController = true; 2199 } else { 2200 // force no volume ramp when volume controller was just disabled or removed 2201 // from effect chain to avoid volume spike 2202 if (track->mHasVolumeController) { 2203 param = AudioMixer::VOLUME; 2204 } 2205 track->mHasVolumeController = false; 2206 } 2207 2208 // Convert volumes from 8.24 to 4.12 format 2209 int16_t left, right, aux; 2210 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2211 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2212 left = int16_t(v_clamped); 2213 v_clamped = (vr + (1 << 11)) >> 12; 2214 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2215 right = int16_t(v_clamped); 2216 2217 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2218 aux = int16_t(va); 2219 2220 // XXX: these things DON'T need to be done each time 2221 mAudioMixer->setBufferProvider(name, track); 2222 mAudioMixer->enable(name); 2223 2224 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2225 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2226 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2227 mAudioMixer->setParameter( 2228 name, 2229 AudioMixer::TRACK, 2230 AudioMixer::FORMAT, (void *)track->format()); 2231 mAudioMixer->setParameter( 2232 name, 2233 AudioMixer::TRACK, 2234 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2235 mAudioMixer->setParameter( 2236 name, 2237 AudioMixer::RESAMPLE, 2238 AudioMixer::SAMPLE_RATE, 2239 (void *)(cblk->sampleRate)); 2240 mAudioMixer->setParameter( 2241 name, 2242 AudioMixer::TRACK, 2243 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2244 mAudioMixer->setParameter( 2245 name, 2246 AudioMixer::TRACK, 2247 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2248 2249 // reset retry count 2250 track->mRetryCount = kMaxTrackRetries; 2251 mixerStatus = MIXER_TRACKS_READY; 2252 } else { 2253 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2254 if (track->isStopped()) { 2255 track->reset(); 2256 } 2257 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2258 // We have consumed all the buffers of this track. 2259 // Remove it from the list of active tracks. 2260 tracksToRemove->add(track); 2261 } else { 2262 // No buffers for this track. Give it a few chances to 2263 // fill a buffer, then remove it from active list. 2264 if (--(track->mRetryCount) <= 0) { 2265 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2266 tracksToRemove->add(track); 2267 // indicate to client process that the track was disabled because of underrun 2268 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2269 } else if (mixerStatus != MIXER_TRACKS_READY) { 2270 mixerStatus = MIXER_TRACKS_ENABLED; 2271 } 2272 } 2273 mAudioMixer->disable(name); 2274 } 2275 } 2276 2277 // remove all the tracks that need to be... 2278 count = tracksToRemove->size(); 2279 if (CC_UNLIKELY(count)) { 2280 for (size_t i=0 ; i<count ; i++) { 2281 const sp<Track>& track = tracksToRemove->itemAt(i); 2282 mActiveTracks.remove(track); 2283 if (track->mainBuffer() != mMixBuffer) { 2284 chain = getEffectChain_l(track->sessionId()); 2285 if (chain != 0) { 2286 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2287 chain->decActiveTrackCnt(); 2288 } 2289 } 2290 if (track->isTerminated()) { 2291 removeTrack_l(track); 2292 } 2293 } 2294 } 2295 2296 // mix buffer must be cleared if all tracks are connected to an 2297 // effect chain as in this case the mixer will not write to 2298 // mix buffer and track effects will accumulate into it 2299 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2300 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2301 } 2302 2303 return mixerStatus; 2304} 2305 2306void AudioFlinger::MixerThread::invalidateTracks(int streamType) 2307{ 2308 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2309 this, streamType, mTracks.size()); 2310 Mutex::Autolock _l(mLock); 2311 2312 size_t size = mTracks.size(); 2313 for (size_t i = 0; i < size; i++) { 2314 sp<Track> t = mTracks[i]; 2315 if (t->type() == streamType) { 2316 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2317 t->mCblk->cv.signal(); 2318 } 2319 } 2320} 2321 2322void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid) 2323{ 2324 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2325 this, streamType, valid); 2326 Mutex::Autolock _l(mLock); 2327 2328 mStreamTypes[streamType].valid = valid; 2329} 2330 2331// getTrackName_l() must be called with ThreadBase::mLock held 2332int AudioFlinger::MixerThread::getTrackName_l() 2333{ 2334 return mAudioMixer->getTrackName(); 2335} 2336 2337// deleteTrackName_l() must be called with ThreadBase::mLock held 2338void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2339{ 2340 ALOGV("remove track (%d) and delete from mixer", name); 2341 mAudioMixer->deleteTrackName(name); 2342} 2343 2344// checkForNewParameters_l() must be called with ThreadBase::mLock held 2345bool AudioFlinger::MixerThread::checkForNewParameters_l() 2346{ 2347 bool reconfig = false; 2348 2349 while (!mNewParameters.isEmpty()) { 2350 status_t status = NO_ERROR; 2351 String8 keyValuePair = mNewParameters[0]; 2352 AudioParameter param = AudioParameter(keyValuePair); 2353 int value; 2354 2355 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2356 reconfig = true; 2357 } 2358 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2359 if (value != AUDIO_FORMAT_PCM_16_BIT) { 2360 status = BAD_VALUE; 2361 } else { 2362 reconfig = true; 2363 } 2364 } 2365 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2366 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2367 status = BAD_VALUE; 2368 } else { 2369 reconfig = true; 2370 } 2371 } 2372 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2373 // do not accept frame count changes if tracks are open as the track buffer 2374 // size depends on frame count and correct behavior would not be guaranteed 2375 // if frame count is changed after track creation 2376 if (!mTracks.isEmpty()) { 2377 status = INVALID_OPERATION; 2378 } else { 2379 reconfig = true; 2380 } 2381 } 2382 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2383 // when changing the audio output device, call addBatteryData to notify 2384 // the change 2385 if ((int)mDevice != value) { 2386 uint32_t params = 0; 2387 // check whether speaker is on 2388 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2389 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2390 } 2391 2392 int deviceWithoutSpeaker 2393 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2394 // check if any other device (except speaker) is on 2395 if (value & deviceWithoutSpeaker ) { 2396 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2397 } 2398 2399 if (params != 0) { 2400 addBatteryData(params); 2401 } 2402 } 2403 2404 // forward device change to effects that have requested to be 2405 // aware of attached audio device. 2406 mDevice = (uint32_t)value; 2407 for (size_t i = 0; i < mEffectChains.size(); i++) { 2408 mEffectChains[i]->setDevice_l(mDevice); 2409 } 2410 } 2411 2412 if (status == NO_ERROR) { 2413 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2414 keyValuePair.string()); 2415 if (!mStandby && status == INVALID_OPERATION) { 2416 mOutput->stream->common.standby(&mOutput->stream->common); 2417 mStandby = true; 2418 mBytesWritten = 0; 2419 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2420 keyValuePair.string()); 2421 } 2422 if (status == NO_ERROR && reconfig) { 2423 delete mAudioMixer; 2424 readOutputParameters(); 2425 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2426 for (size_t i = 0; i < mTracks.size() ; i++) { 2427 int name = getTrackName_l(); 2428 if (name < 0) break; 2429 mTracks[i]->mName = name; 2430 // limit track sample rate to 2 x new output sample rate 2431 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2432 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2433 } 2434 } 2435 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2436 } 2437 } 2438 2439 mNewParameters.removeAt(0); 2440 2441 mParamStatus = status; 2442 mParamCond.signal(); 2443 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2444 // already timed out waiting for the status and will never signal the condition. 2445 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2446 } 2447 return reconfig; 2448} 2449 2450status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2451{ 2452 const size_t SIZE = 256; 2453 char buffer[SIZE]; 2454 String8 result; 2455 2456 PlaybackThread::dumpInternals(fd, args); 2457 2458 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2459 result.append(buffer); 2460 write(fd, result.string(), result.size()); 2461 return NO_ERROR; 2462} 2463 2464uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2465{ 2466 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2467} 2468 2469uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2470{ 2471 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2472} 2473 2474// ---------------------------------------------------------------------------- 2475AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2476 : PlaybackThread(audioFlinger, output, id, device) 2477{ 2478 mType = ThreadBase::DIRECT; 2479} 2480 2481AudioFlinger::DirectOutputThread::~DirectOutputThread() 2482{ 2483} 2484 2485static inline 2486int32_t mul(int16_t in, int16_t v) 2487{ 2488#if defined(__arm__) && !defined(__thumb__) 2489 int32_t out; 2490 asm( "smulbb %[out], %[in], %[v] \n" 2491 : [out]"=r"(out) 2492 : [in]"%r"(in), [v]"r"(v) 2493 : ); 2494 return out; 2495#else 2496 return in * int32_t(v); 2497#endif 2498} 2499 2500void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2501{ 2502 // Do not apply volume on compressed audio 2503 if (!audio_is_linear_pcm(mFormat)) { 2504 return; 2505 } 2506 2507 // convert to signed 16 bit before volume calculation 2508 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2509 size_t count = mFrameCount * mChannelCount; 2510 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2511 int16_t *dst = mMixBuffer + count-1; 2512 while(count--) { 2513 *dst-- = (int16_t)(*src--^0x80) << 8; 2514 } 2515 } 2516 2517 size_t frameCount = mFrameCount; 2518 int16_t *out = mMixBuffer; 2519 if (ramp) { 2520 if (mChannelCount == 1) { 2521 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2522 int32_t vlInc = d / (int32_t)frameCount; 2523 int32_t vl = ((int32_t)mLeftVolShort << 16); 2524 do { 2525 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2526 out++; 2527 vl += vlInc; 2528 } while (--frameCount); 2529 2530 } else { 2531 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2532 int32_t vlInc = d / (int32_t)frameCount; 2533 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2534 int32_t vrInc = d / (int32_t)frameCount; 2535 int32_t vl = ((int32_t)mLeftVolShort << 16); 2536 int32_t vr = ((int32_t)mRightVolShort << 16); 2537 do { 2538 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2539 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2540 out += 2; 2541 vl += vlInc; 2542 vr += vrInc; 2543 } while (--frameCount); 2544 } 2545 } else { 2546 if (mChannelCount == 1) { 2547 do { 2548 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2549 out++; 2550 } while (--frameCount); 2551 } else { 2552 do { 2553 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2554 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2555 out += 2; 2556 } while (--frameCount); 2557 } 2558 } 2559 2560 // convert back to unsigned 8 bit after volume calculation 2561 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2562 size_t count = mFrameCount * mChannelCount; 2563 int16_t *src = mMixBuffer; 2564 uint8_t *dst = (uint8_t *)mMixBuffer; 2565 while(count--) { 2566 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2567 } 2568 } 2569 2570 mLeftVolShort = leftVol; 2571 mRightVolShort = rightVol; 2572} 2573 2574bool AudioFlinger::DirectOutputThread::threadLoop() 2575{ 2576 uint32_t mixerStatus = MIXER_IDLE; 2577 sp<Track> trackToRemove; 2578 sp<Track> activeTrack; 2579 nsecs_t standbyTime = systemTime(); 2580 int8_t *curBuf; 2581 size_t mixBufferSize = mFrameCount*mFrameSize; 2582 uint32_t activeSleepTime = activeSleepTimeUs(); 2583 uint32_t idleSleepTime = idleSleepTimeUs(); 2584 uint32_t sleepTime = idleSleepTime; 2585 // use shorter standby delay as on normal output to release 2586 // hardware resources as soon as possible 2587 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2588 2589 acquireWakeLock(); 2590 2591 while (!exitPending()) 2592 { 2593 bool rampVolume; 2594 uint16_t leftVol; 2595 uint16_t rightVol; 2596 Vector< sp<EffectChain> > effectChains; 2597 2598 processConfigEvents(); 2599 2600 mixerStatus = MIXER_IDLE; 2601 2602 { // scope for the mLock 2603 2604 Mutex::Autolock _l(mLock); 2605 2606 if (checkForNewParameters_l()) { 2607 mixBufferSize = mFrameCount*mFrameSize; 2608 activeSleepTime = activeSleepTimeUs(); 2609 idleSleepTime = idleSleepTimeUs(); 2610 standbyDelay = microseconds(activeSleepTime*2); 2611 } 2612 2613 // put audio hardware into standby after short delay 2614 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2615 mSuspended)) { 2616 // wait until we have something to do... 2617 if (!mStandby) { 2618 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2619 mOutput->stream->common.standby(&mOutput->stream->common); 2620 mStandby = true; 2621 mBytesWritten = 0; 2622 } 2623 2624 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2625 // we're about to wait, flush the binder command buffer 2626 IPCThreadState::self()->flushCommands(); 2627 2628 if (exitPending()) break; 2629 2630 releaseWakeLock_l(); 2631 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2632 mWaitWorkCV.wait(mLock); 2633 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2634 acquireWakeLock_l(); 2635 2636 if (mMasterMute == false) { 2637 char value[PROPERTY_VALUE_MAX]; 2638 property_get("ro.audio.silent", value, "0"); 2639 if (atoi(value)) { 2640 ALOGD("Silence is golden"); 2641 setMasterMute(true); 2642 } 2643 } 2644 2645 standbyTime = systemTime() + standbyDelay; 2646 sleepTime = idleSleepTime; 2647 continue; 2648 } 2649 } 2650 2651 effectChains = mEffectChains; 2652 2653 // find out which tracks need to be processed 2654 if (mActiveTracks.size() != 0) { 2655 sp<Track> t = mActiveTracks[0].promote(); 2656 if (t == 0) continue; 2657 2658 Track* const track = t.get(); 2659 audio_track_cblk_t* cblk = track->cblk(); 2660 2661 // The first time a track is added we wait 2662 // for all its buffers to be filled before processing it 2663 if (cblk->framesReady() && track->isReady() && 2664 !track->isPaused() && !track->isTerminated()) 2665 { 2666 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2667 2668 if (track->mFillingUpStatus == Track::FS_FILLED) { 2669 track->mFillingUpStatus = Track::FS_ACTIVE; 2670 mLeftVolFloat = mRightVolFloat = 0; 2671 mLeftVolShort = mRightVolShort = 0; 2672 if (track->mState == TrackBase::RESUMING) { 2673 track->mState = TrackBase::ACTIVE; 2674 rampVolume = true; 2675 } 2676 } else if (cblk->server != 0) { 2677 // If the track is stopped before the first frame was mixed, 2678 // do not apply ramp 2679 rampVolume = true; 2680 } 2681 // compute volume for this track 2682 float left, right; 2683 if (track->isMuted() || mMasterMute || track->isPausing() || 2684 mStreamTypes[track->type()].mute) { 2685 left = right = 0; 2686 if (track->isPausing()) { 2687 track->setPaused(); 2688 } 2689 } else { 2690 float typeVolume = mStreamTypes[track->type()].volume; 2691 float v = mMasterVolume * typeVolume; 2692 float v_clamped = v * cblk->volume[0]; 2693 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2694 left = v_clamped/MAX_GAIN; 2695 v_clamped = v * cblk->volume[1]; 2696 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2697 right = v_clamped/MAX_GAIN; 2698 } 2699 2700 if (left != mLeftVolFloat || right != mRightVolFloat) { 2701 mLeftVolFloat = left; 2702 mRightVolFloat = right; 2703 2704 // If audio HAL implements volume control, 2705 // force software volume to nominal value 2706 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2707 left = 1.0f; 2708 right = 1.0f; 2709 } 2710 2711 // Convert volumes from float to 8.24 2712 uint32_t vl = (uint32_t)(left * (1 << 24)); 2713 uint32_t vr = (uint32_t)(right * (1 << 24)); 2714 2715 // Delegate volume control to effect in track effect chain if needed 2716 // only one effect chain can be present on DirectOutputThread, so if 2717 // there is one, the track is connected to it 2718 if (!effectChains.isEmpty()) { 2719 // Do not ramp volume if volume is controlled by effect 2720 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2721 rampVolume = false; 2722 } 2723 } 2724 2725 // Convert volumes from 8.24 to 4.12 format 2726 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2727 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2728 leftVol = (uint16_t)v_clamped; 2729 v_clamped = (vr + (1 << 11)) >> 12; 2730 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2731 rightVol = (uint16_t)v_clamped; 2732 } else { 2733 leftVol = mLeftVolShort; 2734 rightVol = mRightVolShort; 2735 rampVolume = false; 2736 } 2737 2738 // reset retry count 2739 track->mRetryCount = kMaxTrackRetriesDirect; 2740 activeTrack = t; 2741 mixerStatus = MIXER_TRACKS_READY; 2742 } else { 2743 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2744 if (track->isStopped()) { 2745 track->reset(); 2746 } 2747 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2748 // We have consumed all the buffers of this track. 2749 // Remove it from the list of active tracks. 2750 trackToRemove = track; 2751 } else { 2752 // No buffers for this track. Give it a few chances to 2753 // fill a buffer, then remove it from active list. 2754 if (--(track->mRetryCount) <= 0) { 2755 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2756 trackToRemove = track; 2757 } else { 2758 mixerStatus = MIXER_TRACKS_ENABLED; 2759 } 2760 } 2761 } 2762 } 2763 2764 // remove all the tracks that need to be... 2765 if (CC_UNLIKELY(trackToRemove != 0)) { 2766 mActiveTracks.remove(trackToRemove); 2767 if (!effectChains.isEmpty()) { 2768 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2769 trackToRemove->sessionId()); 2770 effectChains[0]->decActiveTrackCnt(); 2771 } 2772 if (trackToRemove->isTerminated()) { 2773 removeTrack_l(trackToRemove); 2774 } 2775 } 2776 2777 lockEffectChains_l(effectChains); 2778 } 2779 2780 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2781 AudioBufferProvider::Buffer buffer; 2782 size_t frameCount = mFrameCount; 2783 curBuf = (int8_t *)mMixBuffer; 2784 // output audio to hardware 2785 while (frameCount) { 2786 buffer.frameCount = frameCount; 2787 activeTrack->getNextBuffer(&buffer); 2788 if (CC_UNLIKELY(buffer.raw == NULL)) { 2789 memset(curBuf, 0, frameCount * mFrameSize); 2790 break; 2791 } 2792 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2793 frameCount -= buffer.frameCount; 2794 curBuf += buffer.frameCount * mFrameSize; 2795 activeTrack->releaseBuffer(&buffer); 2796 } 2797 sleepTime = 0; 2798 standbyTime = systemTime() + standbyDelay; 2799 } else { 2800 if (sleepTime == 0) { 2801 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2802 sleepTime = activeSleepTime; 2803 } else { 2804 sleepTime = idleSleepTime; 2805 } 2806 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2807 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2808 sleepTime = 0; 2809 } 2810 } 2811 2812 if (mSuspended) { 2813 sleepTime = suspendSleepTimeUs(); 2814 } 2815 // sleepTime == 0 means we must write to audio hardware 2816 if (sleepTime == 0) { 2817 if (mixerStatus == MIXER_TRACKS_READY) { 2818 applyVolume(leftVol, rightVol, rampVolume); 2819 } 2820 for (size_t i = 0; i < effectChains.size(); i ++) { 2821 effectChains[i]->process_l(); 2822 } 2823 unlockEffectChains(effectChains); 2824 2825 mLastWriteTime = systemTime(); 2826 mInWrite = true; 2827 mBytesWritten += mixBufferSize; 2828 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2829 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2830 mNumWrites++; 2831 mInWrite = false; 2832 mStandby = false; 2833 } else { 2834 unlockEffectChains(effectChains); 2835 usleep(sleepTime); 2836 } 2837 2838 // finally let go of removed track, without the lock held 2839 // since we can't guarantee the destructors won't acquire that 2840 // same lock. 2841 trackToRemove.clear(); 2842 activeTrack.clear(); 2843 2844 // Effect chains will be actually deleted here if they were removed from 2845 // mEffectChains list during mixing or effects processing 2846 effectChains.clear(); 2847 } 2848 2849 if (!mStandby) { 2850 mOutput->stream->common.standby(&mOutput->stream->common); 2851 } 2852 2853 releaseWakeLock(); 2854 2855 ALOGV("DirectOutputThread %p exiting", this); 2856 return false; 2857} 2858 2859// getTrackName_l() must be called with ThreadBase::mLock held 2860int AudioFlinger::DirectOutputThread::getTrackName_l() 2861{ 2862 return 0; 2863} 2864 2865// deleteTrackName_l() must be called with ThreadBase::mLock held 2866void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2867{ 2868} 2869 2870// checkForNewParameters_l() must be called with ThreadBase::mLock held 2871bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2872{ 2873 bool reconfig = false; 2874 2875 while (!mNewParameters.isEmpty()) { 2876 status_t status = NO_ERROR; 2877 String8 keyValuePair = mNewParameters[0]; 2878 AudioParameter param = AudioParameter(keyValuePair); 2879 int value; 2880 2881 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2882 // do not accept frame count changes if tracks are open as the track buffer 2883 // size depends on frame count and correct behavior would not be garantied 2884 // if frame count is changed after track creation 2885 if (!mTracks.isEmpty()) { 2886 status = INVALID_OPERATION; 2887 } else { 2888 reconfig = true; 2889 } 2890 } 2891 if (status == NO_ERROR) { 2892 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2893 keyValuePair.string()); 2894 if (!mStandby && status == INVALID_OPERATION) { 2895 mOutput->stream->common.standby(&mOutput->stream->common); 2896 mStandby = true; 2897 mBytesWritten = 0; 2898 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2899 keyValuePair.string()); 2900 } 2901 if (status == NO_ERROR && reconfig) { 2902 readOutputParameters(); 2903 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2904 } 2905 } 2906 2907 mNewParameters.removeAt(0); 2908 2909 mParamStatus = status; 2910 mParamCond.signal(); 2911 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2912 // already timed out waiting for the status and will never signal the condition. 2913 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2914 } 2915 return reconfig; 2916} 2917 2918uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2919{ 2920 uint32_t time; 2921 if (audio_is_linear_pcm(mFormat)) { 2922 time = PlaybackThread::activeSleepTimeUs(); 2923 } else { 2924 time = 10000; 2925 } 2926 return time; 2927} 2928 2929uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2930{ 2931 uint32_t time; 2932 if (audio_is_linear_pcm(mFormat)) { 2933 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2934 } else { 2935 time = 10000; 2936 } 2937 return time; 2938} 2939 2940uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2941{ 2942 uint32_t time; 2943 if (audio_is_linear_pcm(mFormat)) { 2944 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2945 } else { 2946 time = 10000; 2947 } 2948 return time; 2949} 2950 2951 2952// ---------------------------------------------------------------------------- 2953 2954AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2955 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2956{ 2957 mType = ThreadBase::DUPLICATING; 2958 addOutputTrack(mainThread); 2959} 2960 2961AudioFlinger::DuplicatingThread::~DuplicatingThread() 2962{ 2963 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2964 mOutputTracks[i]->destroy(); 2965 } 2966 mOutputTracks.clear(); 2967} 2968 2969bool AudioFlinger::DuplicatingThread::threadLoop() 2970{ 2971 Vector< sp<Track> > tracksToRemove; 2972 uint32_t mixerStatus = MIXER_IDLE; 2973 nsecs_t standbyTime = systemTime(); 2974 size_t mixBufferSize = mFrameCount*mFrameSize; 2975 SortedVector< sp<OutputTrack> > outputTracks; 2976 uint32_t writeFrames = 0; 2977 uint32_t activeSleepTime = activeSleepTimeUs(); 2978 uint32_t idleSleepTime = idleSleepTimeUs(); 2979 uint32_t sleepTime = idleSleepTime; 2980 Vector< sp<EffectChain> > effectChains; 2981 2982 acquireWakeLock(); 2983 2984 while (!exitPending()) 2985 { 2986 processConfigEvents(); 2987 2988 mixerStatus = MIXER_IDLE; 2989 { // scope for the mLock 2990 2991 Mutex::Autolock _l(mLock); 2992 2993 if (checkForNewParameters_l()) { 2994 mixBufferSize = mFrameCount*mFrameSize; 2995 updateWaitTime(); 2996 activeSleepTime = activeSleepTimeUs(); 2997 idleSleepTime = idleSleepTimeUs(); 2998 } 2999 3000 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3001 3002 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3003 outputTracks.add(mOutputTracks[i]); 3004 } 3005 3006 // put audio hardware into standby after short delay 3007 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3008 mSuspended)) { 3009 if (!mStandby) { 3010 for (size_t i = 0; i < outputTracks.size(); i++) { 3011 outputTracks[i]->stop(); 3012 } 3013 mStandby = true; 3014 mBytesWritten = 0; 3015 } 3016 3017 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3018 // we're about to wait, flush the binder command buffer 3019 IPCThreadState::self()->flushCommands(); 3020 outputTracks.clear(); 3021 3022 if (exitPending()) break; 3023 3024 releaseWakeLock_l(); 3025 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3026 mWaitWorkCV.wait(mLock); 3027 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3028 acquireWakeLock_l(); 3029 3030 if (mMasterMute == false) { 3031 char value[PROPERTY_VALUE_MAX]; 3032 property_get("ro.audio.silent", value, "0"); 3033 if (atoi(value)) { 3034 ALOGD("Silence is golden"); 3035 setMasterMute(true); 3036 } 3037 } 3038 3039 standbyTime = systemTime() + kStandbyTimeInNsecs; 3040 sleepTime = idleSleepTime; 3041 continue; 3042 } 3043 } 3044 3045 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3046 3047 // prevent any changes in effect chain list and in each effect chain 3048 // during mixing and effect process as the audio buffers could be deleted 3049 // or modified if an effect is created or deleted 3050 lockEffectChains_l(effectChains); 3051 } 3052 3053 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3054 // mix buffers... 3055 if (outputsReady(outputTracks)) { 3056 mAudioMixer->process(); 3057 } else { 3058 memset(mMixBuffer, 0, mixBufferSize); 3059 } 3060 sleepTime = 0; 3061 writeFrames = mFrameCount; 3062 } else { 3063 if (sleepTime == 0) { 3064 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3065 sleepTime = activeSleepTime; 3066 } else { 3067 sleepTime = idleSleepTime; 3068 } 3069 } else if (mBytesWritten != 0) { 3070 // flush remaining overflow buffers in output tracks 3071 for (size_t i = 0; i < outputTracks.size(); i++) { 3072 if (outputTracks[i]->isActive()) { 3073 sleepTime = 0; 3074 writeFrames = 0; 3075 memset(mMixBuffer, 0, mixBufferSize); 3076 break; 3077 } 3078 } 3079 } 3080 } 3081 3082 if (mSuspended) { 3083 sleepTime = suspendSleepTimeUs(); 3084 } 3085 // sleepTime == 0 means we must write to audio hardware 3086 if (sleepTime == 0) { 3087 for (size_t i = 0; i < effectChains.size(); i ++) { 3088 effectChains[i]->process_l(); 3089 } 3090 // enable changes in effect chain 3091 unlockEffectChains(effectChains); 3092 3093 standbyTime = systemTime() + kStandbyTimeInNsecs; 3094 for (size_t i = 0; i < outputTracks.size(); i++) { 3095 outputTracks[i]->write(mMixBuffer, writeFrames); 3096 } 3097 mStandby = false; 3098 mBytesWritten += mixBufferSize; 3099 } else { 3100 // enable changes in effect chain 3101 unlockEffectChains(effectChains); 3102 usleep(sleepTime); 3103 } 3104 3105 // finally let go of all our tracks, without the lock held 3106 // since we can't guarantee the destructors won't acquire that 3107 // same lock. 3108 tracksToRemove.clear(); 3109 outputTracks.clear(); 3110 3111 // Effect chains will be actually deleted here if they were removed from 3112 // mEffectChains list during mixing or effects processing 3113 effectChains.clear(); 3114 } 3115 3116 releaseWakeLock(); 3117 3118 return false; 3119} 3120 3121void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3122{ 3123 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3124 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3125 this, 3126 mSampleRate, 3127 mFormat, 3128 mChannelMask, 3129 frameCount); 3130 if (outputTrack->cblk() != NULL) { 3131 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3132 mOutputTracks.add(outputTrack); 3133 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3134 updateWaitTime(); 3135 } 3136} 3137 3138void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3139{ 3140 Mutex::Autolock _l(mLock); 3141 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3142 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3143 mOutputTracks[i]->destroy(); 3144 mOutputTracks.removeAt(i); 3145 updateWaitTime(); 3146 return; 3147 } 3148 } 3149 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3150} 3151 3152void AudioFlinger::DuplicatingThread::updateWaitTime() 3153{ 3154 mWaitTimeMs = UINT_MAX; 3155 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3156 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3157 if (strong != NULL) { 3158 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3159 if (waitTimeMs < mWaitTimeMs) { 3160 mWaitTimeMs = waitTimeMs; 3161 } 3162 } 3163 } 3164} 3165 3166 3167bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3168{ 3169 for (size_t i = 0; i < outputTracks.size(); i++) { 3170 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3171 if (thread == 0) { 3172 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3173 return false; 3174 } 3175 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3176 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3177 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3178 return false; 3179 } 3180 } 3181 return true; 3182} 3183 3184uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3185{ 3186 return (mWaitTimeMs * 1000) / 2; 3187} 3188 3189// ---------------------------------------------------------------------------- 3190 3191// TrackBase constructor must be called with AudioFlinger::mLock held 3192AudioFlinger::ThreadBase::TrackBase::TrackBase( 3193 const wp<ThreadBase>& thread, 3194 const sp<Client>& client, 3195 uint32_t sampleRate, 3196 uint32_t format, 3197 uint32_t channelMask, 3198 int frameCount, 3199 uint32_t flags, 3200 const sp<IMemory>& sharedBuffer, 3201 int sessionId) 3202 : RefBase(), 3203 mThread(thread), 3204 mClient(client), 3205 mCblk(0), 3206 mFrameCount(0), 3207 mState(IDLE), 3208 mClientTid(-1), 3209 mFormat(format), 3210 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3211 mSessionId(sessionId) 3212{ 3213 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3214 3215 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3216 size_t size = sizeof(audio_track_cblk_t); 3217 uint8_t channelCount = popcount(channelMask); 3218 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3219 if (sharedBuffer == 0) { 3220 size += bufferSize; 3221 } 3222 3223 if (client != NULL) { 3224 mCblkMemory = client->heap()->allocate(size); 3225 if (mCblkMemory != 0) { 3226 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3227 if (mCblk) { // construct the shared structure in-place. 3228 new(mCblk) audio_track_cblk_t(); 3229 // clear all buffers 3230 mCblk->frameCount = frameCount; 3231 mCblk->sampleRate = sampleRate; 3232 mChannelCount = channelCount; 3233 mChannelMask = channelMask; 3234 if (sharedBuffer == 0) { 3235 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3236 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3237 // Force underrun condition to avoid false underrun callback until first data is 3238 // written to buffer (other flags are cleared) 3239 mCblk->flags = CBLK_UNDERRUN_ON; 3240 } else { 3241 mBuffer = sharedBuffer->pointer(); 3242 } 3243 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3244 } 3245 } else { 3246 ALOGE("not enough memory for AudioTrack size=%u", size); 3247 client->heap()->dump("AudioTrack"); 3248 return; 3249 } 3250 } else { 3251 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3252 // construct the shared structure in-place. 3253 new(mCblk) audio_track_cblk_t(); 3254 // clear all buffers 3255 mCblk->frameCount = frameCount; 3256 mCblk->sampleRate = sampleRate; 3257 mChannelCount = channelCount; 3258 mChannelMask = channelMask; 3259 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3260 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3261 // Force underrun condition to avoid false underrun callback until first data is 3262 // written to buffer (other flags are cleared) 3263 mCblk->flags = CBLK_UNDERRUN_ON; 3264 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3265 } 3266} 3267 3268AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3269{ 3270 if (mCblk) { 3271 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3272 if (mClient == NULL) { 3273 delete mCblk; 3274 } 3275 } 3276 mCblkMemory.clear(); // and free the shared memory 3277 if (mClient != NULL) { 3278 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3279 mClient.clear(); 3280 } 3281} 3282 3283void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3284{ 3285 buffer->raw = NULL; 3286 mFrameCount = buffer->frameCount; 3287 step(); 3288 buffer->frameCount = 0; 3289} 3290 3291bool AudioFlinger::ThreadBase::TrackBase::step() { 3292 bool result; 3293 audio_track_cblk_t* cblk = this->cblk(); 3294 3295 result = cblk->stepServer(mFrameCount); 3296 if (!result) { 3297 ALOGV("stepServer failed acquiring cblk mutex"); 3298 mFlags |= STEPSERVER_FAILED; 3299 } 3300 return result; 3301} 3302 3303void AudioFlinger::ThreadBase::TrackBase::reset() { 3304 audio_track_cblk_t* cblk = this->cblk(); 3305 3306 cblk->user = 0; 3307 cblk->server = 0; 3308 cblk->userBase = 0; 3309 cblk->serverBase = 0; 3310 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3311 ALOGV("TrackBase::reset"); 3312} 3313 3314sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3315{ 3316 return mCblkMemory; 3317} 3318 3319int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3320 return (int)mCblk->sampleRate; 3321} 3322 3323int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3324 return (const int)mChannelCount; 3325} 3326 3327uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3328 return mChannelMask; 3329} 3330 3331void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3332 audio_track_cblk_t* cblk = this->cblk(); 3333 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 3334 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 3335 3336 // Check validity of returned pointer in case the track control block would have been corrupted. 3337 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3338 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 3339 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3340 server %d, serverBase %d, user %d, userBase %d", 3341 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3342 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3343 return 0; 3344 } 3345 3346 return bufferStart; 3347} 3348 3349// ---------------------------------------------------------------------------- 3350 3351// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3352AudioFlinger::PlaybackThread::Track::Track( 3353 const wp<ThreadBase>& thread, 3354 const sp<Client>& client, 3355 int streamType, 3356 uint32_t sampleRate, 3357 uint32_t format, 3358 uint32_t channelMask, 3359 int frameCount, 3360 const sp<IMemory>& sharedBuffer, 3361 int sessionId) 3362 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3363 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3364 mAuxEffectId(0), mHasVolumeController(false) 3365{ 3366 if (mCblk != NULL) { 3367 sp<ThreadBase> baseThread = thread.promote(); 3368 if (baseThread != 0) { 3369 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3370 mName = playbackThread->getTrackName_l(); 3371 mMainBuffer = playbackThread->mixBuffer(); 3372 } 3373 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3374 if (mName < 0) { 3375 ALOGE("no more track names available"); 3376 } 3377 mVolume[0] = 1.0f; 3378 mVolume[1] = 1.0f; 3379 mStreamType = streamType; 3380 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3381 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3382 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3383 } 3384} 3385 3386AudioFlinger::PlaybackThread::Track::~Track() 3387{ 3388 ALOGV("PlaybackThread::Track destructor"); 3389 sp<ThreadBase> thread = mThread.promote(); 3390 if (thread != 0) { 3391 Mutex::Autolock _l(thread->mLock); 3392 mState = TERMINATED; 3393 } 3394} 3395 3396void AudioFlinger::PlaybackThread::Track::destroy() 3397{ 3398 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3399 // by removing it from mTracks vector, so there is a risk that this Tracks's 3400 // desctructor is called. As the destructor needs to lock mLock, 3401 // we must acquire a strong reference on this Track before locking mLock 3402 // here so that the destructor is called only when exiting this function. 3403 // On the other hand, as long as Track::destroy() is only called by 3404 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3405 // this Track with its member mTrack. 3406 sp<Track> keep(this); 3407 { // scope for mLock 3408 sp<ThreadBase> thread = mThread.promote(); 3409 if (thread != 0) { 3410 if (!isOutputTrack()) { 3411 if (mState == ACTIVE || mState == RESUMING) { 3412 AudioSystem::stopOutput(thread->id(), 3413 (audio_stream_type_t)mStreamType, 3414 mSessionId); 3415 3416 // to track the speaker usage 3417 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3418 } 3419 AudioSystem::releaseOutput(thread->id()); 3420 } 3421 Mutex::Autolock _l(thread->mLock); 3422 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3423 playbackThread->destroyTrack_l(this); 3424 } 3425 } 3426} 3427 3428void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3429{ 3430 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", 3431 mName - AudioMixer::TRACK0, 3432 (mClient == NULL) ? getpid() : mClient->pid(), 3433 mStreamType, 3434 mFormat, 3435 mChannelMask, 3436 mSessionId, 3437 mFrameCount, 3438 mState, 3439 mMute, 3440 mFillingUpStatus, 3441 mCblk->sampleRate, 3442 mCblk->volume[0], 3443 mCblk->volume[1], 3444 mCblk->server, 3445 mCblk->user, 3446 (int)mMainBuffer, 3447 (int)mAuxBuffer); 3448} 3449 3450status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3451{ 3452 audio_track_cblk_t* cblk = this->cblk(); 3453 uint32_t framesReady; 3454 uint32_t framesReq = buffer->frameCount; 3455 3456 // Check if last stepServer failed, try to step now 3457 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3458 if (!step()) goto getNextBuffer_exit; 3459 ALOGV("stepServer recovered"); 3460 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3461 } 3462 3463 framesReady = cblk->framesReady(); 3464 3465 if (CC_LIKELY(framesReady)) { 3466 uint32_t s = cblk->server; 3467 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3468 3469 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3470 if (framesReq > framesReady) { 3471 framesReq = framesReady; 3472 } 3473 if (s + framesReq > bufferEnd) { 3474 framesReq = bufferEnd - s; 3475 } 3476 3477 buffer->raw = getBuffer(s, framesReq); 3478 if (buffer->raw == NULL) goto getNextBuffer_exit; 3479 3480 buffer->frameCount = framesReq; 3481 return NO_ERROR; 3482 } 3483 3484getNextBuffer_exit: 3485 buffer->raw = NULL; 3486 buffer->frameCount = 0; 3487 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3488 return NOT_ENOUGH_DATA; 3489} 3490 3491bool AudioFlinger::PlaybackThread::Track::isReady() const { 3492 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3493 3494 if (mCblk->framesReady() >= mCblk->frameCount || 3495 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3496 mFillingUpStatus = FS_FILLED; 3497 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3498 return true; 3499 } 3500 return false; 3501} 3502 3503status_t AudioFlinger::PlaybackThread::Track::start() 3504{ 3505 status_t status = NO_ERROR; 3506 ALOGV("start(%d), calling thread %d session %d", 3507 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3508 sp<ThreadBase> thread = mThread.promote(); 3509 if (thread != 0) { 3510 Mutex::Autolock _l(thread->mLock); 3511 int state = mState; 3512 // here the track could be either new, or restarted 3513 // in both cases "unstop" the track 3514 if (mState == PAUSED) { 3515 mState = TrackBase::RESUMING; 3516 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3517 } else { 3518 mState = TrackBase::ACTIVE; 3519 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3520 } 3521 3522 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3523 thread->mLock.unlock(); 3524 status = AudioSystem::startOutput(thread->id(), 3525 (audio_stream_type_t)mStreamType, 3526 mSessionId); 3527 thread->mLock.lock(); 3528 3529 // to track the speaker usage 3530 if (status == NO_ERROR) { 3531 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3532 } 3533 } 3534 if (status == NO_ERROR) { 3535 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3536 playbackThread->addTrack_l(this); 3537 } else { 3538 mState = state; 3539 } 3540 } else { 3541 status = BAD_VALUE; 3542 } 3543 return status; 3544} 3545 3546void AudioFlinger::PlaybackThread::Track::stop() 3547{ 3548 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3549 sp<ThreadBase> thread = mThread.promote(); 3550 if (thread != 0) { 3551 Mutex::Autolock _l(thread->mLock); 3552 int state = mState; 3553 if (mState > STOPPED) { 3554 mState = STOPPED; 3555 // If the track is not active (PAUSED and buffers full), flush buffers 3556 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3557 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3558 reset(); 3559 } 3560 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3561 } 3562 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3563 thread->mLock.unlock(); 3564 AudioSystem::stopOutput(thread->id(), 3565 (audio_stream_type_t)mStreamType, 3566 mSessionId); 3567 thread->mLock.lock(); 3568 3569 // to track the speaker usage 3570 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3571 } 3572 } 3573} 3574 3575void AudioFlinger::PlaybackThread::Track::pause() 3576{ 3577 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3578 sp<ThreadBase> thread = mThread.promote(); 3579 if (thread != 0) { 3580 Mutex::Autolock _l(thread->mLock); 3581 if (mState == ACTIVE || mState == RESUMING) { 3582 mState = PAUSING; 3583 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3584 if (!isOutputTrack()) { 3585 thread->mLock.unlock(); 3586 AudioSystem::stopOutput(thread->id(), 3587 (audio_stream_type_t)mStreamType, 3588 mSessionId); 3589 thread->mLock.lock(); 3590 3591 // to track the speaker usage 3592 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3593 } 3594 } 3595 } 3596} 3597 3598void AudioFlinger::PlaybackThread::Track::flush() 3599{ 3600 ALOGV("flush(%d)", mName); 3601 sp<ThreadBase> thread = mThread.promote(); 3602 if (thread != 0) { 3603 Mutex::Autolock _l(thread->mLock); 3604 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3605 return; 3606 } 3607 // No point remaining in PAUSED state after a flush => go to 3608 // STOPPED state 3609 mState = STOPPED; 3610 3611 // do not reset the track if it is still in the process of being stopped or paused. 3612 // this will be done by prepareTracks_l() when the track is stopped. 3613 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3614 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3615 reset(); 3616 } 3617 } 3618} 3619 3620void AudioFlinger::PlaybackThread::Track::reset() 3621{ 3622 // Do not reset twice to avoid discarding data written just after a flush and before 3623 // the audioflinger thread detects the track is stopped. 3624 if (!mResetDone) { 3625 TrackBase::reset(); 3626 // Force underrun condition to avoid false underrun callback until first data is 3627 // written to buffer 3628 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3629 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3630 mFillingUpStatus = FS_FILLING; 3631 mResetDone = true; 3632 } 3633} 3634 3635void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3636{ 3637 mMute = muted; 3638} 3639 3640void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3641{ 3642 mVolume[0] = left; 3643 mVolume[1] = right; 3644} 3645 3646status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3647{ 3648 status_t status = DEAD_OBJECT; 3649 sp<ThreadBase> thread = mThread.promote(); 3650 if (thread != 0) { 3651 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3652 status = playbackThread->attachAuxEffect(this, EffectId); 3653 } 3654 return status; 3655} 3656 3657void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3658{ 3659 mAuxEffectId = EffectId; 3660 mAuxBuffer = buffer; 3661} 3662 3663// ---------------------------------------------------------------------------- 3664 3665// RecordTrack constructor must be called with AudioFlinger::mLock held 3666AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3667 const wp<ThreadBase>& thread, 3668 const sp<Client>& client, 3669 uint32_t sampleRate, 3670 uint32_t format, 3671 uint32_t channelMask, 3672 int frameCount, 3673 uint32_t flags, 3674 int sessionId) 3675 : TrackBase(thread, client, sampleRate, format, 3676 channelMask, frameCount, flags, 0, sessionId), 3677 mOverflow(false) 3678{ 3679 if (mCblk != NULL) { 3680 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3681 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3682 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3683 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3684 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3685 } else { 3686 mCblk->frameSize = sizeof(int8_t); 3687 } 3688 } 3689} 3690 3691AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3692{ 3693 sp<ThreadBase> thread = mThread.promote(); 3694 if (thread != 0) { 3695 AudioSystem::releaseInput(thread->id()); 3696 } 3697} 3698 3699status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3700{ 3701 audio_track_cblk_t* cblk = this->cblk(); 3702 uint32_t framesAvail; 3703 uint32_t framesReq = buffer->frameCount; 3704 3705 // Check if last stepServer failed, try to step now 3706 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3707 if (!step()) goto getNextBuffer_exit; 3708 ALOGV("stepServer recovered"); 3709 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3710 } 3711 3712 framesAvail = cblk->framesAvailable_l(); 3713 3714 if (CC_LIKELY(framesAvail)) { 3715 uint32_t s = cblk->server; 3716 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3717 3718 if (framesReq > framesAvail) { 3719 framesReq = framesAvail; 3720 } 3721 if (s + framesReq > bufferEnd) { 3722 framesReq = bufferEnd - s; 3723 } 3724 3725 buffer->raw = getBuffer(s, framesReq); 3726 if (buffer->raw == NULL) goto getNextBuffer_exit; 3727 3728 buffer->frameCount = framesReq; 3729 return NO_ERROR; 3730 } 3731 3732getNextBuffer_exit: 3733 buffer->raw = NULL; 3734 buffer->frameCount = 0; 3735 return NOT_ENOUGH_DATA; 3736} 3737 3738status_t AudioFlinger::RecordThread::RecordTrack::start() 3739{ 3740 sp<ThreadBase> thread = mThread.promote(); 3741 if (thread != 0) { 3742 RecordThread *recordThread = (RecordThread *)thread.get(); 3743 return recordThread->start(this); 3744 } else { 3745 return BAD_VALUE; 3746 } 3747} 3748 3749void AudioFlinger::RecordThread::RecordTrack::stop() 3750{ 3751 sp<ThreadBase> thread = mThread.promote(); 3752 if (thread != 0) { 3753 RecordThread *recordThread = (RecordThread *)thread.get(); 3754 recordThread->stop(this); 3755 TrackBase::reset(); 3756 // Force overerrun condition to avoid false overrun callback until first data is 3757 // read from buffer 3758 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3759 } 3760} 3761 3762void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3763{ 3764 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3765 (mClient == NULL) ? getpid() : mClient->pid(), 3766 mFormat, 3767 mChannelMask, 3768 mSessionId, 3769 mFrameCount, 3770 mState, 3771 mCblk->sampleRate, 3772 mCblk->server, 3773 mCblk->user); 3774} 3775 3776 3777// ---------------------------------------------------------------------------- 3778 3779AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3780 const wp<ThreadBase>& thread, 3781 DuplicatingThread *sourceThread, 3782 uint32_t sampleRate, 3783 uint32_t format, 3784 uint32_t channelMask, 3785 int frameCount) 3786 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3787 mActive(false), mSourceThread(sourceThread) 3788{ 3789 3790 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3791 if (mCblk != NULL) { 3792 mCblk->flags |= CBLK_DIRECTION_OUT; 3793 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3794 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3795 mOutBuffer.frameCount = 0; 3796 playbackThread->mTracks.add(this); 3797 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3798 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3799 mCblk, mBuffer, mCblk->buffers, 3800 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3801 } else { 3802 ALOGW("Error creating output track on thread %p", playbackThread); 3803 } 3804} 3805 3806AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3807{ 3808 clearBufferQueue(); 3809} 3810 3811status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3812{ 3813 status_t status = Track::start(); 3814 if (status != NO_ERROR) { 3815 return status; 3816 } 3817 3818 mActive = true; 3819 mRetryCount = 127; 3820 return status; 3821} 3822 3823void AudioFlinger::PlaybackThread::OutputTrack::stop() 3824{ 3825 Track::stop(); 3826 clearBufferQueue(); 3827 mOutBuffer.frameCount = 0; 3828 mActive = false; 3829} 3830 3831bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3832{ 3833 Buffer *pInBuffer; 3834 Buffer inBuffer; 3835 uint32_t channelCount = mChannelCount; 3836 bool outputBufferFull = false; 3837 inBuffer.frameCount = frames; 3838 inBuffer.i16 = data; 3839 3840 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3841 3842 if (!mActive && frames != 0) { 3843 start(); 3844 sp<ThreadBase> thread = mThread.promote(); 3845 if (thread != 0) { 3846 MixerThread *mixerThread = (MixerThread *)thread.get(); 3847 if (mCblk->frameCount > frames){ 3848 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3849 uint32_t startFrames = (mCblk->frameCount - frames); 3850 pInBuffer = new Buffer; 3851 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3852 pInBuffer->frameCount = startFrames; 3853 pInBuffer->i16 = pInBuffer->mBuffer; 3854 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3855 mBufferQueue.add(pInBuffer); 3856 } else { 3857 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3858 } 3859 } 3860 } 3861 } 3862 3863 while (waitTimeLeftMs) { 3864 // First write pending buffers, then new data 3865 if (mBufferQueue.size()) { 3866 pInBuffer = mBufferQueue.itemAt(0); 3867 } else { 3868 pInBuffer = &inBuffer; 3869 } 3870 3871 if (pInBuffer->frameCount == 0) { 3872 break; 3873 } 3874 3875 if (mOutBuffer.frameCount == 0) { 3876 mOutBuffer.frameCount = pInBuffer->frameCount; 3877 nsecs_t startTime = systemTime(); 3878 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3879 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3880 outputBufferFull = true; 3881 break; 3882 } 3883 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3884 if (waitTimeLeftMs >= waitTimeMs) { 3885 waitTimeLeftMs -= waitTimeMs; 3886 } else { 3887 waitTimeLeftMs = 0; 3888 } 3889 } 3890 3891 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3892 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3893 mCblk->stepUser(outFrames); 3894 pInBuffer->frameCount -= outFrames; 3895 pInBuffer->i16 += outFrames * channelCount; 3896 mOutBuffer.frameCount -= outFrames; 3897 mOutBuffer.i16 += outFrames * channelCount; 3898 3899 if (pInBuffer->frameCount == 0) { 3900 if (mBufferQueue.size()) { 3901 mBufferQueue.removeAt(0); 3902 delete [] pInBuffer->mBuffer; 3903 delete pInBuffer; 3904 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3905 } else { 3906 break; 3907 } 3908 } 3909 } 3910 3911 // If we could not write all frames, allocate a buffer and queue it for next time. 3912 if (inBuffer.frameCount) { 3913 sp<ThreadBase> thread = mThread.promote(); 3914 if (thread != 0 && !thread->standby()) { 3915 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3916 pInBuffer = new Buffer; 3917 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3918 pInBuffer->frameCount = inBuffer.frameCount; 3919 pInBuffer->i16 = pInBuffer->mBuffer; 3920 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3921 mBufferQueue.add(pInBuffer); 3922 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3923 } else { 3924 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3925 } 3926 } 3927 } 3928 3929 // Calling write() with a 0 length buffer, means that no more data will be written: 3930 // If no more buffers are pending, fill output track buffer to make sure it is started 3931 // by output mixer. 3932 if (frames == 0 && mBufferQueue.size() == 0) { 3933 if (mCblk->user < mCblk->frameCount) { 3934 frames = mCblk->frameCount - mCblk->user; 3935 pInBuffer = new Buffer; 3936 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3937 pInBuffer->frameCount = frames; 3938 pInBuffer->i16 = pInBuffer->mBuffer; 3939 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3940 mBufferQueue.add(pInBuffer); 3941 } else if (mActive) { 3942 stop(); 3943 } 3944 } 3945 3946 return outputBufferFull; 3947} 3948 3949status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3950{ 3951 int active; 3952 status_t result; 3953 audio_track_cblk_t* cblk = mCblk; 3954 uint32_t framesReq = buffer->frameCount; 3955 3956// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3957 buffer->frameCount = 0; 3958 3959 uint32_t framesAvail = cblk->framesAvailable(); 3960 3961 3962 if (framesAvail == 0) { 3963 Mutex::Autolock _l(cblk->lock); 3964 goto start_loop_here; 3965 while (framesAvail == 0) { 3966 active = mActive; 3967 if (CC_UNLIKELY(!active)) { 3968 ALOGV("Not active and NO_MORE_BUFFERS"); 3969 return AudioTrack::NO_MORE_BUFFERS; 3970 } 3971 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3972 if (result != NO_ERROR) { 3973 return AudioTrack::NO_MORE_BUFFERS; 3974 } 3975 // read the server count again 3976 start_loop_here: 3977 framesAvail = cblk->framesAvailable_l(); 3978 } 3979 } 3980 3981// if (framesAvail < framesReq) { 3982// return AudioTrack::NO_MORE_BUFFERS; 3983// } 3984 3985 if (framesReq > framesAvail) { 3986 framesReq = framesAvail; 3987 } 3988 3989 uint32_t u = cblk->user; 3990 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3991 3992 if (u + framesReq > bufferEnd) { 3993 framesReq = bufferEnd - u; 3994 } 3995 3996 buffer->frameCount = framesReq; 3997 buffer->raw = (void *)cblk->buffer(u); 3998 return NO_ERROR; 3999} 4000 4001 4002void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4003{ 4004 size_t size = mBufferQueue.size(); 4005 Buffer *pBuffer; 4006 4007 for (size_t i = 0; i < size; i++) { 4008 pBuffer = mBufferQueue.itemAt(i); 4009 delete [] pBuffer->mBuffer; 4010 delete pBuffer; 4011 } 4012 mBufferQueue.clear(); 4013} 4014 4015// ---------------------------------------------------------------------------- 4016 4017AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4018 : RefBase(), 4019 mAudioFlinger(audioFlinger), 4020 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4021 mPid(pid) 4022{ 4023 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4024} 4025 4026// Client destructor must be called with AudioFlinger::mLock held 4027AudioFlinger::Client::~Client() 4028{ 4029 mAudioFlinger->removeClient_l(mPid); 4030} 4031 4032const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4033{ 4034 return mMemoryDealer; 4035} 4036 4037// ---------------------------------------------------------------------------- 4038 4039AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4040 const sp<IAudioFlingerClient>& client, 4041 pid_t pid) 4042 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4043{ 4044} 4045 4046AudioFlinger::NotificationClient::~NotificationClient() 4047{ 4048 mClient.clear(); 4049} 4050 4051void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4052{ 4053 sp<NotificationClient> keep(this); 4054 { 4055 mAudioFlinger->removeNotificationClient(mPid); 4056 } 4057} 4058 4059// ---------------------------------------------------------------------------- 4060 4061AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4062 : BnAudioTrack(), 4063 mTrack(track) 4064{ 4065} 4066 4067AudioFlinger::TrackHandle::~TrackHandle() { 4068 // just stop the track on deletion, associated resources 4069 // will be freed from the main thread once all pending buffers have 4070 // been played. Unless it's not in the active track list, in which 4071 // case we free everything now... 4072 mTrack->destroy(); 4073} 4074 4075status_t AudioFlinger::TrackHandle::start() { 4076 return mTrack->start(); 4077} 4078 4079void AudioFlinger::TrackHandle::stop() { 4080 mTrack->stop(); 4081} 4082 4083void AudioFlinger::TrackHandle::flush() { 4084 mTrack->flush(); 4085} 4086 4087void AudioFlinger::TrackHandle::mute(bool e) { 4088 mTrack->mute(e); 4089} 4090 4091void AudioFlinger::TrackHandle::pause() { 4092 mTrack->pause(); 4093} 4094 4095void AudioFlinger::TrackHandle::setVolume(float left, float right) { 4096 mTrack->setVolume(left, right); 4097} 4098 4099sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4100 return mTrack->getCblk(); 4101} 4102 4103status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4104{ 4105 return mTrack->attachAuxEffect(EffectId); 4106} 4107 4108status_t AudioFlinger::TrackHandle::onTransact( 4109 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4110{ 4111 return BnAudioTrack::onTransact(code, data, reply, flags); 4112} 4113 4114// ---------------------------------------------------------------------------- 4115 4116sp<IAudioRecord> AudioFlinger::openRecord( 4117 pid_t pid, 4118 int input, 4119 uint32_t sampleRate, 4120 uint32_t format, 4121 uint32_t channelMask, 4122 int frameCount, 4123 uint32_t flags, 4124 int *sessionId, 4125 status_t *status) 4126{ 4127 sp<RecordThread::RecordTrack> recordTrack; 4128 sp<RecordHandle> recordHandle; 4129 sp<Client> client; 4130 wp<Client> wclient; 4131 status_t lStatus; 4132 RecordThread *thread; 4133 size_t inFrameCount; 4134 int lSessionId; 4135 4136 // check calling permissions 4137 if (!recordingAllowed()) { 4138 lStatus = PERMISSION_DENIED; 4139 goto Exit; 4140 } 4141 4142 // add client to list 4143 { // scope for mLock 4144 Mutex::Autolock _l(mLock); 4145 thread = checkRecordThread_l(input); 4146 if (thread == NULL) { 4147 lStatus = BAD_VALUE; 4148 goto Exit; 4149 } 4150 4151 wclient = mClients.valueFor(pid); 4152 if (wclient != NULL) { 4153 client = wclient.promote(); 4154 } else { 4155 client = new Client(this, pid); 4156 mClients.add(pid, client); 4157 } 4158 4159 // If no audio session id is provided, create one here 4160 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4161 lSessionId = *sessionId; 4162 } else { 4163 lSessionId = nextUniqueId(); 4164 if (sessionId != NULL) { 4165 *sessionId = lSessionId; 4166 } 4167 } 4168 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4169 recordTrack = thread->createRecordTrack_l(client, 4170 sampleRate, 4171 format, 4172 channelMask, 4173 frameCount, 4174 flags, 4175 lSessionId, 4176 &lStatus); 4177 } 4178 if (lStatus != NO_ERROR) { 4179 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4180 // destructor is called by the TrackBase destructor with mLock held 4181 client.clear(); 4182 recordTrack.clear(); 4183 goto Exit; 4184 } 4185 4186 // return to handle to client 4187 recordHandle = new RecordHandle(recordTrack); 4188 lStatus = NO_ERROR; 4189 4190Exit: 4191 if (status) { 4192 *status = lStatus; 4193 } 4194 return recordHandle; 4195} 4196 4197// ---------------------------------------------------------------------------- 4198 4199AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4200 : BnAudioRecord(), 4201 mRecordTrack(recordTrack) 4202{ 4203} 4204 4205AudioFlinger::RecordHandle::~RecordHandle() { 4206 stop(); 4207} 4208 4209status_t AudioFlinger::RecordHandle::start() { 4210 ALOGV("RecordHandle::start()"); 4211 return mRecordTrack->start(); 4212} 4213 4214void AudioFlinger::RecordHandle::stop() { 4215 ALOGV("RecordHandle::stop()"); 4216 mRecordTrack->stop(); 4217} 4218 4219sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4220 return mRecordTrack->getCblk(); 4221} 4222 4223status_t AudioFlinger::RecordHandle::onTransact( 4224 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4225{ 4226 return BnAudioRecord::onTransact(code, data, reply, flags); 4227} 4228 4229// ---------------------------------------------------------------------------- 4230 4231AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4232 AudioStreamIn *input, 4233 uint32_t sampleRate, 4234 uint32_t channels, 4235 int id, 4236 uint32_t device) : 4237 ThreadBase(audioFlinger, id, device), 4238 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4239{ 4240 mType = ThreadBase::RECORD; 4241 4242 snprintf(mName, kNameLength, "AudioIn_%d", id); 4243 4244 mReqChannelCount = popcount(channels); 4245 mReqSampleRate = sampleRate; 4246 readInputParameters(); 4247} 4248 4249 4250AudioFlinger::RecordThread::~RecordThread() 4251{ 4252 delete[] mRsmpInBuffer; 4253 if (mResampler != NULL) { 4254 delete mResampler; 4255 delete[] mRsmpOutBuffer; 4256 } 4257} 4258 4259void AudioFlinger::RecordThread::onFirstRef() 4260{ 4261 run(mName, PRIORITY_URGENT_AUDIO); 4262} 4263 4264status_t AudioFlinger::RecordThread::readyToRun() 4265{ 4266 status_t status = initCheck(); 4267 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4268 return status; 4269} 4270 4271bool AudioFlinger::RecordThread::threadLoop() 4272{ 4273 AudioBufferProvider::Buffer buffer; 4274 sp<RecordTrack> activeTrack; 4275 Vector< sp<EffectChain> > effectChains; 4276 4277 nsecs_t lastWarning = 0; 4278 4279 acquireWakeLock(); 4280 4281 // start recording 4282 while (!exitPending()) { 4283 4284 processConfigEvents(); 4285 4286 { // scope for mLock 4287 Mutex::Autolock _l(mLock); 4288 checkForNewParameters_l(); 4289 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4290 if (!mStandby) { 4291 mInput->stream->common.standby(&mInput->stream->common); 4292 mStandby = true; 4293 } 4294 4295 if (exitPending()) break; 4296 4297 releaseWakeLock_l(); 4298 ALOGV("RecordThread: loop stopping"); 4299 // go to sleep 4300 mWaitWorkCV.wait(mLock); 4301 ALOGV("RecordThread: loop starting"); 4302 acquireWakeLock_l(); 4303 continue; 4304 } 4305 if (mActiveTrack != 0) { 4306 if (mActiveTrack->mState == TrackBase::PAUSING) { 4307 if (!mStandby) { 4308 mInput->stream->common.standby(&mInput->stream->common); 4309 mStandby = true; 4310 } 4311 mActiveTrack.clear(); 4312 mStartStopCond.broadcast(); 4313 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4314 if (mReqChannelCount != mActiveTrack->channelCount()) { 4315 mActiveTrack.clear(); 4316 mStartStopCond.broadcast(); 4317 } else if (mBytesRead != 0) { 4318 // record start succeeds only if first read from audio input 4319 // succeeds 4320 if (mBytesRead > 0) { 4321 mActiveTrack->mState = TrackBase::ACTIVE; 4322 } else { 4323 mActiveTrack.clear(); 4324 } 4325 mStartStopCond.broadcast(); 4326 } 4327 mStandby = false; 4328 } 4329 } 4330 lockEffectChains_l(effectChains); 4331 } 4332 4333 if (mActiveTrack != 0) { 4334 if (mActiveTrack->mState != TrackBase::ACTIVE && 4335 mActiveTrack->mState != TrackBase::RESUMING) { 4336 unlockEffectChains(effectChains); 4337 usleep(kRecordThreadSleepUs); 4338 continue; 4339 } 4340 for (size_t i = 0; i < effectChains.size(); i ++) { 4341 effectChains[i]->process_l(); 4342 } 4343 4344 buffer.frameCount = mFrameCount; 4345 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4346 size_t framesOut = buffer.frameCount; 4347 if (mResampler == NULL) { 4348 // no resampling 4349 while (framesOut) { 4350 size_t framesIn = mFrameCount - mRsmpInIndex; 4351 if (framesIn) { 4352 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4353 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4354 if (framesIn > framesOut) 4355 framesIn = framesOut; 4356 mRsmpInIndex += framesIn; 4357 framesOut -= framesIn; 4358 if ((int)mChannelCount == mReqChannelCount || 4359 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4360 memcpy(dst, src, framesIn * mFrameSize); 4361 } else { 4362 int16_t *src16 = (int16_t *)src; 4363 int16_t *dst16 = (int16_t *)dst; 4364 if (mChannelCount == 1) { 4365 while (framesIn--) { 4366 *dst16++ = *src16; 4367 *dst16++ = *src16++; 4368 } 4369 } else { 4370 while (framesIn--) { 4371 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4372 src16 += 2; 4373 } 4374 } 4375 } 4376 } 4377 if (framesOut && mFrameCount == mRsmpInIndex) { 4378 if (framesOut == mFrameCount && 4379 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4380 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4381 framesOut = 0; 4382 } else { 4383 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4384 mRsmpInIndex = 0; 4385 } 4386 if (mBytesRead < 0) { 4387 ALOGE("Error reading audio input"); 4388 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4389 // Force input into standby so that it tries to 4390 // recover at next read attempt 4391 mInput->stream->common.standby(&mInput->stream->common); 4392 usleep(kRecordThreadSleepUs); 4393 } 4394 mRsmpInIndex = mFrameCount; 4395 framesOut = 0; 4396 buffer.frameCount = 0; 4397 } 4398 } 4399 } 4400 } else { 4401 // resampling 4402 4403 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4404 // alter output frame count as if we were expecting stereo samples 4405 if (mChannelCount == 1 && mReqChannelCount == 1) { 4406 framesOut >>= 1; 4407 } 4408 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4409 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4410 // are 32 bit aligned which should be always true. 4411 if (mChannelCount == 2 && mReqChannelCount == 1) { 4412 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4413 // the resampler always outputs stereo samples: do post stereo to mono conversion 4414 int16_t *src = (int16_t *)mRsmpOutBuffer; 4415 int16_t *dst = buffer.i16; 4416 while (framesOut--) { 4417 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4418 src += 2; 4419 } 4420 } else { 4421 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4422 } 4423 4424 } 4425 mActiveTrack->releaseBuffer(&buffer); 4426 mActiveTrack->overflow(); 4427 } 4428 // client isn't retrieving buffers fast enough 4429 else { 4430 if (!mActiveTrack->setOverflow()) { 4431 nsecs_t now = systemTime(); 4432 if ((now - lastWarning) > kWarningThrottleNs) { 4433 ALOGW("RecordThread: buffer overflow"); 4434 lastWarning = now; 4435 } 4436 } 4437 // Release the processor for a while before asking for a new buffer. 4438 // This will give the application more chance to read from the buffer and 4439 // clear the overflow. 4440 usleep(kRecordThreadSleepUs); 4441 } 4442 } 4443 // enable changes in effect chain 4444 unlockEffectChains(effectChains); 4445 effectChains.clear(); 4446 } 4447 4448 if (!mStandby) { 4449 mInput->stream->common.standby(&mInput->stream->common); 4450 } 4451 mActiveTrack.clear(); 4452 4453 mStartStopCond.broadcast(); 4454 4455 releaseWakeLock(); 4456 4457 ALOGV("RecordThread %p exiting", this); 4458 return false; 4459} 4460 4461 4462sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4463 const sp<AudioFlinger::Client>& client, 4464 uint32_t sampleRate, 4465 int format, 4466 int channelMask, 4467 int frameCount, 4468 uint32_t flags, 4469 int sessionId, 4470 status_t *status) 4471{ 4472 sp<RecordTrack> track; 4473 status_t lStatus; 4474 4475 lStatus = initCheck(); 4476 if (lStatus != NO_ERROR) { 4477 ALOGE("Audio driver not initialized."); 4478 goto Exit; 4479 } 4480 4481 { // scope for mLock 4482 Mutex::Autolock _l(mLock); 4483 4484 track = new RecordTrack(this, client, sampleRate, 4485 format, channelMask, frameCount, flags, sessionId); 4486 4487 if (track->getCblk() == NULL) { 4488 lStatus = NO_MEMORY; 4489 goto Exit; 4490 } 4491 4492 mTrack = track.get(); 4493 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4494 bool suspend = audio_is_bluetooth_sco_device( 4495 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4496 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4497 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4498 } 4499 lStatus = NO_ERROR; 4500 4501Exit: 4502 if (status) { 4503 *status = lStatus; 4504 } 4505 return track; 4506} 4507 4508status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4509{ 4510 ALOGV("RecordThread::start"); 4511 sp <ThreadBase> strongMe = this; 4512 status_t status = NO_ERROR; 4513 { 4514 AutoMutex lock(mLock); 4515 if (mActiveTrack != 0) { 4516 if (recordTrack != mActiveTrack.get()) { 4517 status = -EBUSY; 4518 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4519 mActiveTrack->mState = TrackBase::ACTIVE; 4520 } 4521 return status; 4522 } 4523 4524 recordTrack->mState = TrackBase::IDLE; 4525 mActiveTrack = recordTrack; 4526 mLock.unlock(); 4527 status_t status = AudioSystem::startInput(mId); 4528 mLock.lock(); 4529 if (status != NO_ERROR) { 4530 mActiveTrack.clear(); 4531 return status; 4532 } 4533 mRsmpInIndex = mFrameCount; 4534 mBytesRead = 0; 4535 if (mResampler != NULL) { 4536 mResampler->reset(); 4537 } 4538 mActiveTrack->mState = TrackBase::RESUMING; 4539 // signal thread to start 4540 ALOGV("Signal record thread"); 4541 mWaitWorkCV.signal(); 4542 // do not wait for mStartStopCond if exiting 4543 if (mExiting) { 4544 mActiveTrack.clear(); 4545 status = INVALID_OPERATION; 4546 goto startError; 4547 } 4548 mStartStopCond.wait(mLock); 4549 if (mActiveTrack == 0) { 4550 ALOGV("Record failed to start"); 4551 status = BAD_VALUE; 4552 goto startError; 4553 } 4554 ALOGV("Record started OK"); 4555 return status; 4556 } 4557startError: 4558 AudioSystem::stopInput(mId); 4559 return status; 4560} 4561 4562void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4563 ALOGV("RecordThread::stop"); 4564 sp <ThreadBase> strongMe = this; 4565 { 4566 AutoMutex lock(mLock); 4567 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4568 mActiveTrack->mState = TrackBase::PAUSING; 4569 // do not wait for mStartStopCond if exiting 4570 if (mExiting) { 4571 return; 4572 } 4573 mStartStopCond.wait(mLock); 4574 // if we have been restarted, recordTrack == mActiveTrack.get() here 4575 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4576 mLock.unlock(); 4577 AudioSystem::stopInput(mId); 4578 mLock.lock(); 4579 ALOGV("Record stopped OK"); 4580 } 4581 } 4582 } 4583} 4584 4585status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4586{ 4587 const size_t SIZE = 256; 4588 char buffer[SIZE]; 4589 String8 result; 4590 pid_t pid = 0; 4591 4592 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4593 result.append(buffer); 4594 4595 if (mActiveTrack != 0) { 4596 result.append("Active Track:\n"); 4597 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4598 mActiveTrack->dump(buffer, SIZE); 4599 result.append(buffer); 4600 4601 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4602 result.append(buffer); 4603 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4604 result.append(buffer); 4605 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4606 result.append(buffer); 4607 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4608 result.append(buffer); 4609 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4610 result.append(buffer); 4611 4612 4613 } else { 4614 result.append("No record client\n"); 4615 } 4616 write(fd, result.string(), result.size()); 4617 4618 dumpBase(fd, args); 4619 dumpEffectChains(fd, args); 4620 4621 return NO_ERROR; 4622} 4623 4624status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4625{ 4626 size_t framesReq = buffer->frameCount; 4627 size_t framesReady = mFrameCount - mRsmpInIndex; 4628 int channelCount; 4629 4630 if (framesReady == 0) { 4631 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4632 if (mBytesRead < 0) { 4633 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4634 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4635 // Force input into standby so that it tries to 4636 // recover at next read attempt 4637 mInput->stream->common.standby(&mInput->stream->common); 4638 usleep(kRecordThreadSleepUs); 4639 } 4640 buffer->raw = NULL; 4641 buffer->frameCount = 0; 4642 return NOT_ENOUGH_DATA; 4643 } 4644 mRsmpInIndex = 0; 4645 framesReady = mFrameCount; 4646 } 4647 4648 if (framesReq > framesReady) { 4649 framesReq = framesReady; 4650 } 4651 4652 if (mChannelCount == 1 && mReqChannelCount == 2) { 4653 channelCount = 1; 4654 } else { 4655 channelCount = 2; 4656 } 4657 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4658 buffer->frameCount = framesReq; 4659 return NO_ERROR; 4660} 4661 4662void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4663{ 4664 mRsmpInIndex += buffer->frameCount; 4665 buffer->frameCount = 0; 4666} 4667 4668bool AudioFlinger::RecordThread::checkForNewParameters_l() 4669{ 4670 bool reconfig = false; 4671 4672 while (!mNewParameters.isEmpty()) { 4673 status_t status = NO_ERROR; 4674 String8 keyValuePair = mNewParameters[0]; 4675 AudioParameter param = AudioParameter(keyValuePair); 4676 int value; 4677 int reqFormat = mFormat; 4678 int reqSamplingRate = mReqSampleRate; 4679 int reqChannelCount = mReqChannelCount; 4680 4681 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4682 reqSamplingRate = value; 4683 reconfig = true; 4684 } 4685 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4686 reqFormat = value; 4687 reconfig = true; 4688 } 4689 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4690 reqChannelCount = popcount(value); 4691 reconfig = true; 4692 } 4693 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4694 // do not accept frame count changes if tracks are open as the track buffer 4695 // size depends on frame count and correct behavior would not be garantied 4696 // if frame count is changed after track creation 4697 if (mActiveTrack != 0) { 4698 status = INVALID_OPERATION; 4699 } else { 4700 reconfig = true; 4701 } 4702 } 4703 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4704 // forward device change to effects that have requested to be 4705 // aware of attached audio device. 4706 for (size_t i = 0; i < mEffectChains.size(); i++) { 4707 mEffectChains[i]->setDevice_l(value); 4708 } 4709 // store input device and output device but do not forward output device to audio HAL. 4710 // Note that status is ignored by the caller for output device 4711 // (see AudioFlinger::setParameters() 4712 if (value & AUDIO_DEVICE_OUT_ALL) { 4713 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4714 status = BAD_VALUE; 4715 } else { 4716 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4717 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4718 if (mTrack != NULL) { 4719 bool suspend = audio_is_bluetooth_sco_device( 4720 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4721 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4722 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4723 } 4724 } 4725 mDevice |= (uint32_t)value; 4726 } 4727 if (status == NO_ERROR) { 4728 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4729 if (status == INVALID_OPERATION) { 4730 mInput->stream->common.standby(&mInput->stream->common); 4731 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4732 } 4733 if (reconfig) { 4734 if (status == BAD_VALUE && 4735 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4736 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4737 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4738 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4739 (reqChannelCount < 3)) { 4740 status = NO_ERROR; 4741 } 4742 if (status == NO_ERROR) { 4743 readInputParameters(); 4744 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4745 } 4746 } 4747 } 4748 4749 mNewParameters.removeAt(0); 4750 4751 mParamStatus = status; 4752 mParamCond.signal(); 4753 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4754 // already timed out waiting for the status and will never signal the condition. 4755 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4756 } 4757 return reconfig; 4758} 4759 4760String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4761{ 4762 char *s; 4763 String8 out_s8 = String8(); 4764 4765 Mutex::Autolock _l(mLock); 4766 if (initCheck() != NO_ERROR) { 4767 return out_s8; 4768 } 4769 4770 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4771 out_s8 = String8(s); 4772 free(s); 4773 return out_s8; 4774} 4775 4776void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4777 AudioSystem::OutputDescriptor desc; 4778 void *param2 = 0; 4779 4780 switch (event) { 4781 case AudioSystem::INPUT_OPENED: 4782 case AudioSystem::INPUT_CONFIG_CHANGED: 4783 desc.channels = mChannelMask; 4784 desc.samplingRate = mSampleRate; 4785 desc.format = mFormat; 4786 desc.frameCount = mFrameCount; 4787 desc.latency = 0; 4788 param2 = &desc; 4789 break; 4790 4791 case AudioSystem::INPUT_CLOSED: 4792 default: 4793 break; 4794 } 4795 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4796} 4797 4798void AudioFlinger::RecordThread::readInputParameters() 4799{ 4800 if (mRsmpInBuffer) delete mRsmpInBuffer; 4801 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4802 if (mResampler) delete mResampler; 4803 mResampler = NULL; 4804 4805 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4806 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4807 mChannelCount = (uint16_t)popcount(mChannelMask); 4808 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4809 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); 4810 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4811 mFrameCount = mInputBytes / mFrameSize; 4812 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4813 4814 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4815 { 4816 int channelCount; 4817 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4818 // stereo to mono post process as the resampler always outputs stereo. 4819 if (mChannelCount == 1 && mReqChannelCount == 2) { 4820 channelCount = 1; 4821 } else { 4822 channelCount = 2; 4823 } 4824 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4825 mResampler->setSampleRate(mSampleRate); 4826 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4827 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4828 4829 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4830 if (mChannelCount == 1 && mReqChannelCount == 1) { 4831 mFrameCount >>= 1; 4832 } 4833 4834 } 4835 mRsmpInIndex = mFrameCount; 4836} 4837 4838unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4839{ 4840 Mutex::Autolock _l(mLock); 4841 if (initCheck() != NO_ERROR) { 4842 return 0; 4843 } 4844 4845 return mInput->stream->get_input_frames_lost(mInput->stream); 4846} 4847 4848uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4849{ 4850 Mutex::Autolock _l(mLock); 4851 uint32_t result = 0; 4852 if (getEffectChain_l(sessionId) != 0) { 4853 result = EFFECT_SESSION; 4854 } 4855 4856 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4857 result |= TRACK_SESSION; 4858 } 4859 4860 return result; 4861} 4862 4863AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4864{ 4865 Mutex::Autolock _l(mLock); 4866 return mTrack; 4867} 4868 4869AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() 4870{ 4871 Mutex::Autolock _l(mLock); 4872 return mInput; 4873} 4874 4875AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4876{ 4877 Mutex::Autolock _l(mLock); 4878 AudioStreamIn *input = mInput; 4879 mInput = NULL; 4880 return input; 4881} 4882 4883// this method must always be called either with ThreadBase mLock held or inside the thread loop 4884audio_stream_t* AudioFlinger::RecordThread::stream() 4885{ 4886 if (mInput == NULL) { 4887 return NULL; 4888 } 4889 return &mInput->stream->common; 4890} 4891 4892 4893// ---------------------------------------------------------------------------- 4894 4895int AudioFlinger::openOutput(uint32_t *pDevices, 4896 uint32_t *pSamplingRate, 4897 uint32_t *pFormat, 4898 uint32_t *pChannels, 4899 uint32_t *pLatencyMs, 4900 uint32_t flags) 4901{ 4902 status_t status; 4903 PlaybackThread *thread = NULL; 4904 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4905 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4906 uint32_t format = pFormat ? *pFormat : 0; 4907 uint32_t channels = pChannels ? *pChannels : 0; 4908 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4909 audio_stream_out_t *outStream; 4910 audio_hw_device_t *outHwDev; 4911 4912 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4913 pDevices ? *pDevices : 0, 4914 samplingRate, 4915 format, 4916 channels, 4917 flags); 4918 4919 if (pDevices == NULL || *pDevices == 0) { 4920 return 0; 4921 } 4922 4923 Mutex::Autolock _l(mLock); 4924 4925 outHwDev = findSuitableHwDev_l(*pDevices); 4926 if (outHwDev == NULL) 4927 return 0; 4928 4929 status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, 4930 &channels, &samplingRate, &outStream); 4931 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4932 outStream, 4933 samplingRate, 4934 format, 4935 channels, 4936 status); 4937 4938 mHardwareStatus = AUDIO_HW_IDLE; 4939 if (outStream != NULL) { 4940 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4941 int id = nextUniqueId(); 4942 4943 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4944 (format != AUDIO_FORMAT_PCM_16_BIT) || 4945 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4946 thread = new DirectOutputThread(this, output, id, *pDevices); 4947 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4948 } else { 4949 thread = new MixerThread(this, output, id, *pDevices); 4950 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4951 } 4952 mPlaybackThreads.add(id, thread); 4953 4954 if (pSamplingRate) *pSamplingRate = samplingRate; 4955 if (pFormat) *pFormat = format; 4956 if (pChannels) *pChannels = channels; 4957 if (pLatencyMs) *pLatencyMs = thread->latency(); 4958 4959 // notify client processes of the new output creation 4960 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4961 return id; 4962 } 4963 4964 return 0; 4965} 4966 4967int AudioFlinger::openDuplicateOutput(int output1, int output2) 4968{ 4969 Mutex::Autolock _l(mLock); 4970 MixerThread *thread1 = checkMixerThread_l(output1); 4971 MixerThread *thread2 = checkMixerThread_l(output2); 4972 4973 if (thread1 == NULL || thread2 == NULL) { 4974 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4975 return 0; 4976 } 4977 4978 int id = nextUniqueId(); 4979 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4980 thread->addOutputTrack(thread2); 4981 mPlaybackThreads.add(id, thread); 4982 // notify client processes of the new output creation 4983 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4984 return id; 4985} 4986 4987status_t AudioFlinger::closeOutput(int output) 4988{ 4989 // keep strong reference on the playback thread so that 4990 // it is not destroyed while exit() is executed 4991 sp <PlaybackThread> thread; 4992 { 4993 Mutex::Autolock _l(mLock); 4994 thread = checkPlaybackThread_l(output); 4995 if (thread == NULL) { 4996 return BAD_VALUE; 4997 } 4998 4999 ALOGV("closeOutput() %d", output); 5000 5001 if (thread->type() == ThreadBase::MIXER) { 5002 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5003 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5004 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5005 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5006 } 5007 } 5008 } 5009 void *param2 = 0; 5010 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5011 mPlaybackThreads.removeItem(output); 5012 } 5013 thread->exit(); 5014 5015 if (thread->type() != ThreadBase::DUPLICATING) { 5016 AudioStreamOut *out = thread->clearOutput(); 5017 // from now on thread->mOutput is NULL 5018 out->hwDev->close_output_stream(out->hwDev, out->stream); 5019 delete out; 5020 } 5021 return NO_ERROR; 5022} 5023 5024status_t AudioFlinger::suspendOutput(int output) 5025{ 5026 Mutex::Autolock _l(mLock); 5027 PlaybackThread *thread = checkPlaybackThread_l(output); 5028 5029 if (thread == NULL) { 5030 return BAD_VALUE; 5031 } 5032 5033 ALOGV("suspendOutput() %d", output); 5034 thread->suspend(); 5035 5036 return NO_ERROR; 5037} 5038 5039status_t AudioFlinger::restoreOutput(int output) 5040{ 5041 Mutex::Autolock _l(mLock); 5042 PlaybackThread *thread = checkPlaybackThread_l(output); 5043 5044 if (thread == NULL) { 5045 return BAD_VALUE; 5046 } 5047 5048 ALOGV("restoreOutput() %d", output); 5049 5050 thread->restore(); 5051 5052 return NO_ERROR; 5053} 5054 5055int AudioFlinger::openInput(uint32_t *pDevices, 5056 uint32_t *pSamplingRate, 5057 uint32_t *pFormat, 5058 uint32_t *pChannels, 5059 uint32_t acoustics) 5060{ 5061 status_t status; 5062 RecordThread *thread = NULL; 5063 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5064 uint32_t format = pFormat ? *pFormat : 0; 5065 uint32_t channels = pChannels ? *pChannels : 0; 5066 uint32_t reqSamplingRate = samplingRate; 5067 uint32_t reqFormat = format; 5068 uint32_t reqChannels = channels; 5069 audio_stream_in_t *inStream; 5070 audio_hw_device_t *inHwDev; 5071 5072 if (pDevices == NULL || *pDevices == 0) { 5073 return 0; 5074 } 5075 5076 Mutex::Autolock _l(mLock); 5077 5078 inHwDev = findSuitableHwDev_l(*pDevices); 5079 if (inHwDev == NULL) 5080 return 0; 5081 5082 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5083 &channels, &samplingRate, 5084 (audio_in_acoustics_t)acoustics, 5085 &inStream); 5086 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5087 inStream, 5088 samplingRate, 5089 format, 5090 channels, 5091 acoustics, 5092 status); 5093 5094 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5095 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5096 // or stereo to mono conversions on 16 bit PCM inputs. 5097 if (inStream == NULL && status == BAD_VALUE && 5098 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5099 (samplingRate <= 2 * reqSamplingRate) && 5100 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5101 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5102 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5103 &channels, &samplingRate, 5104 (audio_in_acoustics_t)acoustics, 5105 &inStream); 5106 } 5107 5108 if (inStream != NULL) { 5109 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5110 5111 int id = nextUniqueId(); 5112 // Start record thread 5113 // RecorThread require both input and output device indication to forward to audio 5114 // pre processing modules 5115 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5116 thread = new RecordThread(this, 5117 input, 5118 reqSamplingRate, 5119 reqChannels, 5120 id, 5121 device); 5122 mRecordThreads.add(id, thread); 5123 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5124 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5125 if (pFormat) *pFormat = format; 5126 if (pChannels) *pChannels = reqChannels; 5127 5128 input->stream->common.standby(&input->stream->common); 5129 5130 // notify client processes of the new input creation 5131 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5132 return id; 5133 } 5134 5135 return 0; 5136} 5137 5138status_t AudioFlinger::closeInput(int input) 5139{ 5140 // keep strong reference on the record thread so that 5141 // it is not destroyed while exit() is executed 5142 sp <RecordThread> thread; 5143 { 5144 Mutex::Autolock _l(mLock); 5145 thread = checkRecordThread_l(input); 5146 if (thread == NULL) { 5147 return BAD_VALUE; 5148 } 5149 5150 ALOGV("closeInput() %d", input); 5151 void *param2 = 0; 5152 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5153 mRecordThreads.removeItem(input); 5154 } 5155 thread->exit(); 5156 5157 AudioStreamIn *in = thread->clearInput(); 5158 // from now on thread->mInput is NULL 5159 in->hwDev->close_input_stream(in->hwDev, in->stream); 5160 delete in; 5161 5162 return NO_ERROR; 5163} 5164 5165status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 5166{ 5167 Mutex::Autolock _l(mLock); 5168 MixerThread *dstThread = checkMixerThread_l(output); 5169 if (dstThread == NULL) { 5170 ALOGW("setStreamOutput() bad output id %d", output); 5171 return BAD_VALUE; 5172 } 5173 5174 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5175 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5176 5177 dstThread->setStreamValid(stream, true); 5178 5179 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5180 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5181 if (thread != dstThread && 5182 thread->type() != ThreadBase::DIRECT) { 5183 MixerThread *srcThread = (MixerThread *)thread; 5184 srcThread->setStreamValid(stream, false); 5185 srcThread->invalidateTracks(stream); 5186 } 5187 } 5188 5189 return NO_ERROR; 5190} 5191 5192 5193int AudioFlinger::newAudioSessionId() 5194{ 5195 return nextUniqueId(); 5196} 5197 5198void AudioFlinger::acquireAudioSessionId(int audioSession) 5199{ 5200 Mutex::Autolock _l(mLock); 5201 int caller = IPCThreadState::self()->getCallingPid(); 5202 ALOGV("acquiring %d from %d", audioSession, caller); 5203 int num = mAudioSessionRefs.size(); 5204 for (int i = 0; i< num; i++) { 5205 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5206 if (ref->sessionid == audioSession && ref->pid == caller) { 5207 ref->cnt++; 5208 ALOGV(" incremented refcount to %d", ref->cnt); 5209 return; 5210 } 5211 } 5212 AudioSessionRef *ref = new AudioSessionRef(); 5213 ref->sessionid = audioSession; 5214 ref->pid = caller; 5215 ref->cnt = 1; 5216 mAudioSessionRefs.push(ref); 5217 ALOGV(" added new entry for %d", ref->sessionid); 5218} 5219 5220void AudioFlinger::releaseAudioSessionId(int audioSession) 5221{ 5222 Mutex::Autolock _l(mLock); 5223 int caller = IPCThreadState::self()->getCallingPid(); 5224 ALOGV("releasing %d from %d", audioSession, caller); 5225 int num = mAudioSessionRefs.size(); 5226 for (int i = 0; i< num; i++) { 5227 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5228 if (ref->sessionid == audioSession && ref->pid == caller) { 5229 ref->cnt--; 5230 ALOGV(" decremented refcount to %d", ref->cnt); 5231 if (ref->cnt == 0) { 5232 mAudioSessionRefs.removeAt(i); 5233 delete ref; 5234 purgeStaleEffects_l(); 5235 } 5236 return; 5237 } 5238 } 5239 ALOGW("session id %d not found for pid %d", audioSession, caller); 5240} 5241 5242void AudioFlinger::purgeStaleEffects_l() { 5243 5244 ALOGV("purging stale effects"); 5245 5246 Vector< sp<EffectChain> > chains; 5247 5248 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5249 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5250 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5251 sp<EffectChain> ec = t->mEffectChains[j]; 5252 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5253 chains.push(ec); 5254 } 5255 } 5256 } 5257 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5258 sp<RecordThread> t = mRecordThreads.valueAt(i); 5259 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5260 sp<EffectChain> ec = t->mEffectChains[j]; 5261 chains.push(ec); 5262 } 5263 } 5264 5265 for (size_t i = 0; i < chains.size(); i++) { 5266 sp<EffectChain> ec = chains[i]; 5267 int sessionid = ec->sessionId(); 5268 sp<ThreadBase> t = ec->mThread.promote(); 5269 if (t == 0) { 5270 continue; 5271 } 5272 size_t numsessionrefs = mAudioSessionRefs.size(); 5273 bool found = false; 5274 for (size_t k = 0; k < numsessionrefs; k++) { 5275 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5276 if (ref->sessionid == sessionid) { 5277 ALOGV(" session %d still exists for %d with %d refs", 5278 sessionid, ref->pid, ref->cnt); 5279 found = true; 5280 break; 5281 } 5282 } 5283 if (!found) { 5284 // remove all effects from the chain 5285 while (ec->mEffects.size()) { 5286 sp<EffectModule> effect = ec->mEffects[0]; 5287 effect->unPin(); 5288 Mutex::Autolock _l (t->mLock); 5289 t->removeEffect_l(effect); 5290 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5291 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5292 if (handle != 0) { 5293 handle->mEffect.clear(); 5294 if (handle->mHasControl && handle->mEnabled) { 5295 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5296 } 5297 } 5298 } 5299 AudioSystem::unregisterEffect(effect->id()); 5300 } 5301 } 5302 } 5303 return; 5304} 5305 5306// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5307AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5308{ 5309 PlaybackThread *thread = NULL; 5310 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5311 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5312 } 5313 return thread; 5314} 5315 5316// checkMixerThread_l() must be called with AudioFlinger::mLock held 5317AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5318{ 5319 PlaybackThread *thread = checkPlaybackThread_l(output); 5320 if (thread != NULL) { 5321 if (thread->type() == ThreadBase::DIRECT) { 5322 thread = NULL; 5323 } 5324 } 5325 return (MixerThread *)thread; 5326} 5327 5328// checkRecordThread_l() must be called with AudioFlinger::mLock held 5329AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5330{ 5331 RecordThread *thread = NULL; 5332 if (mRecordThreads.indexOfKey(input) >= 0) { 5333 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5334 } 5335 return thread; 5336} 5337 5338uint32_t AudioFlinger::nextUniqueId() 5339{ 5340 return android_atomic_inc(&mNextUniqueId); 5341} 5342 5343AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5344{ 5345 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5346 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5347 AudioStreamOut *output = thread->getOutput(); 5348 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5349 return thread; 5350 } 5351 } 5352 return NULL; 5353} 5354 5355uint32_t AudioFlinger::primaryOutputDevice_l() 5356{ 5357 PlaybackThread *thread = primaryPlaybackThread_l(); 5358 5359 if (thread == NULL) { 5360 return 0; 5361 } 5362 5363 return thread->device(); 5364} 5365 5366 5367// ---------------------------------------------------------------------------- 5368// Effect management 5369// ---------------------------------------------------------------------------- 5370 5371 5372status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5373{ 5374 Mutex::Autolock _l(mLock); 5375 return EffectQueryNumberEffects(numEffects); 5376} 5377 5378status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5379{ 5380 Mutex::Autolock _l(mLock); 5381 return EffectQueryEffect(index, descriptor); 5382} 5383 5384status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5385{ 5386 Mutex::Autolock _l(mLock); 5387 return EffectGetDescriptor(pUuid, descriptor); 5388} 5389 5390 5391sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5392 effect_descriptor_t *pDesc, 5393 const sp<IEffectClient>& effectClient, 5394 int32_t priority, 5395 int io, 5396 int sessionId, 5397 status_t *status, 5398 int *id, 5399 int *enabled) 5400{ 5401 status_t lStatus = NO_ERROR; 5402 sp<EffectHandle> handle; 5403 effect_descriptor_t desc; 5404 sp<Client> client; 5405 wp<Client> wclient; 5406 5407 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5408 pid, effectClient.get(), priority, sessionId, io); 5409 5410 if (pDesc == NULL) { 5411 lStatus = BAD_VALUE; 5412 goto Exit; 5413 } 5414 5415 // check audio settings permission for global effects 5416 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5417 lStatus = PERMISSION_DENIED; 5418 goto Exit; 5419 } 5420 5421 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5422 // that can only be created by audio policy manager (running in same process) 5423 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5424 lStatus = PERMISSION_DENIED; 5425 goto Exit; 5426 } 5427 5428 if (io == 0) { 5429 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5430 // output must be specified by AudioPolicyManager when using session 5431 // AUDIO_SESSION_OUTPUT_STAGE 5432 lStatus = BAD_VALUE; 5433 goto Exit; 5434 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5435 // if the output returned by getOutputForEffect() is removed before we lock the 5436 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5437 // and we will exit safely 5438 io = AudioSystem::getOutputForEffect(&desc); 5439 } 5440 } 5441 5442 { 5443 Mutex::Autolock _l(mLock); 5444 5445 5446 if (!EffectIsNullUuid(&pDesc->uuid)) { 5447 // if uuid is specified, request effect descriptor 5448 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5449 if (lStatus < 0) { 5450 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5451 goto Exit; 5452 } 5453 } else { 5454 // if uuid is not specified, look for an available implementation 5455 // of the required type in effect factory 5456 if (EffectIsNullUuid(&pDesc->type)) { 5457 ALOGW("createEffect() no effect type"); 5458 lStatus = BAD_VALUE; 5459 goto Exit; 5460 } 5461 uint32_t numEffects = 0; 5462 effect_descriptor_t d; 5463 d.flags = 0; // prevent compiler warning 5464 bool found = false; 5465 5466 lStatus = EffectQueryNumberEffects(&numEffects); 5467 if (lStatus < 0) { 5468 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5469 goto Exit; 5470 } 5471 for (uint32_t i = 0; i < numEffects; i++) { 5472 lStatus = EffectQueryEffect(i, &desc); 5473 if (lStatus < 0) { 5474 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5475 continue; 5476 } 5477 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5478 // If matching type found save effect descriptor. If the session is 5479 // 0 and the effect is not auxiliary, continue enumeration in case 5480 // an auxiliary version of this effect type is available 5481 found = true; 5482 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5483 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5484 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5485 break; 5486 } 5487 } 5488 } 5489 if (!found) { 5490 lStatus = BAD_VALUE; 5491 ALOGW("createEffect() effect not found"); 5492 goto Exit; 5493 } 5494 // For same effect type, chose auxiliary version over insert version if 5495 // connect to output mix (Compliance to OpenSL ES) 5496 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5497 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5498 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5499 } 5500 } 5501 5502 // Do not allow auxiliary effects on a session different from 0 (output mix) 5503 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5504 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5505 lStatus = INVALID_OPERATION; 5506 goto Exit; 5507 } 5508 5509 // check recording permission for visualizer 5510 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5511 !recordingAllowed()) { 5512 lStatus = PERMISSION_DENIED; 5513 goto Exit; 5514 } 5515 5516 // return effect descriptor 5517 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5518 5519 // If output is not specified try to find a matching audio session ID in one of the 5520 // output threads. 5521 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5522 // because of code checking output when entering the function. 5523 // Note: io is never 0 when creating an effect on an input 5524 if (io == 0) { 5525 // look for the thread where the specified audio session is present 5526 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5527 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5528 io = mPlaybackThreads.keyAt(i); 5529 break; 5530 } 5531 } 5532 if (io == 0) { 5533 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5534 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5535 io = mRecordThreads.keyAt(i); 5536 break; 5537 } 5538 } 5539 } 5540 // If no output thread contains the requested session ID, default to 5541 // first output. The effect chain will be moved to the correct output 5542 // thread when a track with the same session ID is created 5543 if (io == 0 && mPlaybackThreads.size()) { 5544 io = mPlaybackThreads.keyAt(0); 5545 } 5546 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5547 } 5548 ThreadBase *thread = checkRecordThread_l(io); 5549 if (thread == NULL) { 5550 thread = checkPlaybackThread_l(io); 5551 if (thread == NULL) { 5552 ALOGE("createEffect() unknown output thread"); 5553 lStatus = BAD_VALUE; 5554 goto Exit; 5555 } 5556 } 5557 5558 wclient = mClients.valueFor(pid); 5559 5560 if (wclient != NULL) { 5561 client = wclient.promote(); 5562 } else { 5563 client = new Client(this, pid); 5564 mClients.add(pid, client); 5565 } 5566 5567 // create effect on selected output thread 5568 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5569 &desc, enabled, &lStatus); 5570 if (handle != 0 && id != NULL) { 5571 *id = handle->id(); 5572 } 5573 } 5574 5575Exit: 5576 if(status) { 5577 *status = lStatus; 5578 } 5579 return handle; 5580} 5581 5582status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5583{ 5584 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5585 sessionId, srcOutput, dstOutput); 5586 Mutex::Autolock _l(mLock); 5587 if (srcOutput == dstOutput) { 5588 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5589 return NO_ERROR; 5590 } 5591 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5592 if (srcThread == NULL) { 5593 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5594 return BAD_VALUE; 5595 } 5596 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5597 if (dstThread == NULL) { 5598 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5599 return BAD_VALUE; 5600 } 5601 5602 Mutex::Autolock _dl(dstThread->mLock); 5603 Mutex::Autolock _sl(srcThread->mLock); 5604 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5605 5606 return NO_ERROR; 5607} 5608 5609// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5610status_t AudioFlinger::moveEffectChain_l(int sessionId, 5611 AudioFlinger::PlaybackThread *srcThread, 5612 AudioFlinger::PlaybackThread *dstThread, 5613 bool reRegister) 5614{ 5615 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5616 sessionId, srcThread, dstThread); 5617 5618 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5619 if (chain == 0) { 5620 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5621 sessionId, srcThread); 5622 return INVALID_OPERATION; 5623 } 5624 5625 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5626 // so that a new chain is created with correct parameters when first effect is added. This is 5627 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5628 // removed. 5629 srcThread->removeEffectChain_l(chain); 5630 5631 // transfer all effects one by one so that new effect chain is created on new thread with 5632 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5633 int dstOutput = dstThread->id(); 5634 sp<EffectChain> dstChain; 5635 uint32_t strategy = 0; // prevent compiler warning 5636 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5637 while (effect != 0) { 5638 srcThread->removeEffect_l(effect); 5639 dstThread->addEffect_l(effect); 5640 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5641 if (effect->state() == EffectModule::ACTIVE || 5642 effect->state() == EffectModule::STOPPING) { 5643 effect->start(); 5644 } 5645 // if the move request is not received from audio policy manager, the effect must be 5646 // re-registered with the new strategy and output 5647 if (dstChain == 0) { 5648 dstChain = effect->chain().promote(); 5649 if (dstChain == 0) { 5650 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5651 srcThread->addEffect_l(effect); 5652 return NO_INIT; 5653 } 5654 strategy = dstChain->strategy(); 5655 } 5656 if (reRegister) { 5657 AudioSystem::unregisterEffect(effect->id()); 5658 AudioSystem::registerEffect(&effect->desc(), 5659 dstOutput, 5660 strategy, 5661 sessionId, 5662 effect->id()); 5663 } 5664 effect = chain->getEffectFromId_l(0); 5665 } 5666 5667 return NO_ERROR; 5668} 5669 5670 5671// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5672sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5673 const sp<AudioFlinger::Client>& client, 5674 const sp<IEffectClient>& effectClient, 5675 int32_t priority, 5676 int sessionId, 5677 effect_descriptor_t *desc, 5678 int *enabled, 5679 status_t *status 5680 ) 5681{ 5682 sp<EffectModule> effect; 5683 sp<EffectHandle> handle; 5684 status_t lStatus; 5685 sp<EffectChain> chain; 5686 bool chainCreated = false; 5687 bool effectCreated = false; 5688 bool effectRegistered = false; 5689 5690 lStatus = initCheck(); 5691 if (lStatus != NO_ERROR) { 5692 ALOGW("createEffect_l() Audio driver not initialized."); 5693 goto Exit; 5694 } 5695 5696 // Do not allow effects with session ID 0 on direct output or duplicating threads 5697 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5698 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5699 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5700 desc->name, sessionId); 5701 lStatus = BAD_VALUE; 5702 goto Exit; 5703 } 5704 // Only Pre processor effects are allowed on input threads and only on input threads 5705 if ((mType == RECORD && 5706 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5707 (mType != RECORD && 5708 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5709 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5710 desc->name, desc->flags, mType); 5711 lStatus = BAD_VALUE; 5712 goto Exit; 5713 } 5714 5715 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5716 5717 { // scope for mLock 5718 Mutex::Autolock _l(mLock); 5719 5720 // check for existing effect chain with the requested audio session 5721 chain = getEffectChain_l(sessionId); 5722 if (chain == 0) { 5723 // create a new chain for this session 5724 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5725 chain = new EffectChain(this, sessionId); 5726 addEffectChain_l(chain); 5727 chain->setStrategy(getStrategyForSession_l(sessionId)); 5728 chainCreated = true; 5729 } else { 5730 effect = chain->getEffectFromDesc_l(desc); 5731 } 5732 5733 ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 5734 5735 if (effect == 0) { 5736 int id = mAudioFlinger->nextUniqueId(); 5737 // Check CPU and memory usage 5738 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5739 if (lStatus != NO_ERROR) { 5740 goto Exit; 5741 } 5742 effectRegistered = true; 5743 // create a new effect module if none present in the chain 5744 effect = new EffectModule(this, chain, desc, id, sessionId); 5745 lStatus = effect->status(); 5746 if (lStatus != NO_ERROR) { 5747 goto Exit; 5748 } 5749 lStatus = chain->addEffect_l(effect); 5750 if (lStatus != NO_ERROR) { 5751 goto Exit; 5752 } 5753 effectCreated = true; 5754 5755 effect->setDevice(mDevice); 5756 effect->setMode(mAudioFlinger->getMode()); 5757 } 5758 // create effect handle and connect it to effect module 5759 handle = new EffectHandle(effect, client, effectClient, priority); 5760 lStatus = effect->addHandle(handle); 5761 if (enabled) { 5762 *enabled = (int)effect->isEnabled(); 5763 } 5764 } 5765 5766Exit: 5767 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5768 Mutex::Autolock _l(mLock); 5769 if (effectCreated) { 5770 chain->removeEffect_l(effect); 5771 } 5772 if (effectRegistered) { 5773 AudioSystem::unregisterEffect(effect->id()); 5774 } 5775 if (chainCreated) { 5776 removeEffectChain_l(chain); 5777 } 5778 handle.clear(); 5779 } 5780 5781 if(status) { 5782 *status = lStatus; 5783 } 5784 return handle; 5785} 5786 5787sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5788{ 5789 sp<EffectModule> effect; 5790 5791 sp<EffectChain> chain = getEffectChain_l(sessionId); 5792 if (chain != 0) { 5793 effect = chain->getEffectFromId_l(effectId); 5794 } 5795 return effect; 5796} 5797 5798// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5799// PlaybackThread::mLock held 5800status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5801{ 5802 // check for existing effect chain with the requested audio session 5803 int sessionId = effect->sessionId(); 5804 sp<EffectChain> chain = getEffectChain_l(sessionId); 5805 bool chainCreated = false; 5806 5807 if (chain == 0) { 5808 // create a new chain for this session 5809 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5810 chain = new EffectChain(this, sessionId); 5811 addEffectChain_l(chain); 5812 chain->setStrategy(getStrategyForSession_l(sessionId)); 5813 chainCreated = true; 5814 } 5815 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5816 5817 if (chain->getEffectFromId_l(effect->id()) != 0) { 5818 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5819 this, effect->desc().name, chain.get()); 5820 return BAD_VALUE; 5821 } 5822 5823 status_t status = chain->addEffect_l(effect); 5824 if (status != NO_ERROR) { 5825 if (chainCreated) { 5826 removeEffectChain_l(chain); 5827 } 5828 return status; 5829 } 5830 5831 effect->setDevice(mDevice); 5832 effect->setMode(mAudioFlinger->getMode()); 5833 return NO_ERROR; 5834} 5835 5836void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5837 5838 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5839 effect_descriptor_t desc = effect->desc(); 5840 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5841 detachAuxEffect_l(effect->id()); 5842 } 5843 5844 sp<EffectChain> chain = effect->chain().promote(); 5845 if (chain != 0) { 5846 // remove effect chain if removing last effect 5847 if (chain->removeEffect_l(effect) == 0) { 5848 removeEffectChain_l(chain); 5849 } 5850 } else { 5851 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5852 } 5853} 5854 5855void AudioFlinger::ThreadBase::lockEffectChains_l( 5856 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5857{ 5858 effectChains = mEffectChains; 5859 for (size_t i = 0; i < mEffectChains.size(); i++) { 5860 mEffectChains[i]->lock(); 5861 } 5862} 5863 5864void AudioFlinger::ThreadBase::unlockEffectChains( 5865 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5866{ 5867 for (size_t i = 0; i < effectChains.size(); i++) { 5868 effectChains[i]->unlock(); 5869 } 5870} 5871 5872sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5873{ 5874 Mutex::Autolock _l(mLock); 5875 return getEffectChain_l(sessionId); 5876} 5877 5878sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5879{ 5880 sp<EffectChain> chain; 5881 5882 size_t size = mEffectChains.size(); 5883 for (size_t i = 0; i < size; i++) { 5884 if (mEffectChains[i]->sessionId() == sessionId) { 5885 chain = mEffectChains[i]; 5886 break; 5887 } 5888 } 5889 return chain; 5890} 5891 5892void AudioFlinger::ThreadBase::setMode(uint32_t mode) 5893{ 5894 Mutex::Autolock _l(mLock); 5895 size_t size = mEffectChains.size(); 5896 for (size_t i = 0; i < size; i++) { 5897 mEffectChains[i]->setMode_l(mode); 5898 } 5899} 5900 5901void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5902 const wp<EffectHandle>& handle, 5903 bool unpiniflast) { 5904 5905 Mutex::Autolock _l(mLock); 5906 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5907 // delete the effect module if removing last handle on it 5908 if (effect->removeHandle(handle) == 0) { 5909 if (!effect->isPinned() || unpiniflast) { 5910 removeEffect_l(effect); 5911 AudioSystem::unregisterEffect(effect->id()); 5912 } 5913 } 5914} 5915 5916status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5917{ 5918 int session = chain->sessionId(); 5919 int16_t *buffer = mMixBuffer; 5920 bool ownsBuffer = false; 5921 5922 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5923 if (session > 0) { 5924 // Only one effect chain can be present in direct output thread and it uses 5925 // the mix buffer as input 5926 if (mType != DIRECT) { 5927 size_t numSamples = mFrameCount * mChannelCount; 5928 buffer = new int16_t[numSamples]; 5929 memset(buffer, 0, numSamples * sizeof(int16_t)); 5930 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5931 ownsBuffer = true; 5932 } 5933 5934 // Attach all tracks with same session ID to this chain. 5935 for (size_t i = 0; i < mTracks.size(); ++i) { 5936 sp<Track> track = mTracks[i]; 5937 if (session == track->sessionId()) { 5938 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5939 track->setMainBuffer(buffer); 5940 chain->incTrackCnt(); 5941 } 5942 } 5943 5944 // indicate all active tracks in the chain 5945 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5946 sp<Track> track = mActiveTracks[i].promote(); 5947 if (track == 0) continue; 5948 if (session == track->sessionId()) { 5949 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5950 chain->incActiveTrackCnt(); 5951 } 5952 } 5953 } 5954 5955 chain->setInBuffer(buffer, ownsBuffer); 5956 chain->setOutBuffer(mMixBuffer); 5957 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5958 // chains list in order to be processed last as it contains output stage effects 5959 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5960 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5961 // after track specific effects and before output stage 5962 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5963 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5964 // Effect chain for other sessions are inserted at beginning of effect 5965 // chains list to be processed before output mix effects. Relative order between other 5966 // sessions is not important 5967 size_t size = mEffectChains.size(); 5968 size_t i = 0; 5969 for (i = 0; i < size; i++) { 5970 if (mEffectChains[i]->sessionId() < session) break; 5971 } 5972 mEffectChains.insertAt(chain, i); 5973 checkSuspendOnAddEffectChain_l(chain); 5974 5975 return NO_ERROR; 5976} 5977 5978size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5979{ 5980 int session = chain->sessionId(); 5981 5982 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5983 5984 for (size_t i = 0; i < mEffectChains.size(); i++) { 5985 if (chain == mEffectChains[i]) { 5986 mEffectChains.removeAt(i); 5987 // detach all active tracks from the chain 5988 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5989 sp<Track> track = mActiveTracks[i].promote(); 5990 if (track == 0) continue; 5991 if (session == track->sessionId()) { 5992 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5993 chain.get(), session); 5994 chain->decActiveTrackCnt(); 5995 } 5996 } 5997 5998 // detach all tracks with same session ID from this chain 5999 for (size_t i = 0; i < mTracks.size(); ++i) { 6000 sp<Track> track = mTracks[i]; 6001 if (session == track->sessionId()) { 6002 track->setMainBuffer(mMixBuffer); 6003 chain->decTrackCnt(); 6004 } 6005 } 6006 break; 6007 } 6008 } 6009 return mEffectChains.size(); 6010} 6011 6012status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6013 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6014{ 6015 Mutex::Autolock _l(mLock); 6016 return attachAuxEffect_l(track, EffectId); 6017} 6018 6019status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6020 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6021{ 6022 status_t status = NO_ERROR; 6023 6024 if (EffectId == 0) { 6025 track->setAuxBuffer(0, NULL); 6026 } else { 6027 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6028 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6029 if (effect != 0) { 6030 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6031 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6032 } else { 6033 status = INVALID_OPERATION; 6034 } 6035 } else { 6036 status = BAD_VALUE; 6037 } 6038 } 6039 return status; 6040} 6041 6042void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6043{ 6044 for (size_t i = 0; i < mTracks.size(); ++i) { 6045 sp<Track> track = mTracks[i]; 6046 if (track->auxEffectId() == effectId) { 6047 attachAuxEffect_l(track, 0); 6048 } 6049 } 6050} 6051 6052status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6053{ 6054 // only one chain per input thread 6055 if (mEffectChains.size() != 0) { 6056 return INVALID_OPERATION; 6057 } 6058 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6059 6060 chain->setInBuffer(NULL); 6061 chain->setOutBuffer(NULL); 6062 6063 checkSuspendOnAddEffectChain_l(chain); 6064 6065 mEffectChains.add(chain); 6066 6067 return NO_ERROR; 6068} 6069 6070size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6071{ 6072 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6073 ALOGW_IF(mEffectChains.size() != 1, 6074 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6075 chain.get(), mEffectChains.size(), this); 6076 if (mEffectChains.size() == 1) { 6077 mEffectChains.removeAt(0); 6078 } 6079 return 0; 6080} 6081 6082// ---------------------------------------------------------------------------- 6083// EffectModule implementation 6084// ---------------------------------------------------------------------------- 6085 6086#undef LOG_TAG 6087#define LOG_TAG "AudioFlinger::EffectModule" 6088 6089AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6090 const wp<AudioFlinger::EffectChain>& chain, 6091 effect_descriptor_t *desc, 6092 int id, 6093 int sessionId) 6094 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6095 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6096{ 6097 ALOGV("Constructor %p", this); 6098 int lStatus; 6099 sp<ThreadBase> thread = mThread.promote(); 6100 if (thread == 0) { 6101 return; 6102 } 6103 6104 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6105 6106 // create effect engine from effect factory 6107 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6108 6109 if (mStatus != NO_ERROR) { 6110 return; 6111 } 6112 lStatus = init(); 6113 if (lStatus < 0) { 6114 mStatus = lStatus; 6115 goto Error; 6116 } 6117 6118 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6119 mPinned = true; 6120 } 6121 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6122 return; 6123Error: 6124 EffectRelease(mEffectInterface); 6125 mEffectInterface = NULL; 6126 ALOGV("Constructor Error %d", mStatus); 6127} 6128 6129AudioFlinger::EffectModule::~EffectModule() 6130{ 6131 ALOGV("Destructor %p", this); 6132 if (mEffectInterface != NULL) { 6133 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6134 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6135 sp<ThreadBase> thread = mThread.promote(); 6136 if (thread != 0) { 6137 audio_stream_t *stream = thread->stream(); 6138 if (stream != NULL) { 6139 stream->remove_audio_effect(stream, mEffectInterface); 6140 } 6141 } 6142 } 6143 // release effect engine 6144 EffectRelease(mEffectInterface); 6145 } 6146} 6147 6148status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6149{ 6150 status_t status; 6151 6152 Mutex::Autolock _l(mLock); 6153 // First handle in mHandles has highest priority and controls the effect module 6154 int priority = handle->priority(); 6155 size_t size = mHandles.size(); 6156 sp<EffectHandle> h; 6157 size_t i; 6158 for (i = 0; i < size; i++) { 6159 h = mHandles[i].promote(); 6160 if (h == 0) continue; 6161 if (h->priority() <= priority) break; 6162 } 6163 // if inserted in first place, move effect control from previous owner to this handle 6164 if (i == 0) { 6165 bool enabled = false; 6166 if (h != 0) { 6167 enabled = h->enabled(); 6168 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6169 } 6170 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6171 status = NO_ERROR; 6172 } else { 6173 status = ALREADY_EXISTS; 6174 } 6175 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6176 mHandles.insertAt(handle, i); 6177 return status; 6178} 6179 6180size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6181{ 6182 Mutex::Autolock _l(mLock); 6183 size_t size = mHandles.size(); 6184 size_t i; 6185 for (i = 0; i < size; i++) { 6186 if (mHandles[i] == handle) break; 6187 } 6188 if (i == size) { 6189 return size; 6190 } 6191 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6192 6193 bool enabled = false; 6194 EffectHandle *hdl = handle.unsafe_get(); 6195 if (hdl) { 6196 ALOGV("removeHandle() unsafe_get OK"); 6197 enabled = hdl->enabled(); 6198 } 6199 mHandles.removeAt(i); 6200 size = mHandles.size(); 6201 // if removed from first place, move effect control from this handle to next in line 6202 if (i == 0 && size != 0) { 6203 sp<EffectHandle> h = mHandles[0].promote(); 6204 if (h != 0) { 6205 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6206 } 6207 } 6208 6209 // Prevent calls to process() and other functions on effect interface from now on. 6210 // The effect engine will be released by the destructor when the last strong reference on 6211 // this object is released which can happen after next process is called. 6212 if (size == 0 && !mPinned) { 6213 mState = DESTROYED; 6214 } 6215 6216 return size; 6217} 6218 6219sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6220{ 6221 Mutex::Autolock _l(mLock); 6222 sp<EffectHandle> handle; 6223 if (mHandles.size() != 0) { 6224 handle = mHandles[0].promote(); 6225 } 6226 return handle; 6227} 6228 6229void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6230{ 6231 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6232 // keep a strong reference on this EffectModule to avoid calling the 6233 // destructor before we exit 6234 sp<EffectModule> keep(this); 6235 { 6236 sp<ThreadBase> thread = mThread.promote(); 6237 if (thread != 0) { 6238 thread->disconnectEffect(keep, handle, unpiniflast); 6239 } 6240 } 6241} 6242 6243void AudioFlinger::EffectModule::updateState() { 6244 Mutex::Autolock _l(mLock); 6245 6246 switch (mState) { 6247 case RESTART: 6248 reset_l(); 6249 // FALL THROUGH 6250 6251 case STARTING: 6252 // clear auxiliary effect input buffer for next accumulation 6253 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6254 memset(mConfig.inputCfg.buffer.raw, 6255 0, 6256 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6257 } 6258 start_l(); 6259 mState = ACTIVE; 6260 break; 6261 case STOPPING: 6262 stop_l(); 6263 mDisableWaitCnt = mMaxDisableWaitCnt; 6264 mState = STOPPED; 6265 break; 6266 case STOPPED: 6267 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6268 // turn off sequence. 6269 if (--mDisableWaitCnt == 0) { 6270 reset_l(); 6271 mState = IDLE; 6272 } 6273 break; 6274 default: //IDLE , ACTIVE, DESTROYED 6275 break; 6276 } 6277} 6278 6279void AudioFlinger::EffectModule::process() 6280{ 6281 Mutex::Autolock _l(mLock); 6282 6283 if (mState == DESTROYED || mEffectInterface == NULL || 6284 mConfig.inputCfg.buffer.raw == NULL || 6285 mConfig.outputCfg.buffer.raw == NULL) { 6286 return; 6287 } 6288 6289 if (isProcessEnabled()) { 6290 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6291 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6292 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6293 mConfig.inputCfg.buffer.s32, 6294 mConfig.inputCfg.buffer.frameCount/2); 6295 } 6296 6297 // do the actual processing in the effect engine 6298 int ret = (*mEffectInterface)->process(mEffectInterface, 6299 &mConfig.inputCfg.buffer, 6300 &mConfig.outputCfg.buffer); 6301 6302 // force transition to IDLE state when engine is ready 6303 if (mState == STOPPED && ret == -ENODATA) { 6304 mDisableWaitCnt = 1; 6305 } 6306 6307 // clear auxiliary effect input buffer for next accumulation 6308 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6309 memset(mConfig.inputCfg.buffer.raw, 0, 6310 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6311 } 6312 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6313 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6314 // If an insert effect is idle and input buffer is different from output buffer, 6315 // accumulate input onto output 6316 sp<EffectChain> chain = mChain.promote(); 6317 if (chain != 0 && chain->activeTrackCnt() != 0) { 6318 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6319 int16_t *in = mConfig.inputCfg.buffer.s16; 6320 int16_t *out = mConfig.outputCfg.buffer.s16; 6321 for (size_t i = 0; i < frameCnt; i++) { 6322 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6323 } 6324 } 6325 } 6326} 6327 6328void AudioFlinger::EffectModule::reset_l() 6329{ 6330 if (mEffectInterface == NULL) { 6331 return; 6332 } 6333 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6334} 6335 6336status_t AudioFlinger::EffectModule::configure() 6337{ 6338 uint32_t channels; 6339 if (mEffectInterface == NULL) { 6340 return NO_INIT; 6341 } 6342 6343 sp<ThreadBase> thread = mThread.promote(); 6344 if (thread == 0) { 6345 return DEAD_OBJECT; 6346 } 6347 6348 // TODO: handle configuration of effects replacing track process 6349 if (thread->channelCount() == 1) { 6350 channels = AUDIO_CHANNEL_OUT_MONO; 6351 } else { 6352 channels = AUDIO_CHANNEL_OUT_STEREO; 6353 } 6354 6355 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6356 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6357 } else { 6358 mConfig.inputCfg.channels = channels; 6359 } 6360 mConfig.outputCfg.channels = channels; 6361 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6362 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6363 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6364 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6365 mConfig.inputCfg.bufferProvider.cookie = NULL; 6366 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6367 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6368 mConfig.outputCfg.bufferProvider.cookie = NULL; 6369 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6370 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6371 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6372 // Insert effect: 6373 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6374 // always overwrites output buffer: input buffer == output buffer 6375 // - in other sessions: 6376 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6377 // other effect: overwrites output buffer: input buffer == output buffer 6378 // Auxiliary effect: 6379 // accumulates in output buffer: input buffer != output buffer 6380 // Therefore: accumulate <=> input buffer != output buffer 6381 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6382 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6383 } else { 6384 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6385 } 6386 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6387 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6388 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6389 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6390 6391 ALOGV("configure() %p thread %p buffer %p framecount %d", 6392 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6393 6394 status_t cmdStatus; 6395 uint32_t size = sizeof(int); 6396 status_t status = (*mEffectInterface)->command(mEffectInterface, 6397 EFFECT_CMD_SET_CONFIG, 6398 sizeof(effect_config_t), 6399 &mConfig, 6400 &size, 6401 &cmdStatus); 6402 if (status == 0) { 6403 status = cmdStatus; 6404 } 6405 6406 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6407 (1000 * mConfig.outputCfg.buffer.frameCount); 6408 6409 return status; 6410} 6411 6412status_t AudioFlinger::EffectModule::init() 6413{ 6414 Mutex::Autolock _l(mLock); 6415 if (mEffectInterface == NULL) { 6416 return NO_INIT; 6417 } 6418 status_t cmdStatus; 6419 uint32_t size = sizeof(status_t); 6420 status_t status = (*mEffectInterface)->command(mEffectInterface, 6421 EFFECT_CMD_INIT, 6422 0, 6423 NULL, 6424 &size, 6425 &cmdStatus); 6426 if (status == 0) { 6427 status = cmdStatus; 6428 } 6429 return status; 6430} 6431 6432status_t AudioFlinger::EffectModule::start() 6433{ 6434 Mutex::Autolock _l(mLock); 6435 return start_l(); 6436} 6437 6438status_t AudioFlinger::EffectModule::start_l() 6439{ 6440 if (mEffectInterface == NULL) { 6441 return NO_INIT; 6442 } 6443 status_t cmdStatus; 6444 uint32_t size = sizeof(status_t); 6445 status_t status = (*mEffectInterface)->command(mEffectInterface, 6446 EFFECT_CMD_ENABLE, 6447 0, 6448 NULL, 6449 &size, 6450 &cmdStatus); 6451 if (status == 0) { 6452 status = cmdStatus; 6453 } 6454 if (status == 0 && 6455 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6456 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6457 sp<ThreadBase> thread = mThread.promote(); 6458 if (thread != 0) { 6459 audio_stream_t *stream = thread->stream(); 6460 if (stream != NULL) { 6461 stream->add_audio_effect(stream, mEffectInterface); 6462 } 6463 } 6464 } 6465 return status; 6466} 6467 6468status_t AudioFlinger::EffectModule::stop() 6469{ 6470 Mutex::Autolock _l(mLock); 6471 return stop_l(); 6472} 6473 6474status_t AudioFlinger::EffectModule::stop_l() 6475{ 6476 if (mEffectInterface == NULL) { 6477 return NO_INIT; 6478 } 6479 status_t cmdStatus; 6480 uint32_t size = sizeof(status_t); 6481 status_t status = (*mEffectInterface)->command(mEffectInterface, 6482 EFFECT_CMD_DISABLE, 6483 0, 6484 NULL, 6485 &size, 6486 &cmdStatus); 6487 if (status == 0) { 6488 status = cmdStatus; 6489 } 6490 if (status == 0 && 6491 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6492 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6493 sp<ThreadBase> thread = mThread.promote(); 6494 if (thread != 0) { 6495 audio_stream_t *stream = thread->stream(); 6496 if (stream != NULL) { 6497 stream->remove_audio_effect(stream, mEffectInterface); 6498 } 6499 } 6500 } 6501 return status; 6502} 6503 6504status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6505 uint32_t cmdSize, 6506 void *pCmdData, 6507 uint32_t *replySize, 6508 void *pReplyData) 6509{ 6510 Mutex::Autolock _l(mLock); 6511// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6512 6513 if (mState == DESTROYED || mEffectInterface == NULL) { 6514 return NO_INIT; 6515 } 6516 status_t status = (*mEffectInterface)->command(mEffectInterface, 6517 cmdCode, 6518 cmdSize, 6519 pCmdData, 6520 replySize, 6521 pReplyData); 6522 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6523 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6524 for (size_t i = 1; i < mHandles.size(); i++) { 6525 sp<EffectHandle> h = mHandles[i].promote(); 6526 if (h != 0) { 6527 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6528 } 6529 } 6530 } 6531 return status; 6532} 6533 6534status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6535{ 6536 6537 Mutex::Autolock _l(mLock); 6538 ALOGV("setEnabled %p enabled %d", this, enabled); 6539 6540 if (enabled != isEnabled()) { 6541 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6542 if (enabled && status != NO_ERROR) { 6543 return status; 6544 } 6545 6546 switch (mState) { 6547 // going from disabled to enabled 6548 case IDLE: 6549 mState = STARTING; 6550 break; 6551 case STOPPED: 6552 mState = RESTART; 6553 break; 6554 case STOPPING: 6555 mState = ACTIVE; 6556 break; 6557 6558 // going from enabled to disabled 6559 case RESTART: 6560 mState = STOPPED; 6561 break; 6562 case STARTING: 6563 mState = IDLE; 6564 break; 6565 case ACTIVE: 6566 mState = STOPPING; 6567 break; 6568 case DESTROYED: 6569 return NO_ERROR; // simply ignore as we are being destroyed 6570 } 6571 for (size_t i = 1; i < mHandles.size(); i++) { 6572 sp<EffectHandle> h = mHandles[i].promote(); 6573 if (h != 0) { 6574 h->setEnabled(enabled); 6575 } 6576 } 6577 } 6578 return NO_ERROR; 6579} 6580 6581bool AudioFlinger::EffectModule::isEnabled() 6582{ 6583 switch (mState) { 6584 case RESTART: 6585 case STARTING: 6586 case ACTIVE: 6587 return true; 6588 case IDLE: 6589 case STOPPING: 6590 case STOPPED: 6591 case DESTROYED: 6592 default: 6593 return false; 6594 } 6595} 6596 6597bool AudioFlinger::EffectModule::isProcessEnabled() 6598{ 6599 switch (mState) { 6600 case RESTART: 6601 case ACTIVE: 6602 case STOPPING: 6603 case STOPPED: 6604 return true; 6605 case IDLE: 6606 case STARTING: 6607 case DESTROYED: 6608 default: 6609 return false; 6610 } 6611} 6612 6613status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6614{ 6615 Mutex::Autolock _l(mLock); 6616 status_t status = NO_ERROR; 6617 6618 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6619 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6620 if (isProcessEnabled() && 6621 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6622 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6623 status_t cmdStatus; 6624 uint32_t volume[2]; 6625 uint32_t *pVolume = NULL; 6626 uint32_t size = sizeof(volume); 6627 volume[0] = *left; 6628 volume[1] = *right; 6629 if (controller) { 6630 pVolume = volume; 6631 } 6632 status = (*mEffectInterface)->command(mEffectInterface, 6633 EFFECT_CMD_SET_VOLUME, 6634 size, 6635 volume, 6636 &size, 6637 pVolume); 6638 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6639 *left = volume[0]; 6640 *right = volume[1]; 6641 } 6642 } 6643 return status; 6644} 6645 6646status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6647{ 6648 Mutex::Autolock _l(mLock); 6649 status_t status = NO_ERROR; 6650 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6651 // audio pre processing modules on RecordThread can receive both output and 6652 // input device indication in the same call 6653 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6654 if (dev) { 6655 status_t cmdStatus; 6656 uint32_t size = sizeof(status_t); 6657 6658 status = (*mEffectInterface)->command(mEffectInterface, 6659 EFFECT_CMD_SET_DEVICE, 6660 sizeof(uint32_t), 6661 &dev, 6662 &size, 6663 &cmdStatus); 6664 if (status == NO_ERROR) { 6665 status = cmdStatus; 6666 } 6667 } 6668 dev = device & AUDIO_DEVICE_IN_ALL; 6669 if (dev) { 6670 status_t cmdStatus; 6671 uint32_t size = sizeof(status_t); 6672 6673 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6674 EFFECT_CMD_SET_INPUT_DEVICE, 6675 sizeof(uint32_t), 6676 &dev, 6677 &size, 6678 &cmdStatus); 6679 if (status2 == NO_ERROR) { 6680 status2 = cmdStatus; 6681 } 6682 if (status == NO_ERROR) { 6683 status = status2; 6684 } 6685 } 6686 } 6687 return status; 6688} 6689 6690status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 6691{ 6692 Mutex::Autolock _l(mLock); 6693 status_t status = NO_ERROR; 6694 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6695 status_t cmdStatus; 6696 uint32_t size = sizeof(status_t); 6697 status = (*mEffectInterface)->command(mEffectInterface, 6698 EFFECT_CMD_SET_AUDIO_MODE, 6699 sizeof(int), 6700 &mode, 6701 &size, 6702 &cmdStatus); 6703 if (status == NO_ERROR) { 6704 status = cmdStatus; 6705 } 6706 } 6707 return status; 6708} 6709 6710void AudioFlinger::EffectModule::setSuspended(bool suspended) 6711{ 6712 Mutex::Autolock _l(mLock); 6713 mSuspended = suspended; 6714} 6715 6716bool AudioFlinger::EffectModule::suspended() const 6717{ 6718 Mutex::Autolock _l(mLock); 6719 return mSuspended; 6720} 6721 6722status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6723{ 6724 const size_t SIZE = 256; 6725 char buffer[SIZE]; 6726 String8 result; 6727 6728 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6729 result.append(buffer); 6730 6731 bool locked = tryLock(mLock); 6732 // failed to lock - AudioFlinger is probably deadlocked 6733 if (!locked) { 6734 result.append("\t\tCould not lock Fx mutex:\n"); 6735 } 6736 6737 result.append("\t\tSession Status State Engine:\n"); 6738 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6739 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6740 result.append(buffer); 6741 6742 result.append("\t\tDescriptor:\n"); 6743 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6744 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6745 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6746 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6747 result.append(buffer); 6748 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6749 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6750 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6751 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6752 result.append(buffer); 6753 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6754 mDescriptor.apiVersion, 6755 mDescriptor.flags); 6756 result.append(buffer); 6757 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6758 mDescriptor.name); 6759 result.append(buffer); 6760 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6761 mDescriptor.implementor); 6762 result.append(buffer); 6763 6764 result.append("\t\t- Input configuration:\n"); 6765 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6766 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6767 (uint32_t)mConfig.inputCfg.buffer.raw, 6768 mConfig.inputCfg.buffer.frameCount, 6769 mConfig.inputCfg.samplingRate, 6770 mConfig.inputCfg.channels, 6771 mConfig.inputCfg.format); 6772 result.append(buffer); 6773 6774 result.append("\t\t- Output configuration:\n"); 6775 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6776 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6777 (uint32_t)mConfig.outputCfg.buffer.raw, 6778 mConfig.outputCfg.buffer.frameCount, 6779 mConfig.outputCfg.samplingRate, 6780 mConfig.outputCfg.channels, 6781 mConfig.outputCfg.format); 6782 result.append(buffer); 6783 6784 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6785 result.append(buffer); 6786 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6787 for (size_t i = 0; i < mHandles.size(); ++i) { 6788 sp<EffectHandle> handle = mHandles[i].promote(); 6789 if (handle != 0) { 6790 handle->dump(buffer, SIZE); 6791 result.append(buffer); 6792 } 6793 } 6794 6795 result.append("\n"); 6796 6797 write(fd, result.string(), result.length()); 6798 6799 if (locked) { 6800 mLock.unlock(); 6801 } 6802 6803 return NO_ERROR; 6804} 6805 6806// ---------------------------------------------------------------------------- 6807// EffectHandle implementation 6808// ---------------------------------------------------------------------------- 6809 6810#undef LOG_TAG 6811#define LOG_TAG "AudioFlinger::EffectHandle" 6812 6813AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6814 const sp<AudioFlinger::Client>& client, 6815 const sp<IEffectClient>& effectClient, 6816 int32_t priority) 6817 : BnEffect(), 6818 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6819 mPriority(priority), mHasControl(false), mEnabled(false) 6820{ 6821 ALOGV("constructor %p", this); 6822 6823 if (client == 0) { 6824 return; 6825 } 6826 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6827 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6828 if (mCblkMemory != 0) { 6829 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6830 6831 if (mCblk) { 6832 new(mCblk) effect_param_cblk_t(); 6833 mBuffer = (uint8_t *)mCblk + bufOffset; 6834 } 6835 } else { 6836 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6837 return; 6838 } 6839} 6840 6841AudioFlinger::EffectHandle::~EffectHandle() 6842{ 6843 ALOGV("Destructor %p", this); 6844 disconnect(false); 6845 ALOGV("Destructor DONE %p", this); 6846} 6847 6848status_t AudioFlinger::EffectHandle::enable() 6849{ 6850 ALOGV("enable %p", this); 6851 if (!mHasControl) return INVALID_OPERATION; 6852 if (mEffect == 0) return DEAD_OBJECT; 6853 6854 if (mEnabled) { 6855 return NO_ERROR; 6856 } 6857 6858 mEnabled = true; 6859 6860 sp<ThreadBase> thread = mEffect->thread().promote(); 6861 if (thread != 0) { 6862 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6863 } 6864 6865 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6866 if (mEffect->suspended()) { 6867 return NO_ERROR; 6868 } 6869 6870 status_t status = mEffect->setEnabled(true); 6871 if (status != NO_ERROR) { 6872 if (thread != 0) { 6873 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6874 } 6875 mEnabled = false; 6876 } 6877 return status; 6878} 6879 6880status_t AudioFlinger::EffectHandle::disable() 6881{ 6882 ALOGV("disable %p", this); 6883 if (!mHasControl) return INVALID_OPERATION; 6884 if (mEffect == 0) return DEAD_OBJECT; 6885 6886 if (!mEnabled) { 6887 return NO_ERROR; 6888 } 6889 mEnabled = false; 6890 6891 if (mEffect->suspended()) { 6892 return NO_ERROR; 6893 } 6894 6895 status_t status = mEffect->setEnabled(false); 6896 6897 sp<ThreadBase> thread = mEffect->thread().promote(); 6898 if (thread != 0) { 6899 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6900 } 6901 6902 return status; 6903} 6904 6905void AudioFlinger::EffectHandle::disconnect() 6906{ 6907 disconnect(true); 6908} 6909 6910void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6911{ 6912 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6913 if (mEffect == 0) { 6914 return; 6915 } 6916 mEffect->disconnect(this, unpiniflast); 6917 6918 if (mHasControl && mEnabled) { 6919 sp<ThreadBase> thread = mEffect->thread().promote(); 6920 if (thread != 0) { 6921 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6922 } 6923 } 6924 6925 // release sp on module => module destructor can be called now 6926 mEffect.clear(); 6927 if (mClient != 0) { 6928 if (mCblk) { 6929 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6930 } 6931 mCblkMemory.clear(); // and free the shared memory 6932 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6933 mClient.clear(); 6934 } 6935} 6936 6937status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6938 uint32_t cmdSize, 6939 void *pCmdData, 6940 uint32_t *replySize, 6941 void *pReplyData) 6942{ 6943// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6944// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6945 6946 // only get parameter command is permitted for applications not controlling the effect 6947 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6948 return INVALID_OPERATION; 6949 } 6950 if (mEffect == 0) return DEAD_OBJECT; 6951 if (mClient == 0) return INVALID_OPERATION; 6952 6953 // handle commands that are not forwarded transparently to effect engine 6954 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6955 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6956 // no risk to block the whole media server process or mixer threads is we are stuck here 6957 Mutex::Autolock _l(mCblk->lock); 6958 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6959 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6960 mCblk->serverIndex = 0; 6961 mCblk->clientIndex = 0; 6962 return BAD_VALUE; 6963 } 6964 status_t status = NO_ERROR; 6965 while (mCblk->serverIndex < mCblk->clientIndex) { 6966 int reply; 6967 uint32_t rsize = sizeof(int); 6968 int *p = (int *)(mBuffer + mCblk->serverIndex); 6969 int size = *p++; 6970 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6971 ALOGW("command(): invalid parameter block size"); 6972 break; 6973 } 6974 effect_param_t *param = (effect_param_t *)p; 6975 if (param->psize == 0 || param->vsize == 0) { 6976 ALOGW("command(): null parameter or value size"); 6977 mCblk->serverIndex += size; 6978 continue; 6979 } 6980 uint32_t psize = sizeof(effect_param_t) + 6981 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6982 param->vsize; 6983 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6984 psize, 6985 p, 6986 &rsize, 6987 &reply); 6988 // stop at first error encountered 6989 if (ret != NO_ERROR) { 6990 status = ret; 6991 *(int *)pReplyData = reply; 6992 break; 6993 } else if (reply != NO_ERROR) { 6994 *(int *)pReplyData = reply; 6995 break; 6996 } 6997 mCblk->serverIndex += size; 6998 } 6999 mCblk->serverIndex = 0; 7000 mCblk->clientIndex = 0; 7001 return status; 7002 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7003 *(int *)pReplyData = NO_ERROR; 7004 return enable(); 7005 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7006 *(int *)pReplyData = NO_ERROR; 7007 return disable(); 7008 } 7009 7010 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7011} 7012 7013sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7014 return mCblkMemory; 7015} 7016 7017void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7018{ 7019 ALOGV("setControl %p control %d", this, hasControl); 7020 7021 mHasControl = hasControl; 7022 mEnabled = enabled; 7023 7024 if (signal && mEffectClient != 0) { 7025 mEffectClient->controlStatusChanged(hasControl); 7026 } 7027} 7028 7029void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7030 uint32_t cmdSize, 7031 void *pCmdData, 7032 uint32_t replySize, 7033 void *pReplyData) 7034{ 7035 if (mEffectClient != 0) { 7036 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7037 } 7038} 7039 7040 7041 7042void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7043{ 7044 if (mEffectClient != 0) { 7045 mEffectClient->enableStatusChanged(enabled); 7046 } 7047} 7048 7049status_t AudioFlinger::EffectHandle::onTransact( 7050 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7051{ 7052 return BnEffect::onTransact(code, data, reply, flags); 7053} 7054 7055 7056void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7057{ 7058 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7059 7060 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7061 (mClient == NULL) ? getpid() : mClient->pid(), 7062 mPriority, 7063 mHasControl, 7064 !locked, 7065 mCblk ? mCblk->clientIndex : 0, 7066 mCblk ? mCblk->serverIndex : 0 7067 ); 7068 7069 if (locked) { 7070 mCblk->lock.unlock(); 7071 } 7072} 7073 7074#undef LOG_TAG 7075#define LOG_TAG "AudioFlinger::EffectChain" 7076 7077AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7078 int sessionId) 7079 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7080 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7081 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7082{ 7083 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7084 sp<ThreadBase> thread = mThread.promote(); 7085 if (thread == 0) { 7086 return; 7087 } 7088 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7089 thread->frameCount(); 7090} 7091 7092AudioFlinger::EffectChain::~EffectChain() 7093{ 7094 if (mOwnInBuffer) { 7095 delete mInBuffer; 7096 } 7097 7098} 7099 7100// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7101sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7102{ 7103 sp<EffectModule> effect; 7104 size_t size = mEffects.size(); 7105 7106 for (size_t i = 0; i < size; i++) { 7107 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7108 effect = mEffects[i]; 7109 break; 7110 } 7111 } 7112 return effect; 7113} 7114 7115// getEffectFromId_l() must be called with ThreadBase::mLock held 7116sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7117{ 7118 sp<EffectModule> effect; 7119 size_t size = mEffects.size(); 7120 7121 for (size_t i = 0; i < size; i++) { 7122 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7123 if (id == 0 || mEffects[i]->id() == id) { 7124 effect = mEffects[i]; 7125 break; 7126 } 7127 } 7128 return effect; 7129} 7130 7131// getEffectFromType_l() must be called with ThreadBase::mLock held 7132sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7133 const effect_uuid_t *type) 7134{ 7135 sp<EffectModule> effect; 7136 size_t size = mEffects.size(); 7137 7138 for (size_t i = 0; i < size; i++) { 7139 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7140 effect = mEffects[i]; 7141 break; 7142 } 7143 } 7144 return effect; 7145} 7146 7147// Must be called with EffectChain::mLock locked 7148void AudioFlinger::EffectChain::process_l() 7149{ 7150 sp<ThreadBase> thread = mThread.promote(); 7151 if (thread == 0) { 7152 ALOGW("process_l(): cannot promote mixer thread"); 7153 return; 7154 } 7155 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7156 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7157 // always process effects unless no more tracks are on the session and the effect tail 7158 // has been rendered 7159 bool doProcess = true; 7160 if (!isGlobalSession) { 7161 bool tracksOnSession = (trackCnt() != 0); 7162 7163 if (!tracksOnSession && mTailBufferCount == 0) { 7164 doProcess = false; 7165 } 7166 7167 if (activeTrackCnt() == 0) { 7168 // if no track is active and the effect tail has not been rendered, 7169 // the input buffer must be cleared here as the mixer process will not do it 7170 if (tracksOnSession || mTailBufferCount > 0) { 7171 size_t numSamples = thread->frameCount() * thread->channelCount(); 7172 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7173 if (mTailBufferCount > 0) { 7174 mTailBufferCount--; 7175 } 7176 } 7177 } 7178 } 7179 7180 size_t size = mEffects.size(); 7181 if (doProcess) { 7182 for (size_t i = 0; i < size; i++) { 7183 mEffects[i]->process(); 7184 } 7185 } 7186 for (size_t i = 0; i < size; i++) { 7187 mEffects[i]->updateState(); 7188 } 7189} 7190 7191// addEffect_l() must be called with PlaybackThread::mLock held 7192status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7193{ 7194 effect_descriptor_t desc = effect->desc(); 7195 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7196 7197 Mutex::Autolock _l(mLock); 7198 effect->setChain(this); 7199 sp<ThreadBase> thread = mThread.promote(); 7200 if (thread == 0) { 7201 return NO_INIT; 7202 } 7203 effect->setThread(thread); 7204 7205 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7206 // Auxiliary effects are inserted at the beginning of mEffects vector as 7207 // they are processed first and accumulated in chain input buffer 7208 mEffects.insertAt(effect, 0); 7209 7210 // the input buffer for auxiliary effect contains mono samples in 7211 // 32 bit format. This is to avoid saturation in AudoMixer 7212 // accumulation stage. Saturation is done in EffectModule::process() before 7213 // calling the process in effect engine 7214 size_t numSamples = thread->frameCount(); 7215 int32_t *buffer = new int32_t[numSamples]; 7216 memset(buffer, 0, numSamples * sizeof(int32_t)); 7217 effect->setInBuffer((int16_t *)buffer); 7218 // auxiliary effects output samples to chain input buffer for further processing 7219 // by insert effects 7220 effect->setOutBuffer(mInBuffer); 7221 } else { 7222 // Insert effects are inserted at the end of mEffects vector as they are processed 7223 // after track and auxiliary effects. 7224 // Insert effect order as a function of indicated preference: 7225 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7226 // another effect is present 7227 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7228 // last effect claiming first position 7229 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7230 // first effect claiming last position 7231 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7232 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7233 // already present 7234 7235 int size = (int)mEffects.size(); 7236 int idx_insert = size; 7237 int idx_insert_first = -1; 7238 int idx_insert_last = -1; 7239 7240 for (int i = 0; i < size; i++) { 7241 effect_descriptor_t d = mEffects[i]->desc(); 7242 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7243 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7244 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7245 // check invalid effect chaining combinations 7246 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7247 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7248 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7249 return INVALID_OPERATION; 7250 } 7251 // remember position of first insert effect and by default 7252 // select this as insert position for new effect 7253 if (idx_insert == size) { 7254 idx_insert = i; 7255 } 7256 // remember position of last insert effect claiming 7257 // first position 7258 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7259 idx_insert_first = i; 7260 } 7261 // remember position of first insert effect claiming 7262 // last position 7263 if (iPref == EFFECT_FLAG_INSERT_LAST && 7264 idx_insert_last == -1) { 7265 idx_insert_last = i; 7266 } 7267 } 7268 } 7269 7270 // modify idx_insert from first position if needed 7271 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7272 if (idx_insert_last != -1) { 7273 idx_insert = idx_insert_last; 7274 } else { 7275 idx_insert = size; 7276 } 7277 } else { 7278 if (idx_insert_first != -1) { 7279 idx_insert = idx_insert_first + 1; 7280 } 7281 } 7282 7283 // always read samples from chain input buffer 7284 effect->setInBuffer(mInBuffer); 7285 7286 // if last effect in the chain, output samples to chain 7287 // output buffer, otherwise to chain input buffer 7288 if (idx_insert == size) { 7289 if (idx_insert != 0) { 7290 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7291 mEffects[idx_insert-1]->configure(); 7292 } 7293 effect->setOutBuffer(mOutBuffer); 7294 } else { 7295 effect->setOutBuffer(mInBuffer); 7296 } 7297 mEffects.insertAt(effect, idx_insert); 7298 7299 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7300 } 7301 effect->configure(); 7302 return NO_ERROR; 7303} 7304 7305// removeEffect_l() must be called with PlaybackThread::mLock held 7306size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7307{ 7308 Mutex::Autolock _l(mLock); 7309 int size = (int)mEffects.size(); 7310 int i; 7311 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7312 7313 for (i = 0; i < size; i++) { 7314 if (effect == mEffects[i]) { 7315 // calling stop here will remove pre-processing effect from the audio HAL. 7316 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7317 // the middle of a read from audio HAL 7318 if (mEffects[i]->state() == EffectModule::ACTIVE || 7319 mEffects[i]->state() == EffectModule::STOPPING) { 7320 mEffects[i]->stop(); 7321 } 7322 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7323 delete[] effect->inBuffer(); 7324 } else { 7325 if (i == size - 1 && i != 0) { 7326 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7327 mEffects[i - 1]->configure(); 7328 } 7329 } 7330 mEffects.removeAt(i); 7331 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7332 break; 7333 } 7334 } 7335 7336 return mEffects.size(); 7337} 7338 7339// setDevice_l() must be called with PlaybackThread::mLock held 7340void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7341{ 7342 size_t size = mEffects.size(); 7343 for (size_t i = 0; i < size; i++) { 7344 mEffects[i]->setDevice(device); 7345 } 7346} 7347 7348// setMode_l() must be called with PlaybackThread::mLock held 7349void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 7350{ 7351 size_t size = mEffects.size(); 7352 for (size_t i = 0; i < size; i++) { 7353 mEffects[i]->setMode(mode); 7354 } 7355} 7356 7357// setVolume_l() must be called with PlaybackThread::mLock held 7358bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7359{ 7360 uint32_t newLeft = *left; 7361 uint32_t newRight = *right; 7362 bool hasControl = false; 7363 int ctrlIdx = -1; 7364 size_t size = mEffects.size(); 7365 7366 // first update volume controller 7367 for (size_t i = size; i > 0; i--) { 7368 if (mEffects[i - 1]->isProcessEnabled() && 7369 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7370 ctrlIdx = i - 1; 7371 hasControl = true; 7372 break; 7373 } 7374 } 7375 7376 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7377 if (hasControl) { 7378 *left = mNewLeftVolume; 7379 *right = mNewRightVolume; 7380 } 7381 return hasControl; 7382 } 7383 7384 mVolumeCtrlIdx = ctrlIdx; 7385 mLeftVolume = newLeft; 7386 mRightVolume = newRight; 7387 7388 // second get volume update from volume controller 7389 if (ctrlIdx >= 0) { 7390 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7391 mNewLeftVolume = newLeft; 7392 mNewRightVolume = newRight; 7393 } 7394 // then indicate volume to all other effects in chain. 7395 // Pass altered volume to effects before volume controller 7396 // and requested volume to effects after controller 7397 uint32_t lVol = newLeft; 7398 uint32_t rVol = newRight; 7399 7400 for (size_t i = 0; i < size; i++) { 7401 if ((int)i == ctrlIdx) continue; 7402 // this also works for ctrlIdx == -1 when there is no volume controller 7403 if ((int)i > ctrlIdx) { 7404 lVol = *left; 7405 rVol = *right; 7406 } 7407 mEffects[i]->setVolume(&lVol, &rVol, false); 7408 } 7409 *left = newLeft; 7410 *right = newRight; 7411 7412 return hasControl; 7413} 7414 7415status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7416{ 7417 const size_t SIZE = 256; 7418 char buffer[SIZE]; 7419 String8 result; 7420 7421 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7422 result.append(buffer); 7423 7424 bool locked = tryLock(mLock); 7425 // failed to lock - AudioFlinger is probably deadlocked 7426 if (!locked) { 7427 result.append("\tCould not lock mutex:\n"); 7428 } 7429 7430 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7431 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7432 mEffects.size(), 7433 (uint32_t)mInBuffer, 7434 (uint32_t)mOutBuffer, 7435 mActiveTrackCnt); 7436 result.append(buffer); 7437 write(fd, result.string(), result.size()); 7438 7439 for (size_t i = 0; i < mEffects.size(); ++i) { 7440 sp<EffectModule> effect = mEffects[i]; 7441 if (effect != 0) { 7442 effect->dump(fd, args); 7443 } 7444 } 7445 7446 if (locked) { 7447 mLock.unlock(); 7448 } 7449 7450 return NO_ERROR; 7451} 7452 7453// must be called with ThreadBase::mLock held 7454void AudioFlinger::EffectChain::setEffectSuspended_l( 7455 const effect_uuid_t *type, bool suspend) 7456{ 7457 sp<SuspendedEffectDesc> desc; 7458 // use effect type UUID timelow as key as there is no real risk of identical 7459 // timeLow fields among effect type UUIDs. 7460 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7461 if (suspend) { 7462 if (index >= 0) { 7463 desc = mSuspendedEffects.valueAt(index); 7464 } else { 7465 desc = new SuspendedEffectDesc(); 7466 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7467 mSuspendedEffects.add(type->timeLow, desc); 7468 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7469 } 7470 if (desc->mRefCount++ == 0) { 7471 sp<EffectModule> effect = getEffectIfEnabled(type); 7472 if (effect != 0) { 7473 desc->mEffect = effect; 7474 effect->setSuspended(true); 7475 effect->setEnabled(false); 7476 } 7477 } 7478 } else { 7479 if (index < 0) { 7480 return; 7481 } 7482 desc = mSuspendedEffects.valueAt(index); 7483 if (desc->mRefCount <= 0) { 7484 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7485 desc->mRefCount = 1; 7486 } 7487 if (--desc->mRefCount == 0) { 7488 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7489 if (desc->mEffect != 0) { 7490 sp<EffectModule> effect = desc->mEffect.promote(); 7491 if (effect != 0) { 7492 effect->setSuspended(false); 7493 sp<EffectHandle> handle = effect->controlHandle(); 7494 if (handle != 0) { 7495 effect->setEnabled(handle->enabled()); 7496 } 7497 } 7498 desc->mEffect.clear(); 7499 } 7500 mSuspendedEffects.removeItemsAt(index); 7501 } 7502 } 7503} 7504 7505// must be called with ThreadBase::mLock held 7506void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7507{ 7508 sp<SuspendedEffectDesc> desc; 7509 7510 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7511 if (suspend) { 7512 if (index >= 0) { 7513 desc = mSuspendedEffects.valueAt(index); 7514 } else { 7515 desc = new SuspendedEffectDesc(); 7516 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7517 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7518 } 7519 if (desc->mRefCount++ == 0) { 7520 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7521 for (size_t i = 0; i < effects.size(); i++) { 7522 setEffectSuspended_l(&effects[i]->desc().type, true); 7523 } 7524 } 7525 } else { 7526 if (index < 0) { 7527 return; 7528 } 7529 desc = mSuspendedEffects.valueAt(index); 7530 if (desc->mRefCount <= 0) { 7531 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7532 desc->mRefCount = 1; 7533 } 7534 if (--desc->mRefCount == 0) { 7535 Vector<const effect_uuid_t *> types; 7536 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7537 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7538 continue; 7539 } 7540 types.add(&mSuspendedEffects.valueAt(i)->mType); 7541 } 7542 for (size_t i = 0; i < types.size(); i++) { 7543 setEffectSuspended_l(types[i], false); 7544 } 7545 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7546 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7547 } 7548 } 7549} 7550 7551 7552// The volume effect is used for automated tests only 7553#ifndef OPENSL_ES_H_ 7554static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7555 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7556const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7557#endif //OPENSL_ES_H_ 7558 7559bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7560{ 7561 // auxiliary effects and visualizer are never suspended on output mix 7562 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7563 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7564 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7565 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7566 return false; 7567 } 7568 return true; 7569} 7570 7571Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7572{ 7573 Vector< sp<EffectModule> > effects; 7574 for (size_t i = 0; i < mEffects.size(); i++) { 7575 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7576 continue; 7577 } 7578 effects.add(mEffects[i]); 7579 } 7580 return effects; 7581} 7582 7583sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7584 const effect_uuid_t *type) 7585{ 7586 sp<EffectModule> effect; 7587 effect = getEffectFromType_l(type); 7588 if (effect != 0 && !effect->isEnabled()) { 7589 effect.clear(); 7590 } 7591 return effect; 7592} 7593 7594void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7595 bool enabled) 7596{ 7597 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7598 if (enabled) { 7599 if (index < 0) { 7600 // if the effect is not suspend check if all effects are suspended 7601 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7602 if (index < 0) { 7603 return; 7604 } 7605 if (!isEffectEligibleForSuspend(effect->desc())) { 7606 return; 7607 } 7608 setEffectSuspended_l(&effect->desc().type, enabled); 7609 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7610 if (index < 0) { 7611 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7612 return; 7613 } 7614 } 7615 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7616 effect->desc().type.timeLow); 7617 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7618 // if effect is requested to suspended but was not yet enabled, supend it now. 7619 if (desc->mEffect == 0) { 7620 desc->mEffect = effect; 7621 effect->setEnabled(false); 7622 effect->setSuspended(true); 7623 } 7624 } else { 7625 if (index < 0) { 7626 return; 7627 } 7628 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7629 effect->desc().type.timeLow); 7630 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7631 desc->mEffect.clear(); 7632 effect->setSuspended(false); 7633 } 7634} 7635 7636#undef LOG_TAG 7637#define LOG_TAG "AudioFlinger" 7638 7639// ---------------------------------------------------------------------------- 7640 7641status_t AudioFlinger::onTransact( 7642 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7643{ 7644 return BnAudioFlinger::onTransact(code, data, reply, flags); 7645} 7646 7647}; // namespace android 7648