AudioFlinger.cpp revision 3b21c50ef95fe4e7ac3426ca14b365749e66ff08
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 39#include <media/AudioTrack.h> 40#include <media/AudioRecord.h> 41#include <media/IMediaPlayerService.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <audio_effects/effect_visualizer.h> 54#include <audio_effects/effect_ns.h> 55#include <audio_effects/effect_aec.h> 56 57#include <audio_utils/primitives.h> 58 59#include <cpustats/ThreadCpuUsage.h> 60#include <powermanager/PowerManager.h> 61// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 62 63// ---------------------------------------------------------------------------- 64 65 66namespace android { 67 68static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; 69static const char* kHardwareLockedString = "Hardware lock is taken\n"; 70 71//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 72static const float MAX_GAIN = 4096.0f; 73static const float MAX_GAIN_INT = 0x1000; 74 75// retry counts for buffer fill timeout 76// 50 * ~20msecs = 1 second 77static const int8_t kMaxTrackRetries = 50; 78static const int8_t kMaxTrackStartupRetries = 50; 79// allow less retry attempts on direct output thread. 80// direct outputs can be a scarce resource in audio hardware and should 81// be released as quickly as possible. 82static const int8_t kMaxTrackRetriesDirect = 2; 83 84static const int kDumpLockRetries = 50; 85static const int kDumpLockSleep = 20000; 86 87static const nsecs_t kWarningThrottle = seconds(5); 88 89// RecordThread loop sleep time upon application overrun or audio HAL read error 90static const int kRecordThreadSleepUs = 5000; 91 92static const nsecs_t kSetParametersTimeout = seconds(2); 93 94// minimum sleep time for the mixer thread loop when tracks are active but in underrun 95static const uint32_t kMinThreadSleepTimeUs = 5000; 96// maximum divider applied to the active sleep time in the mixer thread loop 97static const uint32_t kMaxThreadSleepTimeShift = 2; 98 99 100// ---------------------------------------------------------------------------- 101 102static bool recordingAllowed() { 103 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 104 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 105 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 106 return ok; 107} 108 109static bool settingsAllowed() { 110 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 111 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 112 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 113 return ok; 114} 115 116// To collect the amplifier usage 117static void addBatteryData(uint32_t params) { 118 sp<IBinder> binder = 119 defaultServiceManager()->getService(String16("media.player")); 120 sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder); 121 if (service.get() == NULL) { 122 LOGW("Cannot connect to the MediaPlayerService for battery tracking"); 123 return; 124 } 125 126 service->addBatteryData(params); 127} 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char *audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 164 mBtNrecIsOff(false) 165{ 166} 167 168void AudioFlinger::onFirstRef() 169{ 170 int rc = 0; 171 172 Mutex::Autolock _l(mLock); 173 174 /* TODO: move all this work into an Init() function */ 175 mHardwareStatus = AUDIO_HW_IDLE; 176 177 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 178 const hw_module_t *mod; 179 audio_hw_device_t *dev; 180 181 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 182 if (rc) 183 continue; 184 185 LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 186 mod->name, mod->id); 187 mAudioHwDevs.push(dev); 188 189 if (!mPrimaryHardwareDev) { 190 mPrimaryHardwareDev = dev; 191 LOGI("Using '%s' (%s.%s) as the primary audio interface", 192 mod->name, mod->id, audio_interfaces[i]); 193 } 194 } 195 196 mHardwareStatus = AUDIO_HW_INIT; 197 198 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 199 LOGE("Primary audio interface not found"); 200 return; 201 } 202 203 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 204 audio_hw_device_t *dev = mAudioHwDevs[i]; 205 206 mHardwareStatus = AUDIO_HW_INIT; 207 rc = dev->init_check(dev); 208 if (rc == 0) { 209 AutoMutex lock(mHardwareLock); 210 211 mMode = AUDIO_MODE_NORMAL; 212 mHardwareStatus = AUDIO_HW_SET_MODE; 213 dev->set_mode(dev, mMode); 214 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 215 dev->set_master_volume(dev, 1.0f); 216 mHardwareStatus = AUDIO_HW_IDLE; 217 } 218 } 219} 220 221status_t AudioFlinger::initCheck() const 222{ 223 Mutex::Autolock _l(mLock); 224 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 225 return NO_INIT; 226 return NO_ERROR; 227} 228 229AudioFlinger::~AudioFlinger() 230{ 231 int num_devs = mAudioHwDevs.size(); 232 233 while (!mRecordThreads.isEmpty()) { 234 // closeInput() will remove first entry from mRecordThreads 235 closeInput(mRecordThreads.keyAt(0)); 236 } 237 while (!mPlaybackThreads.isEmpty()) { 238 // closeOutput() will remove first entry from mPlaybackThreads 239 closeOutput(mPlaybackThreads.keyAt(0)); 240 } 241 242 for (int i = 0; i < num_devs; i++) { 243 audio_hw_device_t *dev = mAudioHwDevs[i]; 244 audio_hw_device_close(dev); 245 } 246 mAudioHwDevs.clear(); 247} 248 249audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 250{ 251 /* first matching HW device is returned */ 252 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 253 audio_hw_device_t *dev = mAudioHwDevs[i]; 254 if ((dev->get_supported_devices(dev) & devices) == devices) 255 return dev; 256 } 257 return NULL; 258} 259 260status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 261{ 262 const size_t SIZE = 256; 263 char buffer[SIZE]; 264 String8 result; 265 266 result.append("Clients:\n"); 267 for (size_t i = 0; i < mClients.size(); ++i) { 268 wp<Client> wClient = mClients.valueAt(i); 269 if (wClient != 0) { 270 sp<Client> client = wClient.promote(); 271 if (client != 0) { 272 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 273 result.append(buffer); 274 } 275 } 276 } 277 278 result.append("Global session refs:\n"); 279 result.append(" session pid cnt\n"); 280 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 281 AudioSessionRef *r = mAudioSessionRefs[i]; 282 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 283 result.append(buffer); 284 } 285 write(fd, result.string(), result.size()); 286 return NO_ERROR; 287} 288 289 290status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 291{ 292 const size_t SIZE = 256; 293 char buffer[SIZE]; 294 String8 result; 295 int hardwareStatus = mHardwareStatus; 296 297 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 298 result.append(buffer); 299 write(fd, result.string(), result.size()); 300 return NO_ERROR; 301} 302 303status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 304{ 305 const size_t SIZE = 256; 306 char buffer[SIZE]; 307 String8 result; 308 snprintf(buffer, SIZE, "Permission Denial: " 309 "can't dump AudioFlinger from pid=%d, uid=%d\n", 310 IPCThreadState::self()->getCallingPid(), 311 IPCThreadState::self()->getCallingUid()); 312 result.append(buffer); 313 write(fd, result.string(), result.size()); 314 return NO_ERROR; 315} 316 317static bool tryLock(Mutex& mutex) 318{ 319 bool locked = false; 320 for (int i = 0; i < kDumpLockRetries; ++i) { 321 if (mutex.tryLock() == NO_ERROR) { 322 locked = true; 323 break; 324 } 325 usleep(kDumpLockSleep); 326 } 327 return locked; 328} 329 330status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 331{ 332 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 333 dumpPermissionDenial(fd, args); 334 } else { 335 // get state of hardware lock 336 bool hardwareLocked = tryLock(mHardwareLock); 337 if (!hardwareLocked) { 338 String8 result(kHardwareLockedString); 339 write(fd, result.string(), result.size()); 340 } else { 341 mHardwareLock.unlock(); 342 } 343 344 bool locked = tryLock(mLock); 345 346 // failed to lock - AudioFlinger is probably deadlocked 347 if (!locked) { 348 String8 result(kDeadlockedString); 349 write(fd, result.string(), result.size()); 350 } 351 352 dumpClients(fd, args); 353 dumpInternals(fd, args); 354 355 // dump playback threads 356 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 357 mPlaybackThreads.valueAt(i)->dump(fd, args); 358 } 359 360 // dump record threads 361 for (size_t i = 0; i < mRecordThreads.size(); i++) { 362 mRecordThreads.valueAt(i)->dump(fd, args); 363 } 364 365 // dump all hardware devs 366 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 367 audio_hw_device_t *dev = mAudioHwDevs[i]; 368 dev->dump(dev, fd); 369 } 370 if (locked) mLock.unlock(); 371 } 372 return NO_ERROR; 373} 374 375 376// IAudioFlinger interface 377 378 379sp<IAudioTrack> AudioFlinger::createTrack( 380 pid_t pid, 381 int streamType, 382 uint32_t sampleRate, 383 uint32_t format, 384 uint32_t channelMask, 385 int frameCount, 386 uint32_t flags, 387 const sp<IMemory>& sharedBuffer, 388 int output, 389 int *sessionId, 390 status_t *status) 391{ 392 sp<PlaybackThread::Track> track; 393 sp<TrackHandle> trackHandle; 394 sp<Client> client; 395 wp<Client> wclient; 396 status_t lStatus; 397 int lSessionId; 398 399 if (streamType >= AUDIO_STREAM_CNT) { 400 LOGE("invalid stream type"); 401 lStatus = BAD_VALUE; 402 goto Exit; 403 } 404 405 { 406 Mutex::Autolock _l(mLock); 407 PlaybackThread *thread = checkPlaybackThread_l(output); 408 PlaybackThread *effectThread = NULL; 409 if (thread == NULL) { 410 LOGE("unknown output thread"); 411 lStatus = BAD_VALUE; 412 goto Exit; 413 } 414 415 wclient = mClients.valueFor(pid); 416 417 if (wclient != NULL) { 418 client = wclient.promote(); 419 } else { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 425 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 426 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 427 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 428 if (mPlaybackThreads.keyAt(i) != output) { 429 // prevent same audio session on different output threads 430 uint32_t sessions = t->hasAudioSession(*sessionId); 431 if (sessions & PlaybackThread::TRACK_SESSION) { 432 lStatus = BAD_VALUE; 433 goto Exit; 434 } 435 // check if an effect with same session ID is waiting for a track to be created 436 if (sessions & PlaybackThread::EFFECT_SESSION) { 437 effectThread = t.get(); 438 } 439 } 440 } 441 lSessionId = *sessionId; 442 } else { 443 // if no audio session id is provided, create one here 444 lSessionId = nextUniqueId(); 445 if (sessionId != NULL) { 446 *sessionId = lSessionId; 447 } 448 } 449 ALOGV("createTrack() lSessionId: %d", lSessionId); 450 451 track = thread->createTrack_l(client, streamType, sampleRate, format, 452 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 453 454 // move effect chain to this output thread if an effect on same session was waiting 455 // for a track to be created 456 if (lStatus == NO_ERROR && effectThread != NULL) { 457 Mutex::Autolock _dl(thread->mLock); 458 Mutex::Autolock _sl(effectThread->mLock); 459 moveEffectChain_l(lSessionId, effectThread, thread, true); 460 } 461 } 462 if (lStatus == NO_ERROR) { 463 trackHandle = new TrackHandle(track); 464 } else { 465 // remove local strong reference to Client before deleting the Track so that the Client 466 // destructor is called by the TrackBase destructor with mLock held 467 client.clear(); 468 track.clear(); 469 } 470 471Exit: 472 if(status) { 473 *status = lStatus; 474 } 475 return trackHandle; 476} 477 478uint32_t AudioFlinger::sampleRate(int output) const 479{ 480 Mutex::Autolock _l(mLock); 481 PlaybackThread *thread = checkPlaybackThread_l(output); 482 if (thread == NULL) { 483 LOGW("sampleRate() unknown thread %d", output); 484 return 0; 485 } 486 return thread->sampleRate(); 487} 488 489int AudioFlinger::channelCount(int output) const 490{ 491 Mutex::Autolock _l(mLock); 492 PlaybackThread *thread = checkPlaybackThread_l(output); 493 if (thread == NULL) { 494 LOGW("channelCount() unknown thread %d", output); 495 return 0; 496 } 497 return thread->channelCount(); 498} 499 500uint32_t AudioFlinger::format(int output) const 501{ 502 Mutex::Autolock _l(mLock); 503 PlaybackThread *thread = checkPlaybackThread_l(output); 504 if (thread == NULL) { 505 LOGW("format() unknown thread %d", output); 506 return 0; 507 } 508 return thread->format(); 509} 510 511size_t AudioFlinger::frameCount(int output) const 512{ 513 Mutex::Autolock _l(mLock); 514 PlaybackThread *thread = checkPlaybackThread_l(output); 515 if (thread == NULL) { 516 LOGW("frameCount() unknown thread %d", output); 517 return 0; 518 } 519 return thread->frameCount(); 520} 521 522uint32_t AudioFlinger::latency(int output) const 523{ 524 Mutex::Autolock _l(mLock); 525 PlaybackThread *thread = checkPlaybackThread_l(output); 526 if (thread == NULL) { 527 LOGW("latency() unknown thread %d", output); 528 return 0; 529 } 530 return thread->latency(); 531} 532 533status_t AudioFlinger::setMasterVolume(float value) 534{ 535 status_t ret = initCheck(); 536 if (ret != NO_ERROR) { 537 return ret; 538 } 539 540 // check calling permissions 541 if (!settingsAllowed()) { 542 return PERMISSION_DENIED; 543 } 544 545 // when hw supports master volume, don't scale in sw mixer 546 { // scope for the lock 547 AutoMutex lock(mHardwareLock); 548 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 549 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 550 value = 1.0f; 551 } 552 mHardwareStatus = AUDIO_HW_IDLE; 553 } 554 555 Mutex::Autolock _l(mLock); 556 mMasterVolume = value; 557 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 558 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 559 560 return NO_ERROR; 561} 562 563status_t AudioFlinger::setMode(int mode) 564{ 565 status_t ret = initCheck(); 566 if (ret != NO_ERROR) { 567 return ret; 568 } 569 570 // check calling permissions 571 if (!settingsAllowed()) { 572 return PERMISSION_DENIED; 573 } 574 if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) { 575 LOGW("Illegal value: setMode(%d)", mode); 576 return BAD_VALUE; 577 } 578 579 { // scope for the lock 580 AutoMutex lock(mHardwareLock); 581 mHardwareStatus = AUDIO_HW_SET_MODE; 582 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 583 mHardwareStatus = AUDIO_HW_IDLE; 584 } 585 586 if (NO_ERROR == ret) { 587 Mutex::Autolock _l(mLock); 588 mMode = mode; 589 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 590 mPlaybackThreads.valueAt(i)->setMode(mode); 591 } 592 593 return ret; 594} 595 596status_t AudioFlinger::setMicMute(bool state) 597{ 598 status_t ret = initCheck(); 599 if (ret != NO_ERROR) { 600 return ret; 601 } 602 603 // check calling permissions 604 if (!settingsAllowed()) { 605 return PERMISSION_DENIED; 606 } 607 608 AutoMutex lock(mHardwareLock); 609 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 610 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 611 mHardwareStatus = AUDIO_HW_IDLE; 612 return ret; 613} 614 615bool AudioFlinger::getMicMute() const 616{ 617 status_t ret = initCheck(); 618 if (ret != NO_ERROR) { 619 return false; 620 } 621 622 bool state = AUDIO_MODE_INVALID; 623 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 624 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 625 mHardwareStatus = AUDIO_HW_IDLE; 626 return state; 627} 628 629status_t AudioFlinger::setMasterMute(bool muted) 630{ 631 // check calling permissions 632 if (!settingsAllowed()) { 633 return PERMISSION_DENIED; 634 } 635 636 Mutex::Autolock _l(mLock); 637 mMasterMute = muted; 638 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 639 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 640 641 return NO_ERROR; 642} 643 644float AudioFlinger::masterVolume() const 645{ 646 return mMasterVolume; 647} 648 649bool AudioFlinger::masterMute() const 650{ 651 return mMasterMute; 652} 653 654status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 655{ 656 // check calling permissions 657 if (!settingsAllowed()) { 658 return PERMISSION_DENIED; 659 } 660 661 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 662 return BAD_VALUE; 663 } 664 665 AutoMutex lock(mLock); 666 PlaybackThread *thread = NULL; 667 if (output) { 668 thread = checkPlaybackThread_l(output); 669 if (thread == NULL) { 670 return BAD_VALUE; 671 } 672 } 673 674 mStreamTypes[stream].volume = value; 675 676 if (thread == NULL) { 677 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 678 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 679 } 680 } else { 681 thread->setStreamVolume(stream, value); 682 } 683 684 return NO_ERROR; 685} 686 687status_t AudioFlinger::setStreamMute(int stream, bool muted) 688{ 689 // check calling permissions 690 if (!settingsAllowed()) { 691 return PERMISSION_DENIED; 692 } 693 694 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || 695 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 696 return BAD_VALUE; 697 } 698 699 AutoMutex lock(mLock); 700 mStreamTypes[stream].mute = muted; 701 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 702 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 703 704 return NO_ERROR; 705} 706 707float AudioFlinger::streamVolume(int stream, int output) const 708{ 709 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 710 return 0.0f; 711 } 712 713 AutoMutex lock(mLock); 714 float volume; 715 if (output) { 716 PlaybackThread *thread = checkPlaybackThread_l(output); 717 if (thread == NULL) { 718 return 0.0f; 719 } 720 volume = thread->streamVolume(stream); 721 } else { 722 volume = mStreamTypes[stream].volume; 723 } 724 725 return volume; 726} 727 728bool AudioFlinger::streamMute(int stream) const 729{ 730 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 731 return true; 732 } 733 734 return mStreamTypes[stream].mute; 735} 736 737status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 738{ 739 status_t result; 740 741 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 742 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 743 // check calling permissions 744 if (!settingsAllowed()) { 745 return PERMISSION_DENIED; 746 } 747 748 // ioHandle == 0 means the parameters are global to the audio hardware interface 749 if (ioHandle == 0) { 750 AutoMutex lock(mHardwareLock); 751 mHardwareStatus = AUDIO_SET_PARAMETER; 752 status_t final_result = NO_ERROR; 753 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 754 audio_hw_device_t *dev = mAudioHwDevs[i]; 755 result = dev->set_parameters(dev, keyValuePairs.string()); 756 final_result = result ?: final_result; 757 } 758 mHardwareStatus = AUDIO_HW_IDLE; 759 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 760 AudioParameter param = AudioParameter(keyValuePairs); 761 String8 value; 762 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 763 Mutex::Autolock _l(mLock); 764 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 765 if (mBtNrecIsOff != btNrecIsOff) { 766 for (size_t i = 0; i < mRecordThreads.size(); i++) { 767 sp<RecordThread> thread = mRecordThreads.valueAt(i); 768 RecordThread::RecordTrack *track = thread->track(); 769 if (track != NULL) { 770 audio_devices_t device = (audio_devices_t)( 771 thread->device() & AUDIO_DEVICE_IN_ALL); 772 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 773 thread->setEffectSuspended(FX_IID_AEC, 774 suspend, 775 track->sessionId()); 776 thread->setEffectSuspended(FX_IID_NS, 777 suspend, 778 track->sessionId()); 779 } 780 } 781 mBtNrecIsOff = btNrecIsOff; 782 } 783 } 784 return final_result; 785 } 786 787 // hold a strong ref on thread in case closeOutput() or closeInput() is called 788 // and the thread is exited once the lock is released 789 sp<ThreadBase> thread; 790 { 791 Mutex::Autolock _l(mLock); 792 thread = checkPlaybackThread_l(ioHandle); 793 if (thread == NULL) { 794 thread = checkRecordThread_l(ioHandle); 795 } else if (thread.get() == primaryPlaybackThread_l()) { 796 // indicate output device change to all input threads for pre processing 797 AudioParameter param = AudioParameter(keyValuePairs); 798 int value; 799 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 800 for (size_t i = 0; i < mRecordThreads.size(); i++) { 801 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 802 } 803 } 804 } 805 } 806 if (thread != NULL) { 807 result = thread->setParameters(keyValuePairs); 808 return result; 809 } 810 return BAD_VALUE; 811} 812 813String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 814{ 815// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 816// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 817 818 if (ioHandle == 0) { 819 String8 out_s8; 820 821 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 822 audio_hw_device_t *dev = mAudioHwDevs[i]; 823 char *s = dev->get_parameters(dev, keys.string()); 824 out_s8 += String8(s); 825 free(s); 826 } 827 return out_s8; 828 } 829 830 Mutex::Autolock _l(mLock); 831 832 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 833 if (playbackThread != NULL) { 834 return playbackThread->getParameters(keys); 835 } 836 RecordThread *recordThread = checkRecordThread_l(ioHandle); 837 if (recordThread != NULL) { 838 return recordThread->getParameters(keys); 839 } 840 return String8(""); 841} 842 843size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 844{ 845 status_t ret = initCheck(); 846 if (ret != NO_ERROR) { 847 return 0; 848 } 849 850 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 851} 852 853unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 854{ 855 if (ioHandle == 0) { 856 return 0; 857 } 858 859 Mutex::Autolock _l(mLock); 860 861 RecordThread *recordThread = checkRecordThread_l(ioHandle); 862 if (recordThread != NULL) { 863 return recordThread->getInputFramesLost(); 864 } 865 return 0; 866} 867 868status_t AudioFlinger::setVoiceVolume(float value) 869{ 870 status_t ret = initCheck(); 871 if (ret != NO_ERROR) { 872 return ret; 873 } 874 875 // check calling permissions 876 if (!settingsAllowed()) { 877 return PERMISSION_DENIED; 878 } 879 880 AutoMutex lock(mHardwareLock); 881 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 882 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 883 mHardwareStatus = AUDIO_HW_IDLE; 884 885 return ret; 886} 887 888status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 889{ 890 status_t status; 891 892 Mutex::Autolock _l(mLock); 893 894 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 895 if (playbackThread != NULL) { 896 return playbackThread->getRenderPosition(halFrames, dspFrames); 897 } 898 899 return BAD_VALUE; 900} 901 902void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 903{ 904 905 Mutex::Autolock _l(mLock); 906 907 int pid = IPCThreadState::self()->getCallingPid(); 908 if (mNotificationClients.indexOfKey(pid) < 0) { 909 sp<NotificationClient> notificationClient = new NotificationClient(this, 910 client, 911 pid); 912 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 913 914 mNotificationClients.add(pid, notificationClient); 915 916 sp<IBinder> binder = client->asBinder(); 917 binder->linkToDeath(notificationClient); 918 919 // the config change is always sent from playback or record threads to avoid deadlock 920 // with AudioSystem::gLock 921 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 922 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 923 } 924 925 for (size_t i = 0; i < mRecordThreads.size(); i++) { 926 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 927 } 928 } 929} 930 931void AudioFlinger::removeNotificationClient(pid_t pid) 932{ 933 Mutex::Autolock _l(mLock); 934 935 int index = mNotificationClients.indexOfKey(pid); 936 if (index >= 0) { 937 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 938 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 939 mNotificationClients.removeItem(pid); 940 } 941 942 ALOGV("%d died, releasing its sessions", pid); 943 int num = mAudioSessionRefs.size(); 944 bool removed = false; 945 for (int i = 0; i< num; i++) { 946 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 947 ALOGV(" pid %d @ %d", ref->pid, i); 948 if (ref->pid == pid) { 949 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 950 mAudioSessionRefs.removeAt(i); 951 delete ref; 952 removed = true; 953 i--; 954 num--; 955 } 956 } 957 if (removed) { 958 purgeStaleEffects_l(); 959 } 960} 961 962// audioConfigChanged_l() must be called with AudioFlinger::mLock held 963void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 964{ 965 size_t size = mNotificationClients.size(); 966 for (size_t i = 0; i < size; i++) { 967 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 968 } 969} 970 971// removeClient_l() must be called with AudioFlinger::mLock held 972void AudioFlinger::removeClient_l(pid_t pid) 973{ 974 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 975 mClients.removeItem(pid); 976} 977 978 979// ---------------------------------------------------------------------------- 980 981AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 982 : Thread(false), 983 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 984 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), 985 mDevice(device) 986{ 987 mDeathRecipient = new PMDeathRecipient(this); 988} 989 990AudioFlinger::ThreadBase::~ThreadBase() 991{ 992 mParamCond.broadcast(); 993 mNewParameters.clear(); 994 // do not lock the mutex in destructor 995 releaseWakeLock_l(); 996 if (mPowerManager != 0) { 997 sp<IBinder> binder = mPowerManager->asBinder(); 998 binder->unlinkToDeath(mDeathRecipient); 999 } 1000} 1001 1002void AudioFlinger::ThreadBase::exit() 1003{ 1004 // keep a strong ref on ourself so that we wont get 1005 // destroyed in the middle of requestExitAndWait() 1006 sp <ThreadBase> strongMe = this; 1007 1008 ALOGV("ThreadBase::exit"); 1009 { 1010 AutoMutex lock(&mLock); 1011 mExiting = true; 1012 requestExit(); 1013 mWaitWorkCV.signal(); 1014 } 1015 requestExitAndWait(); 1016} 1017 1018uint32_t AudioFlinger::ThreadBase::sampleRate() const 1019{ 1020 return mSampleRate; 1021} 1022 1023int AudioFlinger::ThreadBase::channelCount() const 1024{ 1025 return (int)mChannelCount; 1026} 1027 1028uint32_t AudioFlinger::ThreadBase::format() const 1029{ 1030 return mFormat; 1031} 1032 1033size_t AudioFlinger::ThreadBase::frameCount() const 1034{ 1035 return mFrameCount; 1036} 1037 1038status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1039{ 1040 status_t status; 1041 1042 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1043 Mutex::Autolock _l(mLock); 1044 1045 mNewParameters.add(keyValuePairs); 1046 mWaitWorkCV.signal(); 1047 // wait condition with timeout in case the thread loop has exited 1048 // before the request could be processed 1049 if (mParamCond.waitRelative(mLock, kSetParametersTimeout) == NO_ERROR) { 1050 status = mParamStatus; 1051 mWaitWorkCV.signal(); 1052 } else { 1053 status = TIMED_OUT; 1054 } 1055 return status; 1056} 1057 1058void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1059{ 1060 Mutex::Autolock _l(mLock); 1061 sendConfigEvent_l(event, param); 1062} 1063 1064// sendConfigEvent_l() must be called with ThreadBase::mLock held 1065void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1066{ 1067 ConfigEvent *configEvent = new ConfigEvent(); 1068 configEvent->mEvent = event; 1069 configEvent->mParam = param; 1070 mConfigEvents.add(configEvent); 1071 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1072 mWaitWorkCV.signal(); 1073} 1074 1075void AudioFlinger::ThreadBase::processConfigEvents() 1076{ 1077 mLock.lock(); 1078 while(!mConfigEvents.isEmpty()) { 1079 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1080 ConfigEvent *configEvent = mConfigEvents[0]; 1081 mConfigEvents.removeAt(0); 1082 // release mLock before locking AudioFlinger mLock: lock order is always 1083 // AudioFlinger then ThreadBase to avoid cross deadlock 1084 mLock.unlock(); 1085 mAudioFlinger->mLock.lock(); 1086 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 1087 mAudioFlinger->mLock.unlock(); 1088 delete configEvent; 1089 mLock.lock(); 1090 } 1091 mLock.unlock(); 1092} 1093 1094status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1095{ 1096 const size_t SIZE = 256; 1097 char buffer[SIZE]; 1098 String8 result; 1099 1100 bool locked = tryLock(mLock); 1101 if (!locked) { 1102 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1103 write(fd, buffer, strlen(buffer)); 1104 } 1105 1106 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1107 result.append(buffer); 1108 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1109 result.append(buffer); 1110 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1111 result.append(buffer); 1112 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1113 result.append(buffer); 1114 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1115 result.append(buffer); 1116 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1117 result.append(buffer); 1118 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 1119 result.append(buffer); 1120 1121 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1122 result.append(buffer); 1123 result.append(" Index Command"); 1124 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1125 snprintf(buffer, SIZE, "\n %02d ", i); 1126 result.append(buffer); 1127 result.append(mNewParameters[i]); 1128 } 1129 1130 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1131 result.append(buffer); 1132 snprintf(buffer, SIZE, " Index event param\n"); 1133 result.append(buffer); 1134 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1135 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 1136 result.append(buffer); 1137 } 1138 result.append("\n"); 1139 1140 write(fd, result.string(), result.size()); 1141 1142 if (locked) { 1143 mLock.unlock(); 1144 } 1145 return NO_ERROR; 1146} 1147 1148status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1149{ 1150 const size_t SIZE = 256; 1151 char buffer[SIZE]; 1152 String8 result; 1153 1154 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1155 write(fd, buffer, strlen(buffer)); 1156 1157 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1158 sp<EffectChain> chain = mEffectChains[i]; 1159 if (chain != 0) { 1160 chain->dump(fd, args); 1161 } 1162 } 1163 return NO_ERROR; 1164} 1165 1166void AudioFlinger::ThreadBase::acquireWakeLock() 1167{ 1168 Mutex::Autolock _l(mLock); 1169 acquireWakeLock_l(); 1170} 1171 1172void AudioFlinger::ThreadBase::acquireWakeLock_l() 1173{ 1174 if (mPowerManager == 0) { 1175 // use checkService() to avoid blocking if power service is not up yet 1176 sp<IBinder> binder = 1177 defaultServiceManager()->checkService(String16("power")); 1178 if (binder == 0) { 1179 LOGW("Thread %s cannot connect to the power manager service", mName); 1180 } else { 1181 mPowerManager = interface_cast<IPowerManager>(binder); 1182 binder->linkToDeath(mDeathRecipient); 1183 } 1184 } 1185 if (mPowerManager != 0) { 1186 sp<IBinder> binder = new BBinder(); 1187 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1188 binder, 1189 String16(mName)); 1190 if (status == NO_ERROR) { 1191 mWakeLockToken = binder; 1192 } 1193 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1194 } 1195} 1196 1197void AudioFlinger::ThreadBase::releaseWakeLock() 1198{ 1199 Mutex::Autolock _l(mLock); 1200 releaseWakeLock_l(); 1201} 1202 1203void AudioFlinger::ThreadBase::releaseWakeLock_l() 1204{ 1205 if (mWakeLockToken != 0) { 1206 ALOGV("releaseWakeLock_l() %s", mName); 1207 if (mPowerManager != 0) { 1208 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1209 } 1210 mWakeLockToken.clear(); 1211 } 1212} 1213 1214void AudioFlinger::ThreadBase::clearPowerManager() 1215{ 1216 Mutex::Autolock _l(mLock); 1217 releaseWakeLock_l(); 1218 mPowerManager.clear(); 1219} 1220 1221void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1222{ 1223 sp<ThreadBase> thread = mThread.promote(); 1224 if (thread != 0) { 1225 thread->clearPowerManager(); 1226 } 1227 LOGW("power manager service died !!!"); 1228} 1229 1230void AudioFlinger::ThreadBase::setEffectSuspended( 1231 const effect_uuid_t *type, bool suspend, int sessionId) 1232{ 1233 Mutex::Autolock _l(mLock); 1234 setEffectSuspended_l(type, suspend, sessionId); 1235} 1236 1237void AudioFlinger::ThreadBase::setEffectSuspended_l( 1238 const effect_uuid_t *type, bool suspend, int sessionId) 1239{ 1240 sp<EffectChain> chain; 1241 chain = getEffectChain_l(sessionId); 1242 if (chain != 0) { 1243 if (type != NULL) { 1244 chain->setEffectSuspended_l(type, suspend); 1245 } else { 1246 chain->setEffectSuspendedAll_l(suspend); 1247 } 1248 } 1249 1250 updateSuspendedSessions_l(type, suspend, sessionId); 1251} 1252 1253void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1254{ 1255 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1256 if (index < 0) { 1257 return; 1258 } 1259 1260 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1261 mSuspendedSessions.editValueAt(index); 1262 1263 for (size_t i = 0; i < sessionEffects.size(); i++) { 1264 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1265 for (int j = 0; j < desc->mRefCount; j++) { 1266 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1267 chain->setEffectSuspendedAll_l(true); 1268 } else { 1269 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1270 desc->mType.timeLow); 1271 chain->setEffectSuspended_l(&desc->mType, true); 1272 } 1273 } 1274 } 1275} 1276 1277void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1278 bool suspend, 1279 int sessionId) 1280{ 1281 int index = mSuspendedSessions.indexOfKey(sessionId); 1282 1283 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1284 1285 if (suspend) { 1286 if (index >= 0) { 1287 sessionEffects = mSuspendedSessions.editValueAt(index); 1288 } else { 1289 mSuspendedSessions.add(sessionId, sessionEffects); 1290 } 1291 } else { 1292 if (index < 0) { 1293 return; 1294 } 1295 sessionEffects = mSuspendedSessions.editValueAt(index); 1296 } 1297 1298 1299 int key = EffectChain::kKeyForSuspendAll; 1300 if (type != NULL) { 1301 key = type->timeLow; 1302 } 1303 index = sessionEffects.indexOfKey(key); 1304 1305 sp <SuspendedSessionDesc> desc; 1306 if (suspend) { 1307 if (index >= 0) { 1308 desc = sessionEffects.valueAt(index); 1309 } else { 1310 desc = new SuspendedSessionDesc(); 1311 if (type != NULL) { 1312 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1313 } 1314 sessionEffects.add(key, desc); 1315 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1316 } 1317 desc->mRefCount++; 1318 } else { 1319 if (index < 0) { 1320 return; 1321 } 1322 desc = sessionEffects.valueAt(index); 1323 if (--desc->mRefCount == 0) { 1324 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1325 sessionEffects.removeItemsAt(index); 1326 if (sessionEffects.isEmpty()) { 1327 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1328 sessionId); 1329 mSuspendedSessions.removeItem(sessionId); 1330 } 1331 } 1332 } 1333 if (!sessionEffects.isEmpty()) { 1334 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1335 } 1336} 1337 1338void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1339 bool enabled, 1340 int sessionId) 1341{ 1342 Mutex::Autolock _l(mLock); 1343 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1344} 1345 1346void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1347 bool enabled, 1348 int sessionId) 1349{ 1350 if (mType != RECORD) { 1351 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1352 // another session. This gives the priority to well behaved effect control panels 1353 // and applications not using global effects. 1354 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1355 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1356 } 1357 } 1358 1359 sp<EffectChain> chain = getEffectChain_l(sessionId); 1360 if (chain != 0) { 1361 chain->checkSuspendOnEffectEnabled(effect, enabled); 1362 } 1363} 1364 1365// ---------------------------------------------------------------------------- 1366 1367AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1368 AudioStreamOut* output, 1369 int id, 1370 uint32_t device) 1371 : ThreadBase(audioFlinger, id, device), 1372 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 1373 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1374{ 1375 snprintf(mName, kNameLength, "AudioOut_%d", id); 1376 1377 readOutputParameters(); 1378 1379 mMasterVolume = mAudioFlinger->masterVolume(); 1380 mMasterMute = mAudioFlinger->masterMute(); 1381 1382 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 1383 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1384 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1385 mStreamTypes[stream].valid = true; 1386 } 1387} 1388 1389AudioFlinger::PlaybackThread::~PlaybackThread() 1390{ 1391 delete [] mMixBuffer; 1392} 1393 1394status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1395{ 1396 dumpInternals(fd, args); 1397 dumpTracks(fd, args); 1398 dumpEffectChains(fd, args); 1399 return NO_ERROR; 1400} 1401 1402status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1403{ 1404 const size_t SIZE = 256; 1405 char buffer[SIZE]; 1406 String8 result; 1407 1408 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1409 result.append(buffer); 1410 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1411 for (size_t i = 0; i < mTracks.size(); ++i) { 1412 sp<Track> track = mTracks[i]; 1413 if (track != 0) { 1414 track->dump(buffer, SIZE); 1415 result.append(buffer); 1416 } 1417 } 1418 1419 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1420 result.append(buffer); 1421 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1422 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1423 wp<Track> wTrack = mActiveTracks[i]; 1424 if (wTrack != 0) { 1425 sp<Track> track = wTrack.promote(); 1426 if (track != 0) { 1427 track->dump(buffer, SIZE); 1428 result.append(buffer); 1429 } 1430 } 1431 } 1432 write(fd, result.string(), result.size()); 1433 return NO_ERROR; 1434} 1435 1436status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1437{ 1438 const size_t SIZE = 256; 1439 char buffer[SIZE]; 1440 String8 result; 1441 1442 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1443 result.append(buffer); 1444 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1445 result.append(buffer); 1446 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1447 result.append(buffer); 1448 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1449 result.append(buffer); 1450 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1451 result.append(buffer); 1452 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1453 result.append(buffer); 1454 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1455 result.append(buffer); 1456 write(fd, result.string(), result.size()); 1457 1458 dumpBase(fd, args); 1459 1460 return NO_ERROR; 1461} 1462 1463// Thread virtuals 1464status_t AudioFlinger::PlaybackThread::readyToRun() 1465{ 1466 status_t status = initCheck(); 1467 if (status == NO_ERROR) { 1468 LOGI("AudioFlinger's thread %p ready to run", this); 1469 } else { 1470 LOGE("No working audio driver found."); 1471 } 1472 return status; 1473} 1474 1475void AudioFlinger::PlaybackThread::onFirstRef() 1476{ 1477 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1478} 1479 1480// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1481sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1482 const sp<AudioFlinger::Client>& client, 1483 int streamType, 1484 uint32_t sampleRate, 1485 uint32_t format, 1486 uint32_t channelMask, 1487 int frameCount, 1488 const sp<IMemory>& sharedBuffer, 1489 int sessionId, 1490 status_t *status) 1491{ 1492 sp<Track> track; 1493 status_t lStatus; 1494 1495 if (mType == DIRECT) { 1496 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1497 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1498 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1499 "for output %p with format %d", 1500 sampleRate, format, channelMask, mOutput, mFormat); 1501 lStatus = BAD_VALUE; 1502 goto Exit; 1503 } 1504 } 1505 } else { 1506 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1507 if (sampleRate > mSampleRate*2) { 1508 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1509 lStatus = BAD_VALUE; 1510 goto Exit; 1511 } 1512 } 1513 1514 lStatus = initCheck(); 1515 if (lStatus != NO_ERROR) { 1516 LOGE("Audio driver not initialized."); 1517 goto Exit; 1518 } 1519 1520 { // scope for mLock 1521 Mutex::Autolock _l(mLock); 1522 1523 // all tracks in same audio session must share the same routing strategy otherwise 1524 // conflicts will happen when tracks are moved from one output to another by audio policy 1525 // manager 1526 uint32_t strategy = 1527 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1528 for (size_t i = 0; i < mTracks.size(); ++i) { 1529 sp<Track> t = mTracks[i]; 1530 if (t != 0) { 1531 if (sessionId == t->sessionId() && 1532 strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) { 1533 lStatus = BAD_VALUE; 1534 goto Exit; 1535 } 1536 } 1537 } 1538 1539 track = new Track(this, client, streamType, sampleRate, format, 1540 channelMask, frameCount, sharedBuffer, sessionId); 1541 if (track->getCblk() == NULL || track->name() < 0) { 1542 lStatus = NO_MEMORY; 1543 goto Exit; 1544 } 1545 mTracks.add(track); 1546 1547 sp<EffectChain> chain = getEffectChain_l(sessionId); 1548 if (chain != 0) { 1549 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1550 track->setMainBuffer(chain->inBuffer()); 1551 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1552 chain->incTrackCnt(); 1553 } 1554 1555 // invalidate track immediately if the stream type was moved to another thread since 1556 // createTrack() was called by the client process. 1557 if (!mStreamTypes[streamType].valid) { 1558 LOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1559 this, streamType); 1560 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1561 } 1562 } 1563 lStatus = NO_ERROR; 1564 1565Exit: 1566 if(status) { 1567 *status = lStatus; 1568 } 1569 return track; 1570} 1571 1572uint32_t AudioFlinger::PlaybackThread::latency() const 1573{ 1574 Mutex::Autolock _l(mLock); 1575 if (initCheck() == NO_ERROR) { 1576 return mOutput->stream->get_latency(mOutput->stream); 1577 } else { 1578 return 0; 1579 } 1580} 1581 1582status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1583{ 1584 mMasterVolume = value; 1585 return NO_ERROR; 1586} 1587 1588status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1589{ 1590 mMasterMute = muted; 1591 return NO_ERROR; 1592} 1593 1594float AudioFlinger::PlaybackThread::masterVolume() const 1595{ 1596 return mMasterVolume; 1597} 1598 1599bool AudioFlinger::PlaybackThread::masterMute() const 1600{ 1601 return mMasterMute; 1602} 1603 1604status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1605{ 1606 mStreamTypes[stream].volume = value; 1607 return NO_ERROR; 1608} 1609 1610status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1611{ 1612 mStreamTypes[stream].mute = muted; 1613 return NO_ERROR; 1614} 1615 1616float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1617{ 1618 return mStreamTypes[stream].volume; 1619} 1620 1621bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1622{ 1623 return mStreamTypes[stream].mute; 1624} 1625 1626// addTrack_l() must be called with ThreadBase::mLock held 1627status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1628{ 1629 status_t status = ALREADY_EXISTS; 1630 1631 // set retry count for buffer fill 1632 track->mRetryCount = kMaxTrackStartupRetries; 1633 if (mActiveTracks.indexOf(track) < 0) { 1634 // the track is newly added, make sure it fills up all its 1635 // buffers before playing. This is to ensure the client will 1636 // effectively get the latency it requested. 1637 track->mFillingUpStatus = Track::FS_FILLING; 1638 track->mResetDone = false; 1639 mActiveTracks.add(track); 1640 if (track->mainBuffer() != mMixBuffer) { 1641 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1642 if (chain != 0) { 1643 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1644 chain->incActiveTrackCnt(); 1645 } 1646 } 1647 1648 status = NO_ERROR; 1649 } 1650 1651 ALOGV("mWaitWorkCV.broadcast"); 1652 mWaitWorkCV.broadcast(); 1653 1654 return status; 1655} 1656 1657// destroyTrack_l() must be called with ThreadBase::mLock held 1658void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1659{ 1660 track->mState = TrackBase::TERMINATED; 1661 if (mActiveTracks.indexOf(track) < 0) { 1662 removeTrack_l(track); 1663 } 1664} 1665 1666void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1667{ 1668 mTracks.remove(track); 1669 deleteTrackName_l(track->name()); 1670 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1671 if (chain != 0) { 1672 chain->decTrackCnt(); 1673 } 1674} 1675 1676String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1677{ 1678 String8 out_s8 = String8(""); 1679 char *s; 1680 1681 Mutex::Autolock _l(mLock); 1682 if (initCheck() != NO_ERROR) { 1683 return out_s8; 1684 } 1685 1686 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1687 out_s8 = String8(s); 1688 free(s); 1689 return out_s8; 1690} 1691 1692// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1693void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1694 AudioSystem::OutputDescriptor desc; 1695 void *param2 = 0; 1696 1697 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1698 1699 switch (event) { 1700 case AudioSystem::OUTPUT_OPENED: 1701 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1702 desc.channels = mChannelMask; 1703 desc.samplingRate = mSampleRate; 1704 desc.format = mFormat; 1705 desc.frameCount = mFrameCount; 1706 desc.latency = latency(); 1707 param2 = &desc; 1708 break; 1709 1710 case AudioSystem::STREAM_CONFIG_CHANGED: 1711 param2 = ¶m; 1712 case AudioSystem::OUTPUT_CLOSED: 1713 default: 1714 break; 1715 } 1716 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1717} 1718 1719void AudioFlinger::PlaybackThread::readOutputParameters() 1720{ 1721 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1722 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1723 mChannelCount = (uint16_t)popcount(mChannelMask); 1724 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1725 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); 1726 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1727 1728 // FIXME - Current mixer implementation only supports stereo output: Always 1729 // Allocate a stereo buffer even if HW output is mono. 1730 if (mMixBuffer != NULL) delete[] mMixBuffer; 1731 mMixBuffer = new int16_t[mFrameCount * 2]; 1732 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1733 1734 // force reconfiguration of effect chains and engines to take new buffer size and audio 1735 // parameters into account 1736 // Note that mLock is not held when readOutputParameters() is called from the constructor 1737 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1738 // matter. 1739 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1740 Vector< sp<EffectChain> > effectChains = mEffectChains; 1741 for (size_t i = 0; i < effectChains.size(); i ++) { 1742 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1743 } 1744} 1745 1746status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1747{ 1748 if (halFrames == 0 || dspFrames == 0) { 1749 return BAD_VALUE; 1750 } 1751 Mutex::Autolock _l(mLock); 1752 if (initCheck() != NO_ERROR) { 1753 return INVALID_OPERATION; 1754 } 1755 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1756 1757 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1758} 1759 1760uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1761{ 1762 Mutex::Autolock _l(mLock); 1763 uint32_t result = 0; 1764 if (getEffectChain_l(sessionId) != 0) { 1765 result = EFFECT_SESSION; 1766 } 1767 1768 for (size_t i = 0; i < mTracks.size(); ++i) { 1769 sp<Track> track = mTracks[i]; 1770 if (sessionId == track->sessionId() && 1771 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1772 result |= TRACK_SESSION; 1773 break; 1774 } 1775 } 1776 1777 return result; 1778} 1779 1780uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1781{ 1782 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1783 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1784 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1785 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1786 } 1787 for (size_t i = 0; i < mTracks.size(); i++) { 1788 sp<Track> track = mTracks[i]; 1789 if (sessionId == track->sessionId() && 1790 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1791 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1792 } 1793 } 1794 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1795} 1796 1797 1798AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() 1799{ 1800 Mutex::Autolock _l(mLock); 1801 return mOutput; 1802} 1803 1804AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1805{ 1806 Mutex::Autolock _l(mLock); 1807 AudioStreamOut *output = mOutput; 1808 mOutput = NULL; 1809 return output; 1810} 1811 1812// this method must always be called either with ThreadBase mLock held or inside the thread loop 1813audio_stream_t* AudioFlinger::PlaybackThread::stream() 1814{ 1815 if (mOutput == NULL) { 1816 return NULL; 1817 } 1818 return &mOutput->stream->common; 1819} 1820 1821uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1822{ 1823 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1824 // decoding and transfer time. So sleeping for half of the latency would likely cause 1825 // underruns 1826 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1827 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1828 } else { 1829 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1830 } 1831} 1832 1833// ---------------------------------------------------------------------------- 1834 1835AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1836 : PlaybackThread(audioFlinger, output, id, device), 1837 mAudioMixer(0) 1838{ 1839 mType = ThreadBase::MIXER; 1840 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1841 1842 // FIXME - Current mixer implementation only supports stereo output 1843 if (mChannelCount == 1) { 1844 LOGE("Invalid audio hardware channel count"); 1845 } 1846} 1847 1848AudioFlinger::MixerThread::~MixerThread() 1849{ 1850 delete mAudioMixer; 1851} 1852 1853bool AudioFlinger::MixerThread::threadLoop() 1854{ 1855 Vector< sp<Track> > tracksToRemove; 1856 uint32_t mixerStatus = MIXER_IDLE; 1857 nsecs_t standbyTime = systemTime(); 1858 size_t mixBufferSize = mFrameCount * mFrameSize; 1859 // FIXME: Relaxed timing because of a certain device that can't meet latency 1860 // Should be reduced to 2x after the vendor fixes the driver issue 1861 // increase threshold again due to low power audio mode. The way this warning threshold is 1862 // calculated and its usefulness should be reconsidered anyway. 1863 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1864 nsecs_t lastWarning = 0; 1865 bool longStandbyExit = false; 1866 uint32_t activeSleepTime = activeSleepTimeUs(); 1867 uint32_t idleSleepTime = idleSleepTimeUs(); 1868 uint32_t sleepTime = idleSleepTime; 1869 uint32_t sleepTimeShift = 0; 1870 Vector< sp<EffectChain> > effectChains; 1871#ifdef DEBUG_CPU_USAGE 1872 ThreadCpuUsage cpu; 1873 const CentralTendencyStatistics& stats = cpu.statistics(); 1874#endif 1875 1876 acquireWakeLock(); 1877 1878 while (!exitPending()) 1879 { 1880#ifdef DEBUG_CPU_USAGE 1881 cpu.sampleAndEnable(); 1882 unsigned n = stats.n(); 1883 // cpu.elapsed() is expensive, so don't call it every loop 1884 if ((n & 127) == 1) { 1885 long long elapsed = cpu.elapsed(); 1886 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1887 double perLoop = elapsed / (double) n; 1888 double perLoop100 = perLoop * 0.01; 1889 double mean = stats.mean(); 1890 double stddev = stats.stddev(); 1891 double minimum = stats.minimum(); 1892 double maximum = stats.maximum(); 1893 cpu.resetStatistics(); 1894 LOGI("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", 1895 elapsed * .000000001, n, perLoop * .000001, 1896 mean * .001, 1897 stddev * .001, 1898 minimum * .001, 1899 maximum * .001, 1900 mean / perLoop100, 1901 stddev / perLoop100, 1902 minimum / perLoop100, 1903 maximum / perLoop100); 1904 } 1905 } 1906#endif 1907 processConfigEvents(); 1908 1909 mixerStatus = MIXER_IDLE; 1910 { // scope for mLock 1911 1912 Mutex::Autolock _l(mLock); 1913 1914 if (checkForNewParameters_l()) { 1915 mixBufferSize = mFrameCount * mFrameSize; 1916 // FIXME: Relaxed timing because of a certain device that can't meet latency 1917 // Should be reduced to 2x after the vendor fixes the driver issue 1918 // increase threshold again due to low power audio mode. The way this warning 1919 // threshold is calculated and its usefulness should be reconsidered anyway. 1920 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1921 activeSleepTime = activeSleepTimeUs(); 1922 idleSleepTime = idleSleepTimeUs(); 1923 } 1924 1925 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1926 1927 // put audio hardware into standby after short delay 1928 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1929 mSuspended) { 1930 if (!mStandby) { 1931 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1932 mOutput->stream->common.standby(&mOutput->stream->common); 1933 mStandby = true; 1934 mBytesWritten = 0; 1935 } 1936 1937 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1938 // we're about to wait, flush the binder command buffer 1939 IPCThreadState::self()->flushCommands(); 1940 1941 if (exitPending()) break; 1942 1943 releaseWakeLock_l(); 1944 // wait until we have something to do... 1945 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1946 mWaitWorkCV.wait(mLock); 1947 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1948 acquireWakeLock_l(); 1949 1950 if (mMasterMute == false) { 1951 char value[PROPERTY_VALUE_MAX]; 1952 property_get("ro.audio.silent", value, "0"); 1953 if (atoi(value)) { 1954 LOGD("Silence is golden"); 1955 setMasterMute(true); 1956 } 1957 } 1958 1959 standbyTime = systemTime() + kStandbyTimeInNsecs; 1960 sleepTime = idleSleepTime; 1961 sleepTimeShift = 0; 1962 continue; 1963 } 1964 } 1965 1966 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1967 1968 // prevent any changes in effect chain list and in each effect chain 1969 // during mixing and effect process as the audio buffers could be deleted 1970 // or modified if an effect is created or deleted 1971 lockEffectChains_l(effectChains); 1972 } 1973 1974 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1975 // mix buffers... 1976 mAudioMixer->process(); 1977 sleepTime = 0; 1978 // increase sleep time progressively when application underrun condition clears 1979 if (sleepTimeShift > 0) { 1980 sleepTimeShift--; 1981 } 1982 standbyTime = systemTime() + kStandbyTimeInNsecs; 1983 //TODO: delay standby when effects have a tail 1984 } else { 1985 // If no tracks are ready, sleep once for the duration of an output 1986 // buffer size, then write 0s to the output 1987 if (sleepTime == 0) { 1988 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1989 sleepTime = activeSleepTime >> sleepTimeShift; 1990 if (sleepTime < kMinThreadSleepTimeUs) { 1991 sleepTime = kMinThreadSleepTimeUs; 1992 } 1993 // reduce sleep time in case of consecutive application underruns to avoid 1994 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 1995 // duration we would end up writing less data than needed by the audio HAL if 1996 // the condition persists. 1997 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 1998 sleepTimeShift++; 1999 } 2000 } else { 2001 sleepTime = idleSleepTime; 2002 } 2003 } else if (mBytesWritten != 0 || 2004 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2005 memset (mMixBuffer, 0, mixBufferSize); 2006 sleepTime = 0; 2007 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2008 } 2009 // TODO add standby time extension fct of effect tail 2010 } 2011 2012 if (mSuspended) { 2013 sleepTime = suspendSleepTimeUs(); 2014 } 2015 // sleepTime == 0 means we must write to audio hardware 2016 if (sleepTime == 0) { 2017 for (size_t i = 0; i < effectChains.size(); i ++) { 2018 effectChains[i]->process_l(); 2019 } 2020 // enable changes in effect chain 2021 unlockEffectChains(effectChains); 2022 mLastWriteTime = systemTime(); 2023 mInWrite = true; 2024 mBytesWritten += mixBufferSize; 2025 2026 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2027 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2028 mNumWrites++; 2029 mInWrite = false; 2030 nsecs_t now = systemTime(); 2031 nsecs_t delta = now - mLastWriteTime; 2032 if (!mStandby && delta > maxPeriod) { 2033 mNumDelayedWrites++; 2034 if ((now - lastWarning) > kWarningThrottle) { 2035 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2036 ns2ms(delta), mNumDelayedWrites, this); 2037 lastWarning = now; 2038 } 2039 if (mStandby) { 2040 longStandbyExit = true; 2041 } 2042 } 2043 mStandby = false; 2044 } else { 2045 // enable changes in effect chain 2046 unlockEffectChains(effectChains); 2047 usleep(sleepTime); 2048 } 2049 2050 // finally let go of all our tracks, without the lock held 2051 // since we can't guarantee the destructors won't acquire that 2052 // same lock. 2053 tracksToRemove.clear(); 2054 2055 // Effect chains will be actually deleted here if they were removed from 2056 // mEffectChains list during mixing or effects processing 2057 effectChains.clear(); 2058 } 2059 2060 if (!mStandby) { 2061 mOutput->stream->common.standby(&mOutput->stream->common); 2062 } 2063 2064 releaseWakeLock(); 2065 2066 ALOGV("MixerThread %p exiting", this); 2067 return false; 2068} 2069 2070// prepareTracks_l() must be called with ThreadBase::mLock held 2071uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2072{ 2073 2074 uint32_t mixerStatus = MIXER_IDLE; 2075 // find out which tracks need to be processed 2076 size_t count = activeTracks.size(); 2077 size_t mixedTracks = 0; 2078 size_t tracksWithEffect = 0; 2079 2080 float masterVolume = mMasterVolume; 2081 bool masterMute = mMasterMute; 2082 2083 if (masterMute) { 2084 masterVolume = 0; 2085 } 2086 // Delegate master volume control to effect in output mix effect chain if needed 2087 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2088 if (chain != 0) { 2089 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2090 chain->setVolume_l(&v, &v); 2091 masterVolume = (float)((v + (1 << 23)) >> 24); 2092 chain.clear(); 2093 } 2094 2095 for (size_t i=0 ; i<count ; i++) { 2096 sp<Track> t = activeTracks[i].promote(); 2097 if (t == 0) continue; 2098 2099 Track* const track = t.get(); 2100 audio_track_cblk_t* cblk = track->cblk(); 2101 2102 // The first time a track is added we wait 2103 // for all its buffers to be filled before processing it 2104 mAudioMixer->setActiveTrack(track->name()); 2105 // make sure that we have enough frames to mix one full buffer. 2106 // enforce this condition only once to enable draining the buffer in case the client 2107 // app does not call stop() and relies on underrun to stop: 2108 // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed 2109 // during last round 2110 uint32_t minFrames = 1; 2111 if (!track->isStopped() && !track->isPausing() && 2112 (track->mRetryCount >= kMaxTrackRetries)) { 2113 if (t->sampleRate() == (int)mSampleRate) { 2114 minFrames = mFrameCount; 2115 } else { 2116 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1; 2117 } 2118 } 2119 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2120 !track->isPaused() && !track->isTerminated()) 2121 { 2122 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 2123 2124 mixedTracks++; 2125 2126 // track->mainBuffer() != mMixBuffer means there is an effect chain 2127 // connected to the track 2128 chain.clear(); 2129 if (track->mainBuffer() != mMixBuffer) { 2130 chain = getEffectChain_l(track->sessionId()); 2131 // Delegate volume control to effect in track effect chain if needed 2132 if (chain != 0) { 2133 tracksWithEffect++; 2134 } else { 2135 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 2136 track->name(), track->sessionId()); 2137 } 2138 } 2139 2140 2141 int param = AudioMixer::VOLUME; 2142 if (track->mFillingUpStatus == Track::FS_FILLED) { 2143 // no ramp for the first volume setting 2144 track->mFillingUpStatus = Track::FS_ACTIVE; 2145 if (track->mState == TrackBase::RESUMING) { 2146 track->mState = TrackBase::ACTIVE; 2147 param = AudioMixer::RAMP_VOLUME; 2148 } 2149 mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2150 } else if (cblk->server != 0) { 2151 // If the track is stopped before the first frame was mixed, 2152 // do not apply ramp 2153 param = AudioMixer::RAMP_VOLUME; 2154 } 2155 2156 // compute volume for this track 2157 uint32_t vl, vr, va; 2158 if (track->isMuted() || track->isPausing() || 2159 mStreamTypes[track->type()].mute) { 2160 vl = vr = va = 0; 2161 if (track->isPausing()) { 2162 track->setPaused(); 2163 } 2164 } else { 2165 2166 // read original volumes with volume control 2167 float typeVolume = mStreamTypes[track->type()].volume; 2168 float v = masterVolume * typeVolume; 2169 vl = (uint32_t)(v * cblk->volume[0]) << 12; 2170 vr = (uint32_t)(v * cblk->volume[1]) << 12; 2171 2172 va = (uint32_t)(v * cblk->sendLevel); 2173 } 2174 // Delegate volume control to effect in track effect chain if needed 2175 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2176 // Do not ramp volume if volume is controlled by effect 2177 param = AudioMixer::VOLUME; 2178 track->mHasVolumeController = true; 2179 } else { 2180 // force no volume ramp when volume controller was just disabled or removed 2181 // from effect chain to avoid volume spike 2182 if (track->mHasVolumeController) { 2183 param = AudioMixer::VOLUME; 2184 } 2185 track->mHasVolumeController = false; 2186 } 2187 2188 // Convert volumes from 8.24 to 4.12 format 2189 int16_t left, right, aux; 2190 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2191 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2192 left = int16_t(v_clamped); 2193 v_clamped = (vr + (1 << 11)) >> 12; 2194 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2195 right = int16_t(v_clamped); 2196 2197 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2198 aux = int16_t(va); 2199 2200 // XXX: these things DON'T need to be done each time 2201 mAudioMixer->setBufferProvider(track); 2202 mAudioMixer->enable(AudioMixer::MIXING); 2203 2204 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 2205 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 2206 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 2207 mAudioMixer->setParameter( 2208 AudioMixer::TRACK, 2209 AudioMixer::FORMAT, (void *)track->format()); 2210 mAudioMixer->setParameter( 2211 AudioMixer::TRACK, 2212 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2213 mAudioMixer->setParameter( 2214 AudioMixer::RESAMPLE, 2215 AudioMixer::SAMPLE_RATE, 2216 (void *)(cblk->sampleRate)); 2217 mAudioMixer->setParameter( 2218 AudioMixer::TRACK, 2219 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2220 mAudioMixer->setParameter( 2221 AudioMixer::TRACK, 2222 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2223 2224 // reset retry count 2225 track->mRetryCount = kMaxTrackRetries; 2226 mixerStatus = MIXER_TRACKS_READY; 2227 } else { 2228 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 2229 if (track->isStopped()) { 2230 track->reset(); 2231 } 2232 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2233 // We have consumed all the buffers of this track. 2234 // Remove it from the list of active tracks. 2235 tracksToRemove->add(track); 2236 } else { 2237 // No buffers for this track. Give it a few chances to 2238 // fill a buffer, then remove it from active list. 2239 if (--(track->mRetryCount) <= 0) { 2240 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 2241 tracksToRemove->add(track); 2242 // indicate to client process that the track was disabled because of underrun 2243 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2244 } else if (mixerStatus != MIXER_TRACKS_READY) { 2245 mixerStatus = MIXER_TRACKS_ENABLED; 2246 } 2247 } 2248 mAudioMixer->disable(AudioMixer::MIXING); 2249 } 2250 } 2251 2252 // remove all the tracks that need to be... 2253 count = tracksToRemove->size(); 2254 if (UNLIKELY(count)) { 2255 for (size_t i=0 ; i<count ; i++) { 2256 const sp<Track>& track = tracksToRemove->itemAt(i); 2257 mActiveTracks.remove(track); 2258 if (track->mainBuffer() != mMixBuffer) { 2259 chain = getEffectChain_l(track->sessionId()); 2260 if (chain != 0) { 2261 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2262 chain->decActiveTrackCnt(); 2263 } 2264 } 2265 if (track->isTerminated()) { 2266 removeTrack_l(track); 2267 } 2268 } 2269 } 2270 2271 // mix buffer must be cleared if all tracks are connected to an 2272 // effect chain as in this case the mixer will not write to 2273 // mix buffer and track effects will accumulate into it 2274 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2275 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2276 } 2277 2278 return mixerStatus; 2279} 2280 2281void AudioFlinger::MixerThread::invalidateTracks(int streamType) 2282{ 2283 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2284 this, streamType, mTracks.size()); 2285 Mutex::Autolock _l(mLock); 2286 2287 size_t size = mTracks.size(); 2288 for (size_t i = 0; i < size; i++) { 2289 sp<Track> t = mTracks[i]; 2290 if (t->type() == streamType) { 2291 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2292 t->mCblk->cv.signal(); 2293 } 2294 } 2295} 2296 2297void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid) 2298{ 2299 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2300 this, streamType, valid); 2301 Mutex::Autolock _l(mLock); 2302 2303 mStreamTypes[streamType].valid = valid; 2304} 2305 2306// getTrackName_l() must be called with ThreadBase::mLock held 2307int AudioFlinger::MixerThread::getTrackName_l() 2308{ 2309 return mAudioMixer->getTrackName(); 2310} 2311 2312// deleteTrackName_l() must be called with ThreadBase::mLock held 2313void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2314{ 2315 ALOGV("remove track (%d) and delete from mixer", name); 2316 mAudioMixer->deleteTrackName(name); 2317} 2318 2319// checkForNewParameters_l() must be called with ThreadBase::mLock held 2320bool AudioFlinger::MixerThread::checkForNewParameters_l() 2321{ 2322 bool reconfig = false; 2323 2324 while (!mNewParameters.isEmpty()) { 2325 status_t status = NO_ERROR; 2326 String8 keyValuePair = mNewParameters[0]; 2327 AudioParameter param = AudioParameter(keyValuePair); 2328 int value; 2329 2330 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2331 reconfig = true; 2332 } 2333 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2334 if (value != AUDIO_FORMAT_PCM_16_BIT) { 2335 status = BAD_VALUE; 2336 } else { 2337 reconfig = true; 2338 } 2339 } 2340 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2341 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2342 status = BAD_VALUE; 2343 } else { 2344 reconfig = true; 2345 } 2346 } 2347 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2348 // do not accept frame count changes if tracks are open as the track buffer 2349 // size depends on frame count and correct behavior would not be garantied 2350 // if frame count is changed after track creation 2351 if (!mTracks.isEmpty()) { 2352 status = INVALID_OPERATION; 2353 } else { 2354 reconfig = true; 2355 } 2356 } 2357 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2358 // when changing the audio output device, call addBatteryData to notify 2359 // the change 2360 if ((int)mDevice != value) { 2361 uint32_t params = 0; 2362 // check whether speaker is on 2363 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2364 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2365 } 2366 2367 int deviceWithoutSpeaker 2368 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2369 // check if any other device (except speaker) is on 2370 if (value & deviceWithoutSpeaker ) { 2371 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2372 } 2373 2374 if (params != 0) { 2375 addBatteryData(params); 2376 } 2377 } 2378 2379 // forward device change to effects that have requested to be 2380 // aware of attached audio device. 2381 mDevice = (uint32_t)value; 2382 for (size_t i = 0; i < mEffectChains.size(); i++) { 2383 mEffectChains[i]->setDevice_l(mDevice); 2384 } 2385 } 2386 2387 if (status == NO_ERROR) { 2388 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2389 keyValuePair.string()); 2390 if (!mStandby && status == INVALID_OPERATION) { 2391 mOutput->stream->common.standby(&mOutput->stream->common); 2392 mStandby = true; 2393 mBytesWritten = 0; 2394 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2395 keyValuePair.string()); 2396 } 2397 if (status == NO_ERROR && reconfig) { 2398 delete mAudioMixer; 2399 readOutputParameters(); 2400 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2401 for (size_t i = 0; i < mTracks.size() ; i++) { 2402 int name = getTrackName_l(); 2403 if (name < 0) break; 2404 mTracks[i]->mName = name; 2405 // limit track sample rate to 2 x new output sample rate 2406 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2407 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2408 } 2409 } 2410 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2411 } 2412 } 2413 2414 mNewParameters.removeAt(0); 2415 2416 mParamStatus = status; 2417 mParamCond.signal(); 2418 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2419 // already timed out waiting for the status and will never signal the condition. 2420 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout); 2421 } 2422 return reconfig; 2423} 2424 2425status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2426{ 2427 const size_t SIZE = 256; 2428 char buffer[SIZE]; 2429 String8 result; 2430 2431 PlaybackThread::dumpInternals(fd, args); 2432 2433 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2434 result.append(buffer); 2435 write(fd, result.string(), result.size()); 2436 return NO_ERROR; 2437} 2438 2439uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2440{ 2441 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2442} 2443 2444uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2445{ 2446 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2447} 2448 2449// ---------------------------------------------------------------------------- 2450AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2451 : PlaybackThread(audioFlinger, output, id, device) 2452{ 2453 mType = ThreadBase::DIRECT; 2454} 2455 2456AudioFlinger::DirectOutputThread::~DirectOutputThread() 2457{ 2458} 2459 2460static inline 2461int32_t mul(int16_t in, int16_t v) 2462{ 2463#if defined(__arm__) && !defined(__thumb__) 2464 int32_t out; 2465 asm( "smulbb %[out], %[in], %[v] \n" 2466 : [out]"=r"(out) 2467 : [in]"%r"(in), [v]"r"(v) 2468 : ); 2469 return out; 2470#else 2471 return in * int32_t(v); 2472#endif 2473} 2474 2475void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2476{ 2477 // Do not apply volume on compressed audio 2478 if (!audio_is_linear_pcm(mFormat)) { 2479 return; 2480 } 2481 2482 // convert to signed 16 bit before volume calculation 2483 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2484 size_t count = mFrameCount * mChannelCount; 2485 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2486 int16_t *dst = mMixBuffer + count-1; 2487 while(count--) { 2488 *dst-- = (int16_t)(*src--^0x80) << 8; 2489 } 2490 } 2491 2492 size_t frameCount = mFrameCount; 2493 int16_t *out = mMixBuffer; 2494 if (ramp) { 2495 if (mChannelCount == 1) { 2496 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2497 int32_t vlInc = d / (int32_t)frameCount; 2498 int32_t vl = ((int32_t)mLeftVolShort << 16); 2499 do { 2500 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2501 out++; 2502 vl += vlInc; 2503 } while (--frameCount); 2504 2505 } else { 2506 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2507 int32_t vlInc = d / (int32_t)frameCount; 2508 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2509 int32_t vrInc = d / (int32_t)frameCount; 2510 int32_t vl = ((int32_t)mLeftVolShort << 16); 2511 int32_t vr = ((int32_t)mRightVolShort << 16); 2512 do { 2513 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2514 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2515 out += 2; 2516 vl += vlInc; 2517 vr += vrInc; 2518 } while (--frameCount); 2519 } 2520 } else { 2521 if (mChannelCount == 1) { 2522 do { 2523 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2524 out++; 2525 } while (--frameCount); 2526 } else { 2527 do { 2528 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2529 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2530 out += 2; 2531 } while (--frameCount); 2532 } 2533 } 2534 2535 // convert back to unsigned 8 bit after volume calculation 2536 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2537 size_t count = mFrameCount * mChannelCount; 2538 int16_t *src = mMixBuffer; 2539 uint8_t *dst = (uint8_t *)mMixBuffer; 2540 while(count--) { 2541 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2542 } 2543 } 2544 2545 mLeftVolShort = leftVol; 2546 mRightVolShort = rightVol; 2547} 2548 2549bool AudioFlinger::DirectOutputThread::threadLoop() 2550{ 2551 uint32_t mixerStatus = MIXER_IDLE; 2552 sp<Track> trackToRemove; 2553 sp<Track> activeTrack; 2554 nsecs_t standbyTime = systemTime(); 2555 int8_t *curBuf; 2556 size_t mixBufferSize = mFrameCount*mFrameSize; 2557 uint32_t activeSleepTime = activeSleepTimeUs(); 2558 uint32_t idleSleepTime = idleSleepTimeUs(); 2559 uint32_t sleepTime = idleSleepTime; 2560 // use shorter standby delay as on normal output to release 2561 // hardware resources as soon as possible 2562 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2563 2564 acquireWakeLock(); 2565 2566 while (!exitPending()) 2567 { 2568 bool rampVolume; 2569 uint16_t leftVol; 2570 uint16_t rightVol; 2571 Vector< sp<EffectChain> > effectChains; 2572 2573 processConfigEvents(); 2574 2575 mixerStatus = MIXER_IDLE; 2576 2577 { // scope for the mLock 2578 2579 Mutex::Autolock _l(mLock); 2580 2581 if (checkForNewParameters_l()) { 2582 mixBufferSize = mFrameCount*mFrameSize; 2583 activeSleepTime = activeSleepTimeUs(); 2584 idleSleepTime = idleSleepTimeUs(); 2585 standbyDelay = microseconds(activeSleepTime*2); 2586 } 2587 2588 // put audio hardware into standby after short delay 2589 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2590 mSuspended) { 2591 // wait until we have something to do... 2592 if (!mStandby) { 2593 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2594 mOutput->stream->common.standby(&mOutput->stream->common); 2595 mStandby = true; 2596 mBytesWritten = 0; 2597 } 2598 2599 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2600 // we're about to wait, flush the binder command buffer 2601 IPCThreadState::self()->flushCommands(); 2602 2603 if (exitPending()) break; 2604 2605 releaseWakeLock_l(); 2606 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2607 mWaitWorkCV.wait(mLock); 2608 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2609 acquireWakeLock_l(); 2610 2611 if (mMasterMute == false) { 2612 char value[PROPERTY_VALUE_MAX]; 2613 property_get("ro.audio.silent", value, "0"); 2614 if (atoi(value)) { 2615 LOGD("Silence is golden"); 2616 setMasterMute(true); 2617 } 2618 } 2619 2620 standbyTime = systemTime() + standbyDelay; 2621 sleepTime = idleSleepTime; 2622 continue; 2623 } 2624 } 2625 2626 effectChains = mEffectChains; 2627 2628 // find out which tracks need to be processed 2629 if (mActiveTracks.size() != 0) { 2630 sp<Track> t = mActiveTracks[0].promote(); 2631 if (t == 0) continue; 2632 2633 Track* const track = t.get(); 2634 audio_track_cblk_t* cblk = track->cblk(); 2635 2636 // The first time a track is added we wait 2637 // for all its buffers to be filled before processing it 2638 if (cblk->framesReady() && track->isReady() && 2639 !track->isPaused() && !track->isTerminated()) 2640 { 2641 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2642 2643 if (track->mFillingUpStatus == Track::FS_FILLED) { 2644 track->mFillingUpStatus = Track::FS_ACTIVE; 2645 mLeftVolFloat = mRightVolFloat = 0; 2646 mLeftVolShort = mRightVolShort = 0; 2647 if (track->mState == TrackBase::RESUMING) { 2648 track->mState = TrackBase::ACTIVE; 2649 rampVolume = true; 2650 } 2651 } else if (cblk->server != 0) { 2652 // If the track is stopped before the first frame was mixed, 2653 // do not apply ramp 2654 rampVolume = true; 2655 } 2656 // compute volume for this track 2657 float left, right; 2658 if (track->isMuted() || mMasterMute || track->isPausing() || 2659 mStreamTypes[track->type()].mute) { 2660 left = right = 0; 2661 if (track->isPausing()) { 2662 track->setPaused(); 2663 } 2664 } else { 2665 float typeVolume = mStreamTypes[track->type()].volume; 2666 float v = mMasterVolume * typeVolume; 2667 float v_clamped = v * cblk->volume[0]; 2668 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2669 left = v_clamped/MAX_GAIN; 2670 v_clamped = v * cblk->volume[1]; 2671 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2672 right = v_clamped/MAX_GAIN; 2673 } 2674 2675 if (left != mLeftVolFloat || right != mRightVolFloat) { 2676 mLeftVolFloat = left; 2677 mRightVolFloat = right; 2678 2679 // If audio HAL implements volume control, 2680 // force software volume to nominal value 2681 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2682 left = 1.0f; 2683 right = 1.0f; 2684 } 2685 2686 // Convert volumes from float to 8.24 2687 uint32_t vl = (uint32_t)(left * (1 << 24)); 2688 uint32_t vr = (uint32_t)(right * (1 << 24)); 2689 2690 // Delegate volume control to effect in track effect chain if needed 2691 // only one effect chain can be present on DirectOutputThread, so if 2692 // there is one, the track is connected to it 2693 if (!effectChains.isEmpty()) { 2694 // Do not ramp volume if volume is controlled by effect 2695 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2696 rampVolume = false; 2697 } 2698 } 2699 2700 // Convert volumes from 8.24 to 4.12 format 2701 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2702 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2703 leftVol = (uint16_t)v_clamped; 2704 v_clamped = (vr + (1 << 11)) >> 12; 2705 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2706 rightVol = (uint16_t)v_clamped; 2707 } else { 2708 leftVol = mLeftVolShort; 2709 rightVol = mRightVolShort; 2710 rampVolume = false; 2711 } 2712 2713 // reset retry count 2714 track->mRetryCount = kMaxTrackRetriesDirect; 2715 activeTrack = t; 2716 mixerStatus = MIXER_TRACKS_READY; 2717 } else { 2718 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2719 if (track->isStopped()) { 2720 track->reset(); 2721 } 2722 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2723 // We have consumed all the buffers of this track. 2724 // Remove it from the list of active tracks. 2725 trackToRemove = track; 2726 } else { 2727 // No buffers for this track. Give it a few chances to 2728 // fill a buffer, then remove it from active list. 2729 if (--(track->mRetryCount) <= 0) { 2730 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2731 trackToRemove = track; 2732 } else { 2733 mixerStatus = MIXER_TRACKS_ENABLED; 2734 } 2735 } 2736 } 2737 } 2738 2739 // remove all the tracks that need to be... 2740 if (UNLIKELY(trackToRemove != 0)) { 2741 mActiveTracks.remove(trackToRemove); 2742 if (!effectChains.isEmpty()) { 2743 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2744 trackToRemove->sessionId()); 2745 effectChains[0]->decActiveTrackCnt(); 2746 } 2747 if (trackToRemove->isTerminated()) { 2748 removeTrack_l(trackToRemove); 2749 } 2750 } 2751 2752 lockEffectChains_l(effectChains); 2753 } 2754 2755 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2756 AudioBufferProvider::Buffer buffer; 2757 size_t frameCount = mFrameCount; 2758 curBuf = (int8_t *)mMixBuffer; 2759 // output audio to hardware 2760 while (frameCount) { 2761 buffer.frameCount = frameCount; 2762 activeTrack->getNextBuffer(&buffer); 2763 if (UNLIKELY(buffer.raw == 0)) { 2764 memset(curBuf, 0, frameCount * mFrameSize); 2765 break; 2766 } 2767 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2768 frameCount -= buffer.frameCount; 2769 curBuf += buffer.frameCount * mFrameSize; 2770 activeTrack->releaseBuffer(&buffer); 2771 } 2772 sleepTime = 0; 2773 standbyTime = systemTime() + standbyDelay; 2774 } else { 2775 if (sleepTime == 0) { 2776 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2777 sleepTime = activeSleepTime; 2778 } else { 2779 sleepTime = idleSleepTime; 2780 } 2781 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2782 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2783 sleepTime = 0; 2784 } 2785 } 2786 2787 if (mSuspended) { 2788 sleepTime = suspendSleepTimeUs(); 2789 } 2790 // sleepTime == 0 means we must write to audio hardware 2791 if (sleepTime == 0) { 2792 if (mixerStatus == MIXER_TRACKS_READY) { 2793 applyVolume(leftVol, rightVol, rampVolume); 2794 } 2795 for (size_t i = 0; i < effectChains.size(); i ++) { 2796 effectChains[i]->process_l(); 2797 } 2798 unlockEffectChains(effectChains); 2799 2800 mLastWriteTime = systemTime(); 2801 mInWrite = true; 2802 mBytesWritten += mixBufferSize; 2803 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2804 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2805 mNumWrites++; 2806 mInWrite = false; 2807 mStandby = false; 2808 } else { 2809 unlockEffectChains(effectChains); 2810 usleep(sleepTime); 2811 } 2812 2813 // finally let go of removed track, without the lock held 2814 // since we can't guarantee the destructors won't acquire that 2815 // same lock. 2816 trackToRemove.clear(); 2817 activeTrack.clear(); 2818 2819 // Effect chains will be actually deleted here if they were removed from 2820 // mEffectChains list during mixing or effects processing 2821 effectChains.clear(); 2822 } 2823 2824 if (!mStandby) { 2825 mOutput->stream->common.standby(&mOutput->stream->common); 2826 } 2827 2828 releaseWakeLock(); 2829 2830 ALOGV("DirectOutputThread %p exiting", this); 2831 return false; 2832} 2833 2834// getTrackName_l() must be called with ThreadBase::mLock held 2835int AudioFlinger::DirectOutputThread::getTrackName_l() 2836{ 2837 return 0; 2838} 2839 2840// deleteTrackName_l() must be called with ThreadBase::mLock held 2841void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2842{ 2843} 2844 2845// checkForNewParameters_l() must be called with ThreadBase::mLock held 2846bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2847{ 2848 bool reconfig = false; 2849 2850 while (!mNewParameters.isEmpty()) { 2851 status_t status = NO_ERROR; 2852 String8 keyValuePair = mNewParameters[0]; 2853 AudioParameter param = AudioParameter(keyValuePair); 2854 int value; 2855 2856 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2857 // do not accept frame count changes if tracks are open as the track buffer 2858 // size depends on frame count and correct behavior would not be garantied 2859 // if frame count is changed after track creation 2860 if (!mTracks.isEmpty()) { 2861 status = INVALID_OPERATION; 2862 } else { 2863 reconfig = true; 2864 } 2865 } 2866 if (status == NO_ERROR) { 2867 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2868 keyValuePair.string()); 2869 if (!mStandby && status == INVALID_OPERATION) { 2870 mOutput->stream->common.standby(&mOutput->stream->common); 2871 mStandby = true; 2872 mBytesWritten = 0; 2873 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2874 keyValuePair.string()); 2875 } 2876 if (status == NO_ERROR && reconfig) { 2877 readOutputParameters(); 2878 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2879 } 2880 } 2881 2882 mNewParameters.removeAt(0); 2883 2884 mParamStatus = status; 2885 mParamCond.signal(); 2886 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2887 // already timed out waiting for the status and will never signal the condition. 2888 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout); 2889 } 2890 return reconfig; 2891} 2892 2893uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2894{ 2895 uint32_t time; 2896 if (audio_is_linear_pcm(mFormat)) { 2897 time = PlaybackThread::activeSleepTimeUs(); 2898 } else { 2899 time = 10000; 2900 } 2901 return time; 2902} 2903 2904uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2905{ 2906 uint32_t time; 2907 if (audio_is_linear_pcm(mFormat)) { 2908 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2909 } else { 2910 time = 10000; 2911 } 2912 return time; 2913} 2914 2915uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2916{ 2917 uint32_t time; 2918 if (audio_is_linear_pcm(mFormat)) { 2919 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2920 } else { 2921 time = 10000; 2922 } 2923 return time; 2924} 2925 2926 2927// ---------------------------------------------------------------------------- 2928 2929AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2930 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2931{ 2932 mType = ThreadBase::DUPLICATING; 2933 addOutputTrack(mainThread); 2934} 2935 2936AudioFlinger::DuplicatingThread::~DuplicatingThread() 2937{ 2938 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2939 mOutputTracks[i]->destroy(); 2940 } 2941 mOutputTracks.clear(); 2942} 2943 2944bool AudioFlinger::DuplicatingThread::threadLoop() 2945{ 2946 Vector< sp<Track> > tracksToRemove; 2947 uint32_t mixerStatus = MIXER_IDLE; 2948 nsecs_t standbyTime = systemTime(); 2949 size_t mixBufferSize = mFrameCount*mFrameSize; 2950 SortedVector< sp<OutputTrack> > outputTracks; 2951 uint32_t writeFrames = 0; 2952 uint32_t activeSleepTime = activeSleepTimeUs(); 2953 uint32_t idleSleepTime = idleSleepTimeUs(); 2954 uint32_t sleepTime = idleSleepTime; 2955 Vector< sp<EffectChain> > effectChains; 2956 2957 acquireWakeLock(); 2958 2959 while (!exitPending()) 2960 { 2961 processConfigEvents(); 2962 2963 mixerStatus = MIXER_IDLE; 2964 { // scope for the mLock 2965 2966 Mutex::Autolock _l(mLock); 2967 2968 if (checkForNewParameters_l()) { 2969 mixBufferSize = mFrameCount*mFrameSize; 2970 updateWaitTime(); 2971 activeSleepTime = activeSleepTimeUs(); 2972 idleSleepTime = idleSleepTimeUs(); 2973 } 2974 2975 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2976 2977 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2978 outputTracks.add(mOutputTracks[i]); 2979 } 2980 2981 // put audio hardware into standby after short delay 2982 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2983 mSuspended) { 2984 if (!mStandby) { 2985 for (size_t i = 0; i < outputTracks.size(); i++) { 2986 outputTracks[i]->stop(); 2987 } 2988 mStandby = true; 2989 mBytesWritten = 0; 2990 } 2991 2992 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2993 // we're about to wait, flush the binder command buffer 2994 IPCThreadState::self()->flushCommands(); 2995 outputTracks.clear(); 2996 2997 if (exitPending()) break; 2998 2999 releaseWakeLock_l(); 3000 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3001 mWaitWorkCV.wait(mLock); 3002 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3003 acquireWakeLock_l(); 3004 3005 if (mMasterMute == false) { 3006 char value[PROPERTY_VALUE_MAX]; 3007 property_get("ro.audio.silent", value, "0"); 3008 if (atoi(value)) { 3009 LOGD("Silence is golden"); 3010 setMasterMute(true); 3011 } 3012 } 3013 3014 standbyTime = systemTime() + kStandbyTimeInNsecs; 3015 sleepTime = idleSleepTime; 3016 continue; 3017 } 3018 } 3019 3020 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3021 3022 // prevent any changes in effect chain list and in each effect chain 3023 // during mixing and effect process as the audio buffers could be deleted 3024 // or modified if an effect is created or deleted 3025 lockEffectChains_l(effectChains); 3026 } 3027 3028 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3029 // mix buffers... 3030 if (outputsReady(outputTracks)) { 3031 mAudioMixer->process(); 3032 } else { 3033 memset(mMixBuffer, 0, mixBufferSize); 3034 } 3035 sleepTime = 0; 3036 writeFrames = mFrameCount; 3037 } else { 3038 if (sleepTime == 0) { 3039 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3040 sleepTime = activeSleepTime; 3041 } else { 3042 sleepTime = idleSleepTime; 3043 } 3044 } else if (mBytesWritten != 0) { 3045 // flush remaining overflow buffers in output tracks 3046 for (size_t i = 0; i < outputTracks.size(); i++) { 3047 if (outputTracks[i]->isActive()) { 3048 sleepTime = 0; 3049 writeFrames = 0; 3050 memset(mMixBuffer, 0, mixBufferSize); 3051 break; 3052 } 3053 } 3054 } 3055 } 3056 3057 if (mSuspended) { 3058 sleepTime = suspendSleepTimeUs(); 3059 } 3060 // sleepTime == 0 means we must write to audio hardware 3061 if (sleepTime == 0) { 3062 for (size_t i = 0; i < effectChains.size(); i ++) { 3063 effectChains[i]->process_l(); 3064 } 3065 // enable changes in effect chain 3066 unlockEffectChains(effectChains); 3067 3068 standbyTime = systemTime() + kStandbyTimeInNsecs; 3069 for (size_t i = 0; i < outputTracks.size(); i++) { 3070 outputTracks[i]->write(mMixBuffer, writeFrames); 3071 } 3072 mStandby = false; 3073 mBytesWritten += mixBufferSize; 3074 } else { 3075 // enable changes in effect chain 3076 unlockEffectChains(effectChains); 3077 usleep(sleepTime); 3078 } 3079 3080 // finally let go of all our tracks, without the lock held 3081 // since we can't guarantee the destructors won't acquire that 3082 // same lock. 3083 tracksToRemove.clear(); 3084 outputTracks.clear(); 3085 3086 // Effect chains will be actually deleted here if they were removed from 3087 // mEffectChains list during mixing or effects processing 3088 effectChains.clear(); 3089 } 3090 3091 releaseWakeLock(); 3092 3093 return false; 3094} 3095 3096void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3097{ 3098 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3099 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3100 this, 3101 mSampleRate, 3102 mFormat, 3103 mChannelMask, 3104 frameCount); 3105 if (outputTrack->cblk() != NULL) { 3106 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3107 mOutputTracks.add(outputTrack); 3108 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3109 updateWaitTime(); 3110 } 3111} 3112 3113void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3114{ 3115 Mutex::Autolock _l(mLock); 3116 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3117 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3118 mOutputTracks[i]->destroy(); 3119 mOutputTracks.removeAt(i); 3120 updateWaitTime(); 3121 return; 3122 } 3123 } 3124 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3125} 3126 3127void AudioFlinger::DuplicatingThread::updateWaitTime() 3128{ 3129 mWaitTimeMs = UINT_MAX; 3130 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3131 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3132 if (strong != NULL) { 3133 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3134 if (waitTimeMs < mWaitTimeMs) { 3135 mWaitTimeMs = waitTimeMs; 3136 } 3137 } 3138 } 3139} 3140 3141 3142bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3143{ 3144 for (size_t i = 0; i < outputTracks.size(); i++) { 3145 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3146 if (thread == 0) { 3147 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3148 return false; 3149 } 3150 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3151 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3152 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3153 return false; 3154 } 3155 } 3156 return true; 3157} 3158 3159uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3160{ 3161 return (mWaitTimeMs * 1000) / 2; 3162} 3163 3164// ---------------------------------------------------------------------------- 3165 3166// TrackBase constructor must be called with AudioFlinger::mLock held 3167AudioFlinger::ThreadBase::TrackBase::TrackBase( 3168 const wp<ThreadBase>& thread, 3169 const sp<Client>& client, 3170 uint32_t sampleRate, 3171 uint32_t format, 3172 uint32_t channelMask, 3173 int frameCount, 3174 uint32_t flags, 3175 const sp<IMemory>& sharedBuffer, 3176 int sessionId) 3177 : RefBase(), 3178 mThread(thread), 3179 mClient(client), 3180 mCblk(0), 3181 mFrameCount(0), 3182 mState(IDLE), 3183 mClientTid(-1), 3184 mFormat(format), 3185 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3186 mSessionId(sessionId) 3187{ 3188 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3189 3190 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3191 size_t size = sizeof(audio_track_cblk_t); 3192 uint8_t channelCount = popcount(channelMask); 3193 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3194 if (sharedBuffer == 0) { 3195 size += bufferSize; 3196 } 3197 3198 if (client != NULL) { 3199 mCblkMemory = client->heap()->allocate(size); 3200 if (mCblkMemory != 0) { 3201 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3202 if (mCblk) { // construct the shared structure in-place. 3203 new(mCblk) audio_track_cblk_t(); 3204 // clear all buffers 3205 mCblk->frameCount = frameCount; 3206 mCblk->sampleRate = sampleRate; 3207 mChannelCount = channelCount; 3208 mChannelMask = channelMask; 3209 if (sharedBuffer == 0) { 3210 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3211 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3212 // Force underrun condition to avoid false underrun callback until first data is 3213 // written to buffer (other flags are cleared) 3214 mCblk->flags = CBLK_UNDERRUN_ON; 3215 } else { 3216 mBuffer = sharedBuffer->pointer(); 3217 } 3218 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3219 } 3220 } else { 3221 LOGE("not enough memory for AudioTrack size=%u", size); 3222 client->heap()->dump("AudioTrack"); 3223 return; 3224 } 3225 } else { 3226 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 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 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3235 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3236 // Force underrun condition to avoid false underrun callback until first data is 3237 // written to buffer (other flags are cleared) 3238 mCblk->flags = CBLK_UNDERRUN_ON; 3239 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3240 } 3241 } 3242} 3243 3244AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3245{ 3246 if (mCblk) { 3247 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3248 if (mClient == NULL) { 3249 delete mCblk; 3250 } 3251 } 3252 mCblkMemory.clear(); // and free the shared memory 3253 if (mClient != NULL) { 3254 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3255 mClient.clear(); 3256 } 3257} 3258 3259void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3260{ 3261 buffer->raw = 0; 3262 mFrameCount = buffer->frameCount; 3263 step(); 3264 buffer->frameCount = 0; 3265} 3266 3267bool AudioFlinger::ThreadBase::TrackBase::step() { 3268 bool result; 3269 audio_track_cblk_t* cblk = this->cblk(); 3270 3271 result = cblk->stepServer(mFrameCount); 3272 if (!result) { 3273 ALOGV("stepServer failed acquiring cblk mutex"); 3274 mFlags |= STEPSERVER_FAILED; 3275 } 3276 return result; 3277} 3278 3279void AudioFlinger::ThreadBase::TrackBase::reset() { 3280 audio_track_cblk_t* cblk = this->cblk(); 3281 3282 cblk->user = 0; 3283 cblk->server = 0; 3284 cblk->userBase = 0; 3285 cblk->serverBase = 0; 3286 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3287 ALOGV("TrackBase::reset"); 3288} 3289 3290sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3291{ 3292 return mCblkMemory; 3293} 3294 3295int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3296 return (int)mCblk->sampleRate; 3297} 3298 3299int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3300 return (const int)mChannelCount; 3301} 3302 3303uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3304 return mChannelMask; 3305} 3306 3307void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3308 audio_track_cblk_t* cblk = this->cblk(); 3309 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 3310 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 3311 3312 // Check validity of returned pointer in case the track control block would have been corrupted. 3313 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3314 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 3315 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3316 server %d, serverBase %d, user %d, userBase %d", 3317 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3318 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3319 return 0; 3320 } 3321 3322 return bufferStart; 3323} 3324 3325// ---------------------------------------------------------------------------- 3326 3327// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3328AudioFlinger::PlaybackThread::Track::Track( 3329 const wp<ThreadBase>& thread, 3330 const sp<Client>& client, 3331 int streamType, 3332 uint32_t sampleRate, 3333 uint32_t format, 3334 uint32_t channelMask, 3335 int frameCount, 3336 const sp<IMemory>& sharedBuffer, 3337 int sessionId) 3338 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3339 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3340 mAuxEffectId(0), mHasVolumeController(false) 3341{ 3342 if (mCblk != NULL) { 3343 sp<ThreadBase> baseThread = thread.promote(); 3344 if (baseThread != 0) { 3345 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3346 mName = playbackThread->getTrackName_l(); 3347 mMainBuffer = playbackThread->mixBuffer(); 3348 } 3349 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3350 if (mName < 0) { 3351 LOGE("no more track names available"); 3352 } 3353 mVolume[0] = 1.0f; 3354 mVolume[1] = 1.0f; 3355 mStreamType = streamType; 3356 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3357 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3358 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3359 } 3360} 3361 3362AudioFlinger::PlaybackThread::Track::~Track() 3363{ 3364 ALOGV("PlaybackThread::Track destructor"); 3365 sp<ThreadBase> thread = mThread.promote(); 3366 if (thread != 0) { 3367 Mutex::Autolock _l(thread->mLock); 3368 mState = TERMINATED; 3369 } 3370} 3371 3372void AudioFlinger::PlaybackThread::Track::destroy() 3373{ 3374 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3375 // by removing it from mTracks vector, so there is a risk that this Tracks's 3376 // desctructor is called. As the destructor needs to lock mLock, 3377 // we must acquire a strong reference on this Track before locking mLock 3378 // here so that the destructor is called only when exiting this function. 3379 // On the other hand, as long as Track::destroy() is only called by 3380 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3381 // this Track with its member mTrack. 3382 sp<Track> keep(this); 3383 { // scope for mLock 3384 sp<ThreadBase> thread = mThread.promote(); 3385 if (thread != 0) { 3386 if (!isOutputTrack()) { 3387 if (mState == ACTIVE || mState == RESUMING) { 3388 AudioSystem::stopOutput(thread->id(), 3389 (audio_stream_type_t)mStreamType, 3390 mSessionId); 3391 3392 // to track the speaker usage 3393 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3394 } 3395 AudioSystem::releaseOutput(thread->id()); 3396 } 3397 Mutex::Autolock _l(thread->mLock); 3398 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3399 playbackThread->destroyTrack_l(this); 3400 } 3401 } 3402} 3403 3404void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3405{ 3406 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", 3407 mName - AudioMixer::TRACK0, 3408 (mClient == NULL) ? getpid() : mClient->pid(), 3409 mStreamType, 3410 mFormat, 3411 mChannelMask, 3412 mSessionId, 3413 mFrameCount, 3414 mState, 3415 mMute, 3416 mFillingUpStatus, 3417 mCblk->sampleRate, 3418 mCblk->volume[0], 3419 mCblk->volume[1], 3420 mCblk->server, 3421 mCblk->user, 3422 (int)mMainBuffer, 3423 (int)mAuxBuffer); 3424} 3425 3426status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3427{ 3428 audio_track_cblk_t* cblk = this->cblk(); 3429 uint32_t framesReady; 3430 uint32_t framesReq = buffer->frameCount; 3431 3432 // Check if last stepServer failed, try to step now 3433 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3434 if (!step()) goto getNextBuffer_exit; 3435 ALOGV("stepServer recovered"); 3436 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3437 } 3438 3439 framesReady = cblk->framesReady(); 3440 3441 if (LIKELY(framesReady)) { 3442 uint32_t s = cblk->server; 3443 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3444 3445 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3446 if (framesReq > framesReady) { 3447 framesReq = framesReady; 3448 } 3449 if (s + framesReq > bufferEnd) { 3450 framesReq = bufferEnd - s; 3451 } 3452 3453 buffer->raw = getBuffer(s, framesReq); 3454 if (buffer->raw == 0) goto getNextBuffer_exit; 3455 3456 buffer->frameCount = framesReq; 3457 return NO_ERROR; 3458 } 3459 3460getNextBuffer_exit: 3461 buffer->raw = 0; 3462 buffer->frameCount = 0; 3463 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3464 return NOT_ENOUGH_DATA; 3465} 3466 3467bool AudioFlinger::PlaybackThread::Track::isReady() const { 3468 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3469 3470 if (mCblk->framesReady() >= mCblk->frameCount || 3471 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3472 mFillingUpStatus = FS_FILLED; 3473 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3474 return true; 3475 } 3476 return false; 3477} 3478 3479status_t AudioFlinger::PlaybackThread::Track::start() 3480{ 3481 status_t status = NO_ERROR; 3482 ALOGV("start(%d), calling thread %d session %d", 3483 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3484 sp<ThreadBase> thread = mThread.promote(); 3485 if (thread != 0) { 3486 Mutex::Autolock _l(thread->mLock); 3487 int state = mState; 3488 // here the track could be either new, or restarted 3489 // in both cases "unstop" the track 3490 if (mState == PAUSED) { 3491 mState = TrackBase::RESUMING; 3492 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3493 } else { 3494 mState = TrackBase::ACTIVE; 3495 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3496 } 3497 3498 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3499 thread->mLock.unlock(); 3500 status = AudioSystem::startOutput(thread->id(), 3501 (audio_stream_type_t)mStreamType, 3502 mSessionId); 3503 thread->mLock.lock(); 3504 3505 // to track the speaker usage 3506 if (status == NO_ERROR) { 3507 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3508 } 3509 } 3510 if (status == NO_ERROR) { 3511 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3512 playbackThread->addTrack_l(this); 3513 } else { 3514 mState = state; 3515 } 3516 } else { 3517 status = BAD_VALUE; 3518 } 3519 return status; 3520} 3521 3522void AudioFlinger::PlaybackThread::Track::stop() 3523{ 3524 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3525 sp<ThreadBase> thread = mThread.promote(); 3526 if (thread != 0) { 3527 Mutex::Autolock _l(thread->mLock); 3528 int state = mState; 3529 if (mState > STOPPED) { 3530 mState = STOPPED; 3531 // If the track is not active (PAUSED and buffers full), flush buffers 3532 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3533 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3534 reset(); 3535 } 3536 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3537 } 3538 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3539 thread->mLock.unlock(); 3540 AudioSystem::stopOutput(thread->id(), 3541 (audio_stream_type_t)mStreamType, 3542 mSessionId); 3543 thread->mLock.lock(); 3544 3545 // to track the speaker usage 3546 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3547 } 3548 } 3549} 3550 3551void AudioFlinger::PlaybackThread::Track::pause() 3552{ 3553 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3554 sp<ThreadBase> thread = mThread.promote(); 3555 if (thread != 0) { 3556 Mutex::Autolock _l(thread->mLock); 3557 if (mState == ACTIVE || mState == RESUMING) { 3558 mState = PAUSING; 3559 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3560 if (!isOutputTrack()) { 3561 thread->mLock.unlock(); 3562 AudioSystem::stopOutput(thread->id(), 3563 (audio_stream_type_t)mStreamType, 3564 mSessionId); 3565 thread->mLock.lock(); 3566 3567 // to track the speaker usage 3568 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3569 } 3570 } 3571 } 3572} 3573 3574void AudioFlinger::PlaybackThread::Track::flush() 3575{ 3576 ALOGV("flush(%d)", mName); 3577 sp<ThreadBase> thread = mThread.promote(); 3578 if (thread != 0) { 3579 Mutex::Autolock _l(thread->mLock); 3580 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3581 return; 3582 } 3583 // No point remaining in PAUSED state after a flush => go to 3584 // STOPPED state 3585 mState = STOPPED; 3586 3587 // do not reset the track if it is still in the process of being stopped or paused. 3588 // this will be done by prepareTracks_l() when the track is stopped. 3589 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3590 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3591 reset(); 3592 } 3593 } 3594} 3595 3596void AudioFlinger::PlaybackThread::Track::reset() 3597{ 3598 // Do not reset twice to avoid discarding data written just after a flush and before 3599 // the audioflinger thread detects the track is stopped. 3600 if (!mResetDone) { 3601 TrackBase::reset(); 3602 // Force underrun condition to avoid false underrun callback until first data is 3603 // written to buffer 3604 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3605 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3606 mFillingUpStatus = FS_FILLING; 3607 mResetDone = true; 3608 } 3609} 3610 3611void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3612{ 3613 mMute = muted; 3614} 3615 3616void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3617{ 3618 mVolume[0] = left; 3619 mVolume[1] = right; 3620} 3621 3622status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3623{ 3624 status_t status = DEAD_OBJECT; 3625 sp<ThreadBase> thread = mThread.promote(); 3626 if (thread != 0) { 3627 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3628 status = playbackThread->attachAuxEffect(this, EffectId); 3629 } 3630 return status; 3631} 3632 3633void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3634{ 3635 mAuxEffectId = EffectId; 3636 mAuxBuffer = buffer; 3637} 3638 3639// ---------------------------------------------------------------------------- 3640 3641// RecordTrack constructor must be called with AudioFlinger::mLock held 3642AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3643 const wp<ThreadBase>& thread, 3644 const sp<Client>& client, 3645 uint32_t sampleRate, 3646 uint32_t format, 3647 uint32_t channelMask, 3648 int frameCount, 3649 uint32_t flags, 3650 int sessionId) 3651 : TrackBase(thread, client, sampleRate, format, 3652 channelMask, frameCount, flags, 0, sessionId), 3653 mOverflow(false) 3654{ 3655 if (mCblk != NULL) { 3656 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3657 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3658 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3659 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3660 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3661 } else { 3662 mCblk->frameSize = sizeof(int8_t); 3663 } 3664 } 3665} 3666 3667AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3668{ 3669 sp<ThreadBase> thread = mThread.promote(); 3670 if (thread != 0) { 3671 AudioSystem::releaseInput(thread->id()); 3672 } 3673} 3674 3675status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3676{ 3677 audio_track_cblk_t* cblk = this->cblk(); 3678 uint32_t framesAvail; 3679 uint32_t framesReq = buffer->frameCount; 3680 3681 // Check if last stepServer failed, try to step now 3682 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3683 if (!step()) goto getNextBuffer_exit; 3684 ALOGV("stepServer recovered"); 3685 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3686 } 3687 3688 framesAvail = cblk->framesAvailable_l(); 3689 3690 if (LIKELY(framesAvail)) { 3691 uint32_t s = cblk->server; 3692 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3693 3694 if (framesReq > framesAvail) { 3695 framesReq = framesAvail; 3696 } 3697 if (s + framesReq > bufferEnd) { 3698 framesReq = bufferEnd - s; 3699 } 3700 3701 buffer->raw = getBuffer(s, framesReq); 3702 if (buffer->raw == 0) goto getNextBuffer_exit; 3703 3704 buffer->frameCount = framesReq; 3705 return NO_ERROR; 3706 } 3707 3708getNextBuffer_exit: 3709 buffer->raw = 0; 3710 buffer->frameCount = 0; 3711 return NOT_ENOUGH_DATA; 3712} 3713 3714status_t AudioFlinger::RecordThread::RecordTrack::start() 3715{ 3716 sp<ThreadBase> thread = mThread.promote(); 3717 if (thread != 0) { 3718 RecordThread *recordThread = (RecordThread *)thread.get(); 3719 return recordThread->start(this); 3720 } else { 3721 return BAD_VALUE; 3722 } 3723} 3724 3725void AudioFlinger::RecordThread::RecordTrack::stop() 3726{ 3727 sp<ThreadBase> thread = mThread.promote(); 3728 if (thread != 0) { 3729 RecordThread *recordThread = (RecordThread *)thread.get(); 3730 recordThread->stop(this); 3731 TrackBase::reset(); 3732 // Force overerrun condition to avoid false overrun callback until first data is 3733 // read from buffer 3734 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3735 } 3736} 3737 3738void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3739{ 3740 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3741 (mClient == NULL) ? getpid() : mClient->pid(), 3742 mFormat, 3743 mChannelMask, 3744 mSessionId, 3745 mFrameCount, 3746 mState, 3747 mCblk->sampleRate, 3748 mCblk->server, 3749 mCblk->user); 3750} 3751 3752 3753// ---------------------------------------------------------------------------- 3754 3755AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3756 const wp<ThreadBase>& thread, 3757 DuplicatingThread *sourceThread, 3758 uint32_t sampleRate, 3759 uint32_t format, 3760 uint32_t channelMask, 3761 int frameCount) 3762 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3763 mActive(false), mSourceThread(sourceThread) 3764{ 3765 3766 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3767 if (mCblk != NULL) { 3768 mCblk->flags |= CBLK_DIRECTION_OUT; 3769 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3770 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3771 mOutBuffer.frameCount = 0; 3772 playbackThread->mTracks.add(this); 3773 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3774 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3775 mCblk, mBuffer, mCblk->buffers, 3776 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3777 } else { 3778 LOGW("Error creating output track on thread %p", playbackThread); 3779 } 3780} 3781 3782AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3783{ 3784 clearBufferQueue(); 3785} 3786 3787status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3788{ 3789 status_t status = Track::start(); 3790 if (status != NO_ERROR) { 3791 return status; 3792 } 3793 3794 mActive = true; 3795 mRetryCount = 127; 3796 return status; 3797} 3798 3799void AudioFlinger::PlaybackThread::OutputTrack::stop() 3800{ 3801 Track::stop(); 3802 clearBufferQueue(); 3803 mOutBuffer.frameCount = 0; 3804 mActive = false; 3805} 3806 3807bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3808{ 3809 Buffer *pInBuffer; 3810 Buffer inBuffer; 3811 uint32_t channelCount = mChannelCount; 3812 bool outputBufferFull = false; 3813 inBuffer.frameCount = frames; 3814 inBuffer.i16 = data; 3815 3816 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3817 3818 if (!mActive && frames != 0) { 3819 start(); 3820 sp<ThreadBase> thread = mThread.promote(); 3821 if (thread != 0) { 3822 MixerThread *mixerThread = (MixerThread *)thread.get(); 3823 if (mCblk->frameCount > frames){ 3824 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3825 uint32_t startFrames = (mCblk->frameCount - frames); 3826 pInBuffer = new Buffer; 3827 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3828 pInBuffer->frameCount = startFrames; 3829 pInBuffer->i16 = pInBuffer->mBuffer; 3830 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3831 mBufferQueue.add(pInBuffer); 3832 } else { 3833 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3834 } 3835 } 3836 } 3837 } 3838 3839 while (waitTimeLeftMs) { 3840 // First write pending buffers, then new data 3841 if (mBufferQueue.size()) { 3842 pInBuffer = mBufferQueue.itemAt(0); 3843 } else { 3844 pInBuffer = &inBuffer; 3845 } 3846 3847 if (pInBuffer->frameCount == 0) { 3848 break; 3849 } 3850 3851 if (mOutBuffer.frameCount == 0) { 3852 mOutBuffer.frameCount = pInBuffer->frameCount; 3853 nsecs_t startTime = systemTime(); 3854 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3855 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3856 outputBufferFull = true; 3857 break; 3858 } 3859 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3860 if (waitTimeLeftMs >= waitTimeMs) { 3861 waitTimeLeftMs -= waitTimeMs; 3862 } else { 3863 waitTimeLeftMs = 0; 3864 } 3865 } 3866 3867 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3868 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3869 mCblk->stepUser(outFrames); 3870 pInBuffer->frameCount -= outFrames; 3871 pInBuffer->i16 += outFrames * channelCount; 3872 mOutBuffer.frameCount -= outFrames; 3873 mOutBuffer.i16 += outFrames * channelCount; 3874 3875 if (pInBuffer->frameCount == 0) { 3876 if (mBufferQueue.size()) { 3877 mBufferQueue.removeAt(0); 3878 delete [] pInBuffer->mBuffer; 3879 delete pInBuffer; 3880 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3881 } else { 3882 break; 3883 } 3884 } 3885 } 3886 3887 // If we could not write all frames, allocate a buffer and queue it for next time. 3888 if (inBuffer.frameCount) { 3889 sp<ThreadBase> thread = mThread.promote(); 3890 if (thread != 0 && !thread->standby()) { 3891 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3892 pInBuffer = new Buffer; 3893 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3894 pInBuffer->frameCount = inBuffer.frameCount; 3895 pInBuffer->i16 = pInBuffer->mBuffer; 3896 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3897 mBufferQueue.add(pInBuffer); 3898 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3899 } else { 3900 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3901 } 3902 } 3903 } 3904 3905 // Calling write() with a 0 length buffer, means that no more data will be written: 3906 // If no more buffers are pending, fill output track buffer to make sure it is started 3907 // by output mixer. 3908 if (frames == 0 && mBufferQueue.size() == 0) { 3909 if (mCblk->user < mCblk->frameCount) { 3910 frames = mCblk->frameCount - mCblk->user; 3911 pInBuffer = new Buffer; 3912 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3913 pInBuffer->frameCount = frames; 3914 pInBuffer->i16 = pInBuffer->mBuffer; 3915 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3916 mBufferQueue.add(pInBuffer); 3917 } else if (mActive) { 3918 stop(); 3919 } 3920 } 3921 3922 return outputBufferFull; 3923} 3924 3925status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3926{ 3927 int active; 3928 status_t result; 3929 audio_track_cblk_t* cblk = mCblk; 3930 uint32_t framesReq = buffer->frameCount; 3931 3932// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3933 buffer->frameCount = 0; 3934 3935 uint32_t framesAvail = cblk->framesAvailable(); 3936 3937 3938 if (framesAvail == 0) { 3939 Mutex::Autolock _l(cblk->lock); 3940 goto start_loop_here; 3941 while (framesAvail == 0) { 3942 active = mActive; 3943 if (UNLIKELY(!active)) { 3944 ALOGV("Not active and NO_MORE_BUFFERS"); 3945 return AudioTrack::NO_MORE_BUFFERS; 3946 } 3947 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3948 if (result != NO_ERROR) { 3949 return AudioTrack::NO_MORE_BUFFERS; 3950 } 3951 // read the server count again 3952 start_loop_here: 3953 framesAvail = cblk->framesAvailable_l(); 3954 } 3955 } 3956 3957// if (framesAvail < framesReq) { 3958// return AudioTrack::NO_MORE_BUFFERS; 3959// } 3960 3961 if (framesReq > framesAvail) { 3962 framesReq = framesAvail; 3963 } 3964 3965 uint32_t u = cblk->user; 3966 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3967 3968 if (u + framesReq > bufferEnd) { 3969 framesReq = bufferEnd - u; 3970 } 3971 3972 buffer->frameCount = framesReq; 3973 buffer->raw = (void *)cblk->buffer(u); 3974 return NO_ERROR; 3975} 3976 3977 3978void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3979{ 3980 size_t size = mBufferQueue.size(); 3981 Buffer *pBuffer; 3982 3983 for (size_t i = 0; i < size; i++) { 3984 pBuffer = mBufferQueue.itemAt(i); 3985 delete [] pBuffer->mBuffer; 3986 delete pBuffer; 3987 } 3988 mBufferQueue.clear(); 3989} 3990 3991// ---------------------------------------------------------------------------- 3992 3993AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3994 : RefBase(), 3995 mAudioFlinger(audioFlinger), 3996 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3997 mPid(pid) 3998{ 3999 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4000} 4001 4002// Client destructor must be called with AudioFlinger::mLock held 4003AudioFlinger::Client::~Client() 4004{ 4005 mAudioFlinger->removeClient_l(mPid); 4006} 4007 4008const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4009{ 4010 return mMemoryDealer; 4011} 4012 4013// ---------------------------------------------------------------------------- 4014 4015AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4016 const sp<IAudioFlingerClient>& client, 4017 pid_t pid) 4018 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4019{ 4020} 4021 4022AudioFlinger::NotificationClient::~NotificationClient() 4023{ 4024 mClient.clear(); 4025} 4026 4027void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4028{ 4029 sp<NotificationClient> keep(this); 4030 { 4031 mAudioFlinger->removeNotificationClient(mPid); 4032 } 4033} 4034 4035// ---------------------------------------------------------------------------- 4036 4037AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4038 : BnAudioTrack(), 4039 mTrack(track) 4040{ 4041} 4042 4043AudioFlinger::TrackHandle::~TrackHandle() { 4044 // just stop the track on deletion, associated resources 4045 // will be freed from the main thread once all pending buffers have 4046 // been played. Unless it's not in the active track list, in which 4047 // case we free everything now... 4048 mTrack->destroy(); 4049} 4050 4051status_t AudioFlinger::TrackHandle::start() { 4052 return mTrack->start(); 4053} 4054 4055void AudioFlinger::TrackHandle::stop() { 4056 mTrack->stop(); 4057} 4058 4059void AudioFlinger::TrackHandle::flush() { 4060 mTrack->flush(); 4061} 4062 4063void AudioFlinger::TrackHandle::mute(bool e) { 4064 mTrack->mute(e); 4065} 4066 4067void AudioFlinger::TrackHandle::pause() { 4068 mTrack->pause(); 4069} 4070 4071void AudioFlinger::TrackHandle::setVolume(float left, float right) { 4072 mTrack->setVolume(left, right); 4073} 4074 4075sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4076 return mTrack->getCblk(); 4077} 4078 4079status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4080{ 4081 return mTrack->attachAuxEffect(EffectId); 4082} 4083 4084status_t AudioFlinger::TrackHandle::onTransact( 4085 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4086{ 4087 return BnAudioTrack::onTransact(code, data, reply, flags); 4088} 4089 4090// ---------------------------------------------------------------------------- 4091 4092sp<IAudioRecord> AudioFlinger::openRecord( 4093 pid_t pid, 4094 int input, 4095 uint32_t sampleRate, 4096 uint32_t format, 4097 uint32_t channelMask, 4098 int frameCount, 4099 uint32_t flags, 4100 int *sessionId, 4101 status_t *status) 4102{ 4103 sp<RecordThread::RecordTrack> recordTrack; 4104 sp<RecordHandle> recordHandle; 4105 sp<Client> client; 4106 wp<Client> wclient; 4107 status_t lStatus; 4108 RecordThread *thread; 4109 size_t inFrameCount; 4110 int lSessionId; 4111 4112 // check calling permissions 4113 if (!recordingAllowed()) { 4114 lStatus = PERMISSION_DENIED; 4115 goto Exit; 4116 } 4117 4118 // add client to list 4119 { // scope for mLock 4120 Mutex::Autolock _l(mLock); 4121 thread = checkRecordThread_l(input); 4122 if (thread == NULL) { 4123 lStatus = BAD_VALUE; 4124 goto Exit; 4125 } 4126 4127 wclient = mClients.valueFor(pid); 4128 if (wclient != NULL) { 4129 client = wclient.promote(); 4130 } else { 4131 client = new Client(this, pid); 4132 mClients.add(pid, client); 4133 } 4134 4135 // If no audio session id is provided, create one here 4136 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4137 lSessionId = *sessionId; 4138 } else { 4139 lSessionId = nextUniqueId(); 4140 if (sessionId != NULL) { 4141 *sessionId = lSessionId; 4142 } 4143 } 4144 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4145 recordTrack = thread->createRecordTrack_l(client, 4146 sampleRate, 4147 format, 4148 channelMask, 4149 frameCount, 4150 flags, 4151 lSessionId, 4152 &lStatus); 4153 } 4154 if (lStatus != NO_ERROR) { 4155 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4156 // destructor is called by the TrackBase destructor with mLock held 4157 client.clear(); 4158 recordTrack.clear(); 4159 goto Exit; 4160 } 4161 4162 // return to handle to client 4163 recordHandle = new RecordHandle(recordTrack); 4164 lStatus = NO_ERROR; 4165 4166Exit: 4167 if (status) { 4168 *status = lStatus; 4169 } 4170 return recordHandle; 4171} 4172 4173// ---------------------------------------------------------------------------- 4174 4175AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4176 : BnAudioRecord(), 4177 mRecordTrack(recordTrack) 4178{ 4179} 4180 4181AudioFlinger::RecordHandle::~RecordHandle() { 4182 stop(); 4183} 4184 4185status_t AudioFlinger::RecordHandle::start() { 4186 ALOGV("RecordHandle::start()"); 4187 return mRecordTrack->start(); 4188} 4189 4190void AudioFlinger::RecordHandle::stop() { 4191 ALOGV("RecordHandle::stop()"); 4192 mRecordTrack->stop(); 4193} 4194 4195sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4196 return mRecordTrack->getCblk(); 4197} 4198 4199status_t AudioFlinger::RecordHandle::onTransact( 4200 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4201{ 4202 return BnAudioRecord::onTransact(code, data, reply, flags); 4203} 4204 4205// ---------------------------------------------------------------------------- 4206 4207AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4208 AudioStreamIn *input, 4209 uint32_t sampleRate, 4210 uint32_t channels, 4211 int id, 4212 uint32_t device) : 4213 ThreadBase(audioFlinger, id, device), 4214 mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 4215{ 4216 mType = ThreadBase::RECORD; 4217 4218 snprintf(mName, kNameLength, "AudioIn_%d", id); 4219 4220 mReqChannelCount = popcount(channels); 4221 mReqSampleRate = sampleRate; 4222 readInputParameters(); 4223} 4224 4225 4226AudioFlinger::RecordThread::~RecordThread() 4227{ 4228 delete[] mRsmpInBuffer; 4229 if (mResampler != 0) { 4230 delete mResampler; 4231 delete[] mRsmpOutBuffer; 4232 } 4233} 4234 4235void AudioFlinger::RecordThread::onFirstRef() 4236{ 4237 run(mName, PRIORITY_URGENT_AUDIO); 4238} 4239 4240status_t AudioFlinger::RecordThread::readyToRun() 4241{ 4242 status_t status = initCheck(); 4243 LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4244 return status; 4245} 4246 4247bool AudioFlinger::RecordThread::threadLoop() 4248{ 4249 AudioBufferProvider::Buffer buffer; 4250 sp<RecordTrack> activeTrack; 4251 Vector< sp<EffectChain> > effectChains; 4252 4253 nsecs_t lastWarning = 0; 4254 4255 acquireWakeLock(); 4256 4257 // start recording 4258 while (!exitPending()) { 4259 4260 processConfigEvents(); 4261 4262 { // scope for mLock 4263 Mutex::Autolock _l(mLock); 4264 checkForNewParameters_l(); 4265 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4266 if (!mStandby) { 4267 mInput->stream->common.standby(&mInput->stream->common); 4268 mStandby = true; 4269 } 4270 4271 if (exitPending()) break; 4272 4273 releaseWakeLock_l(); 4274 ALOGV("RecordThread: loop stopping"); 4275 // go to sleep 4276 mWaitWorkCV.wait(mLock); 4277 ALOGV("RecordThread: loop starting"); 4278 acquireWakeLock_l(); 4279 continue; 4280 } 4281 if (mActiveTrack != 0) { 4282 if (mActiveTrack->mState == TrackBase::PAUSING) { 4283 if (!mStandby) { 4284 mInput->stream->common.standby(&mInput->stream->common); 4285 mStandby = true; 4286 } 4287 mActiveTrack.clear(); 4288 mStartStopCond.broadcast(); 4289 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4290 if (mReqChannelCount != mActiveTrack->channelCount()) { 4291 mActiveTrack.clear(); 4292 mStartStopCond.broadcast(); 4293 } else if (mBytesRead != 0) { 4294 // record start succeeds only if first read from audio input 4295 // succeeds 4296 if (mBytesRead > 0) { 4297 mActiveTrack->mState = TrackBase::ACTIVE; 4298 } else { 4299 mActiveTrack.clear(); 4300 } 4301 mStartStopCond.broadcast(); 4302 } 4303 mStandby = false; 4304 } 4305 } 4306 lockEffectChains_l(effectChains); 4307 } 4308 4309 if (mActiveTrack != 0) { 4310 if (mActiveTrack->mState != TrackBase::ACTIVE && 4311 mActiveTrack->mState != TrackBase::RESUMING) { 4312 unlockEffectChains(effectChains); 4313 usleep(kRecordThreadSleepUs); 4314 continue; 4315 } 4316 for (size_t i = 0; i < effectChains.size(); i ++) { 4317 effectChains[i]->process_l(); 4318 } 4319 4320 buffer.frameCount = mFrameCount; 4321 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4322 size_t framesOut = buffer.frameCount; 4323 if (mResampler == 0) { 4324 // no resampling 4325 while (framesOut) { 4326 size_t framesIn = mFrameCount - mRsmpInIndex; 4327 if (framesIn) { 4328 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4329 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4330 if (framesIn > framesOut) 4331 framesIn = framesOut; 4332 mRsmpInIndex += framesIn; 4333 framesOut -= framesIn; 4334 if ((int)mChannelCount == mReqChannelCount || 4335 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4336 memcpy(dst, src, framesIn * mFrameSize); 4337 } else { 4338 int16_t *src16 = (int16_t *)src; 4339 int16_t *dst16 = (int16_t *)dst; 4340 if (mChannelCount == 1) { 4341 while (framesIn--) { 4342 *dst16++ = *src16; 4343 *dst16++ = *src16++; 4344 } 4345 } else { 4346 while (framesIn--) { 4347 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4348 src16 += 2; 4349 } 4350 } 4351 } 4352 } 4353 if (framesOut && mFrameCount == mRsmpInIndex) { 4354 if (framesOut == mFrameCount && 4355 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4356 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4357 framesOut = 0; 4358 } else { 4359 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4360 mRsmpInIndex = 0; 4361 } 4362 if (mBytesRead < 0) { 4363 LOGE("Error reading audio input"); 4364 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4365 // Force input into standby so that it tries to 4366 // recover at next read attempt 4367 mInput->stream->common.standby(&mInput->stream->common); 4368 usleep(kRecordThreadSleepUs); 4369 } 4370 mRsmpInIndex = mFrameCount; 4371 framesOut = 0; 4372 buffer.frameCount = 0; 4373 } 4374 } 4375 } 4376 } else { 4377 // resampling 4378 4379 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4380 // alter output frame count as if we were expecting stereo samples 4381 if (mChannelCount == 1 && mReqChannelCount == 1) { 4382 framesOut >>= 1; 4383 } 4384 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4385 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4386 // are 32 bit aligned which should be always true. 4387 if (mChannelCount == 2 && mReqChannelCount == 1) { 4388 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4389 // the resampler always outputs stereo samples: do post stereo to mono conversion 4390 int16_t *src = (int16_t *)mRsmpOutBuffer; 4391 int16_t *dst = buffer.i16; 4392 while (framesOut--) { 4393 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4394 src += 2; 4395 } 4396 } else { 4397 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4398 } 4399 4400 } 4401 mActiveTrack->releaseBuffer(&buffer); 4402 mActiveTrack->overflow(); 4403 } 4404 // client isn't retrieving buffers fast enough 4405 else { 4406 if (!mActiveTrack->setOverflow()) { 4407 nsecs_t now = systemTime(); 4408 if ((now - lastWarning) > kWarningThrottle) { 4409 LOGW("RecordThread: buffer overflow"); 4410 lastWarning = now; 4411 } 4412 } 4413 // Release the processor for a while before asking for a new buffer. 4414 // This will give the application more chance to read from the buffer and 4415 // clear the overflow. 4416 usleep(kRecordThreadSleepUs); 4417 } 4418 } 4419 // enable changes in effect chain 4420 unlockEffectChains(effectChains); 4421 effectChains.clear(); 4422 } 4423 4424 if (!mStandby) { 4425 mInput->stream->common.standby(&mInput->stream->common); 4426 } 4427 mActiveTrack.clear(); 4428 4429 mStartStopCond.broadcast(); 4430 4431 releaseWakeLock(); 4432 4433 ALOGV("RecordThread %p exiting", this); 4434 return false; 4435} 4436 4437 4438sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4439 const sp<AudioFlinger::Client>& client, 4440 uint32_t sampleRate, 4441 int format, 4442 int channelMask, 4443 int frameCount, 4444 uint32_t flags, 4445 int sessionId, 4446 status_t *status) 4447{ 4448 sp<RecordTrack> track; 4449 status_t lStatus; 4450 4451 lStatus = initCheck(); 4452 if (lStatus != NO_ERROR) { 4453 LOGE("Audio driver not initialized."); 4454 goto Exit; 4455 } 4456 4457 { // scope for mLock 4458 Mutex::Autolock _l(mLock); 4459 4460 track = new RecordTrack(this, client, sampleRate, 4461 format, channelMask, frameCount, flags, sessionId); 4462 4463 if (track->getCblk() == NULL) { 4464 lStatus = NO_MEMORY; 4465 goto Exit; 4466 } 4467 4468 mTrack = track.get(); 4469 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4470 bool suspend = audio_is_bluetooth_sco_device( 4471 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4472 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4473 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4474 } 4475 lStatus = NO_ERROR; 4476 4477Exit: 4478 if (status) { 4479 *status = lStatus; 4480 } 4481 return track; 4482} 4483 4484status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4485{ 4486 ALOGV("RecordThread::start"); 4487 sp <ThreadBase> strongMe = this; 4488 status_t status = NO_ERROR; 4489 { 4490 AutoMutex lock(&mLock); 4491 if (mActiveTrack != 0) { 4492 if (recordTrack != mActiveTrack.get()) { 4493 status = -EBUSY; 4494 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4495 mActiveTrack->mState = TrackBase::ACTIVE; 4496 } 4497 return status; 4498 } 4499 4500 recordTrack->mState = TrackBase::IDLE; 4501 mActiveTrack = recordTrack; 4502 mLock.unlock(); 4503 status_t status = AudioSystem::startInput(mId); 4504 mLock.lock(); 4505 if (status != NO_ERROR) { 4506 mActiveTrack.clear(); 4507 return status; 4508 } 4509 mRsmpInIndex = mFrameCount; 4510 mBytesRead = 0; 4511 if (mResampler != NULL) { 4512 mResampler->reset(); 4513 } 4514 mActiveTrack->mState = TrackBase::RESUMING; 4515 // signal thread to start 4516 ALOGV("Signal record thread"); 4517 mWaitWorkCV.signal(); 4518 // do not wait for mStartStopCond if exiting 4519 if (mExiting) { 4520 mActiveTrack.clear(); 4521 status = INVALID_OPERATION; 4522 goto startError; 4523 } 4524 mStartStopCond.wait(mLock); 4525 if (mActiveTrack == 0) { 4526 ALOGV("Record failed to start"); 4527 status = BAD_VALUE; 4528 goto startError; 4529 } 4530 ALOGV("Record started OK"); 4531 return status; 4532 } 4533startError: 4534 AudioSystem::stopInput(mId); 4535 return status; 4536} 4537 4538void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4539 ALOGV("RecordThread::stop"); 4540 sp <ThreadBase> strongMe = this; 4541 { 4542 AutoMutex lock(&mLock); 4543 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4544 mActiveTrack->mState = TrackBase::PAUSING; 4545 // do not wait for mStartStopCond if exiting 4546 if (mExiting) { 4547 return; 4548 } 4549 mStartStopCond.wait(mLock); 4550 // if we have been restarted, recordTrack == mActiveTrack.get() here 4551 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4552 mLock.unlock(); 4553 AudioSystem::stopInput(mId); 4554 mLock.lock(); 4555 ALOGV("Record stopped OK"); 4556 } 4557 } 4558 } 4559} 4560 4561status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4562{ 4563 const size_t SIZE = 256; 4564 char buffer[SIZE]; 4565 String8 result; 4566 pid_t pid = 0; 4567 4568 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4569 result.append(buffer); 4570 4571 if (mActiveTrack != 0) { 4572 result.append("Active Track:\n"); 4573 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4574 mActiveTrack->dump(buffer, SIZE); 4575 result.append(buffer); 4576 4577 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4578 result.append(buffer); 4579 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4580 result.append(buffer); 4581 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4582 result.append(buffer); 4583 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4584 result.append(buffer); 4585 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4586 result.append(buffer); 4587 4588 4589 } else { 4590 result.append("No record client\n"); 4591 } 4592 write(fd, result.string(), result.size()); 4593 4594 dumpBase(fd, args); 4595 dumpEffectChains(fd, args); 4596 4597 return NO_ERROR; 4598} 4599 4600status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4601{ 4602 size_t framesReq = buffer->frameCount; 4603 size_t framesReady = mFrameCount - mRsmpInIndex; 4604 int channelCount; 4605 4606 if (framesReady == 0) { 4607 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4608 if (mBytesRead < 0) { 4609 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4610 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4611 // Force input into standby so that it tries to 4612 // recover at next read attempt 4613 mInput->stream->common.standby(&mInput->stream->common); 4614 usleep(kRecordThreadSleepUs); 4615 } 4616 buffer->raw = 0; 4617 buffer->frameCount = 0; 4618 return NOT_ENOUGH_DATA; 4619 } 4620 mRsmpInIndex = 0; 4621 framesReady = mFrameCount; 4622 } 4623 4624 if (framesReq > framesReady) { 4625 framesReq = framesReady; 4626 } 4627 4628 if (mChannelCount == 1 && mReqChannelCount == 2) { 4629 channelCount = 1; 4630 } else { 4631 channelCount = 2; 4632 } 4633 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4634 buffer->frameCount = framesReq; 4635 return NO_ERROR; 4636} 4637 4638void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4639{ 4640 mRsmpInIndex += buffer->frameCount; 4641 buffer->frameCount = 0; 4642} 4643 4644bool AudioFlinger::RecordThread::checkForNewParameters_l() 4645{ 4646 bool reconfig = false; 4647 4648 while (!mNewParameters.isEmpty()) { 4649 status_t status = NO_ERROR; 4650 String8 keyValuePair = mNewParameters[0]; 4651 AudioParameter param = AudioParameter(keyValuePair); 4652 int value; 4653 int reqFormat = mFormat; 4654 int reqSamplingRate = mReqSampleRate; 4655 int reqChannelCount = mReqChannelCount; 4656 4657 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4658 reqSamplingRate = value; 4659 reconfig = true; 4660 } 4661 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4662 reqFormat = value; 4663 reconfig = true; 4664 } 4665 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4666 reqChannelCount = popcount(value); 4667 reconfig = true; 4668 } 4669 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4670 // do not accept frame count changes if tracks are open as the track buffer 4671 // size depends on frame count and correct behavior would not be garantied 4672 // if frame count is changed after track creation 4673 if (mActiveTrack != 0) { 4674 status = INVALID_OPERATION; 4675 } else { 4676 reconfig = true; 4677 } 4678 } 4679 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4680 // forward device change to effects that have requested to be 4681 // aware of attached audio device. 4682 for (size_t i = 0; i < mEffectChains.size(); i++) { 4683 mEffectChains[i]->setDevice_l(value); 4684 } 4685 // store input device and output device but do not forward output device to audio HAL. 4686 // Note that status is ignored by the caller for output device 4687 // (see AudioFlinger::setParameters() 4688 if (value & AUDIO_DEVICE_OUT_ALL) { 4689 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4690 status = BAD_VALUE; 4691 } else { 4692 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4693 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4694 if (mTrack != NULL) { 4695 bool suspend = audio_is_bluetooth_sco_device( 4696 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4697 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4698 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4699 } 4700 } 4701 mDevice |= (uint32_t)value; 4702 } 4703 if (status == NO_ERROR) { 4704 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4705 if (status == INVALID_OPERATION) { 4706 mInput->stream->common.standby(&mInput->stream->common); 4707 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4708 } 4709 if (reconfig) { 4710 if (status == BAD_VALUE && 4711 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4712 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4713 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4714 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4715 (reqChannelCount < 3)) { 4716 status = NO_ERROR; 4717 } 4718 if (status == NO_ERROR) { 4719 readInputParameters(); 4720 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4721 } 4722 } 4723 } 4724 4725 mNewParameters.removeAt(0); 4726 4727 mParamStatus = status; 4728 mParamCond.signal(); 4729 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4730 // already timed out waiting for the status and will never signal the condition. 4731 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout); 4732 } 4733 return reconfig; 4734} 4735 4736String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4737{ 4738 char *s; 4739 String8 out_s8 = String8(); 4740 4741 Mutex::Autolock _l(mLock); 4742 if (initCheck() != NO_ERROR) { 4743 return out_s8; 4744 } 4745 4746 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4747 out_s8 = String8(s); 4748 free(s); 4749 return out_s8; 4750} 4751 4752void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4753 AudioSystem::OutputDescriptor desc; 4754 void *param2 = 0; 4755 4756 switch (event) { 4757 case AudioSystem::INPUT_OPENED: 4758 case AudioSystem::INPUT_CONFIG_CHANGED: 4759 desc.channels = mChannelMask; 4760 desc.samplingRate = mSampleRate; 4761 desc.format = mFormat; 4762 desc.frameCount = mFrameCount; 4763 desc.latency = 0; 4764 param2 = &desc; 4765 break; 4766 4767 case AudioSystem::INPUT_CLOSED: 4768 default: 4769 break; 4770 } 4771 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4772} 4773 4774void AudioFlinger::RecordThread::readInputParameters() 4775{ 4776 if (mRsmpInBuffer) delete mRsmpInBuffer; 4777 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4778 if (mResampler) delete mResampler; 4779 mResampler = 0; 4780 4781 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4782 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4783 mChannelCount = (uint16_t)popcount(mChannelMask); 4784 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4785 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); 4786 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4787 mFrameCount = mInputBytes / mFrameSize; 4788 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4789 4790 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4791 { 4792 int channelCount; 4793 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4794 // stereo to mono post process as the resampler always outputs stereo. 4795 if (mChannelCount == 1 && mReqChannelCount == 2) { 4796 channelCount = 1; 4797 } else { 4798 channelCount = 2; 4799 } 4800 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4801 mResampler->setSampleRate(mSampleRate); 4802 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4803 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4804 4805 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4806 if (mChannelCount == 1 && mReqChannelCount == 1) { 4807 mFrameCount >>= 1; 4808 } 4809 4810 } 4811 mRsmpInIndex = mFrameCount; 4812} 4813 4814unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4815{ 4816 Mutex::Autolock _l(mLock); 4817 if (initCheck() != NO_ERROR) { 4818 return 0; 4819 } 4820 4821 return mInput->stream->get_input_frames_lost(mInput->stream); 4822} 4823 4824uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4825{ 4826 Mutex::Autolock _l(mLock); 4827 uint32_t result = 0; 4828 if (getEffectChain_l(sessionId) != 0) { 4829 result = EFFECT_SESSION; 4830 } 4831 4832 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4833 result |= TRACK_SESSION; 4834 } 4835 4836 return result; 4837} 4838 4839AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4840{ 4841 Mutex::Autolock _l(mLock); 4842 return mTrack; 4843} 4844 4845AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() 4846{ 4847 Mutex::Autolock _l(mLock); 4848 return mInput; 4849} 4850 4851AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4852{ 4853 Mutex::Autolock _l(mLock); 4854 AudioStreamIn *input = mInput; 4855 mInput = NULL; 4856 return input; 4857} 4858 4859// this method must always be called either with ThreadBase mLock held or inside the thread loop 4860audio_stream_t* AudioFlinger::RecordThread::stream() 4861{ 4862 if (mInput == NULL) { 4863 return NULL; 4864 } 4865 return &mInput->stream->common; 4866} 4867 4868 4869// ---------------------------------------------------------------------------- 4870 4871int AudioFlinger::openOutput(uint32_t *pDevices, 4872 uint32_t *pSamplingRate, 4873 uint32_t *pFormat, 4874 uint32_t *pChannels, 4875 uint32_t *pLatencyMs, 4876 uint32_t flags) 4877{ 4878 status_t status; 4879 PlaybackThread *thread = NULL; 4880 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4881 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4882 uint32_t format = pFormat ? *pFormat : 0; 4883 uint32_t channels = pChannels ? *pChannels : 0; 4884 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4885 audio_stream_out_t *outStream; 4886 audio_hw_device_t *outHwDev; 4887 4888 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4889 pDevices ? *pDevices : 0, 4890 samplingRate, 4891 format, 4892 channels, 4893 flags); 4894 4895 if (pDevices == NULL || *pDevices == 0) { 4896 return 0; 4897 } 4898 4899 Mutex::Autolock _l(mLock); 4900 4901 outHwDev = findSuitableHwDev_l(*pDevices); 4902 if (outHwDev == NULL) 4903 return 0; 4904 4905 status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, 4906 &channels, &samplingRate, &outStream); 4907 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4908 outStream, 4909 samplingRate, 4910 format, 4911 channels, 4912 status); 4913 4914 mHardwareStatus = AUDIO_HW_IDLE; 4915 if (outStream != NULL) { 4916 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4917 int id = nextUniqueId(); 4918 4919 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4920 (format != AUDIO_FORMAT_PCM_16_BIT) || 4921 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4922 thread = new DirectOutputThread(this, output, id, *pDevices); 4923 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4924 } else { 4925 thread = new MixerThread(this, output, id, *pDevices); 4926 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4927 } 4928 mPlaybackThreads.add(id, thread); 4929 4930 if (pSamplingRate) *pSamplingRate = samplingRate; 4931 if (pFormat) *pFormat = format; 4932 if (pChannels) *pChannels = channels; 4933 if (pLatencyMs) *pLatencyMs = thread->latency(); 4934 4935 // notify client processes of the new output creation 4936 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4937 return id; 4938 } 4939 4940 return 0; 4941} 4942 4943int AudioFlinger::openDuplicateOutput(int output1, int output2) 4944{ 4945 Mutex::Autolock _l(mLock); 4946 MixerThread *thread1 = checkMixerThread_l(output1); 4947 MixerThread *thread2 = checkMixerThread_l(output2); 4948 4949 if (thread1 == NULL || thread2 == NULL) { 4950 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4951 return 0; 4952 } 4953 4954 int id = nextUniqueId(); 4955 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4956 thread->addOutputTrack(thread2); 4957 mPlaybackThreads.add(id, thread); 4958 // notify client processes of the new output creation 4959 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4960 return id; 4961} 4962 4963status_t AudioFlinger::closeOutput(int output) 4964{ 4965 // keep strong reference on the playback thread so that 4966 // it is not destroyed while exit() is executed 4967 sp <PlaybackThread> thread; 4968 { 4969 Mutex::Autolock _l(mLock); 4970 thread = checkPlaybackThread_l(output); 4971 if (thread == NULL) { 4972 return BAD_VALUE; 4973 } 4974 4975 ALOGV("closeOutput() %d", output); 4976 4977 if (thread->type() == ThreadBase::MIXER) { 4978 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4979 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4980 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4981 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4982 } 4983 } 4984 } 4985 void *param2 = 0; 4986 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4987 mPlaybackThreads.removeItem(output); 4988 } 4989 thread->exit(); 4990 4991 if (thread->type() != ThreadBase::DUPLICATING) { 4992 AudioStreamOut *out = thread->clearOutput(); 4993 // from now on thread->mOutput is NULL 4994 out->hwDev->close_output_stream(out->hwDev, out->stream); 4995 delete out; 4996 } 4997 return NO_ERROR; 4998} 4999 5000status_t AudioFlinger::suspendOutput(int output) 5001{ 5002 Mutex::Autolock _l(mLock); 5003 PlaybackThread *thread = checkPlaybackThread_l(output); 5004 5005 if (thread == NULL) { 5006 return BAD_VALUE; 5007 } 5008 5009 ALOGV("suspendOutput() %d", output); 5010 thread->suspend(); 5011 5012 return NO_ERROR; 5013} 5014 5015status_t AudioFlinger::restoreOutput(int output) 5016{ 5017 Mutex::Autolock _l(mLock); 5018 PlaybackThread *thread = checkPlaybackThread_l(output); 5019 5020 if (thread == NULL) { 5021 return BAD_VALUE; 5022 } 5023 5024 ALOGV("restoreOutput() %d", output); 5025 5026 thread->restore(); 5027 5028 return NO_ERROR; 5029} 5030 5031int AudioFlinger::openInput(uint32_t *pDevices, 5032 uint32_t *pSamplingRate, 5033 uint32_t *pFormat, 5034 uint32_t *pChannels, 5035 uint32_t acoustics) 5036{ 5037 status_t status; 5038 RecordThread *thread = NULL; 5039 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5040 uint32_t format = pFormat ? *pFormat : 0; 5041 uint32_t channels = pChannels ? *pChannels : 0; 5042 uint32_t reqSamplingRate = samplingRate; 5043 uint32_t reqFormat = format; 5044 uint32_t reqChannels = channels; 5045 audio_stream_in_t *inStream; 5046 audio_hw_device_t *inHwDev; 5047 5048 if (pDevices == NULL || *pDevices == 0) { 5049 return 0; 5050 } 5051 5052 Mutex::Autolock _l(mLock); 5053 5054 inHwDev = findSuitableHwDev_l(*pDevices); 5055 if (inHwDev == NULL) 5056 return 0; 5057 5058 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5059 &channels, &samplingRate, 5060 (audio_in_acoustics_t)acoustics, 5061 &inStream); 5062 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5063 inStream, 5064 samplingRate, 5065 format, 5066 channels, 5067 acoustics, 5068 status); 5069 5070 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5071 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5072 // or stereo to mono conversions on 16 bit PCM inputs. 5073 if (inStream == NULL && status == BAD_VALUE && 5074 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5075 (samplingRate <= 2 * reqSamplingRate) && 5076 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5077 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5078 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5079 &channels, &samplingRate, 5080 (audio_in_acoustics_t)acoustics, 5081 &inStream); 5082 } 5083 5084 if (inStream != NULL) { 5085 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5086 5087 int id = nextUniqueId(); 5088 // Start record thread 5089 // RecorThread require both input and output device indication to forward to audio 5090 // pre processing modules 5091 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5092 thread = new RecordThread(this, 5093 input, 5094 reqSamplingRate, 5095 reqChannels, 5096 id, 5097 device); 5098 mRecordThreads.add(id, thread); 5099 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5100 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5101 if (pFormat) *pFormat = format; 5102 if (pChannels) *pChannels = reqChannels; 5103 5104 input->stream->common.standby(&input->stream->common); 5105 5106 // notify client processes of the new input creation 5107 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5108 return id; 5109 } 5110 5111 return 0; 5112} 5113 5114status_t AudioFlinger::closeInput(int input) 5115{ 5116 // keep strong reference on the record thread so that 5117 // it is not destroyed while exit() is executed 5118 sp <RecordThread> thread; 5119 { 5120 Mutex::Autolock _l(mLock); 5121 thread = checkRecordThread_l(input); 5122 if (thread == NULL) { 5123 return BAD_VALUE; 5124 } 5125 5126 ALOGV("closeInput() %d", input); 5127 void *param2 = 0; 5128 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5129 mRecordThreads.removeItem(input); 5130 } 5131 thread->exit(); 5132 5133 AudioStreamIn *in = thread->clearInput(); 5134 // from now on thread->mInput is NULL 5135 in->hwDev->close_input_stream(in->hwDev, in->stream); 5136 delete in; 5137 5138 return NO_ERROR; 5139} 5140 5141status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 5142{ 5143 Mutex::Autolock _l(mLock); 5144 MixerThread *dstThread = checkMixerThread_l(output); 5145 if (dstThread == NULL) { 5146 LOGW("setStreamOutput() bad output id %d", output); 5147 return BAD_VALUE; 5148 } 5149 5150 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5151 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5152 5153 dstThread->setStreamValid(stream, true); 5154 5155 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5156 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5157 if (thread != dstThread && 5158 thread->type() != ThreadBase::DIRECT) { 5159 MixerThread *srcThread = (MixerThread *)thread; 5160 srcThread->setStreamValid(stream, false); 5161 srcThread->invalidateTracks(stream); 5162 } 5163 } 5164 5165 return NO_ERROR; 5166} 5167 5168 5169int AudioFlinger::newAudioSessionId() 5170{ 5171 return nextUniqueId(); 5172} 5173 5174void AudioFlinger::acquireAudioSessionId(int audioSession) 5175{ 5176 Mutex::Autolock _l(mLock); 5177 int caller = IPCThreadState::self()->getCallingPid(); 5178 ALOGV("acquiring %d from %d", audioSession, caller); 5179 int num = mAudioSessionRefs.size(); 5180 for (int i = 0; i< num; i++) { 5181 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5182 if (ref->sessionid == audioSession && ref->pid == caller) { 5183 ref->cnt++; 5184 ALOGV(" incremented refcount to %d", ref->cnt); 5185 return; 5186 } 5187 } 5188 AudioSessionRef *ref = new AudioSessionRef(); 5189 ref->sessionid = audioSession; 5190 ref->pid = caller; 5191 ref->cnt = 1; 5192 mAudioSessionRefs.push(ref); 5193 ALOGV(" added new entry for %d", ref->sessionid); 5194} 5195 5196void AudioFlinger::releaseAudioSessionId(int audioSession) 5197{ 5198 Mutex::Autolock _l(mLock); 5199 int caller = IPCThreadState::self()->getCallingPid(); 5200 ALOGV("releasing %d from %d", audioSession, caller); 5201 int num = mAudioSessionRefs.size(); 5202 for (int i = 0; i< num; i++) { 5203 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5204 if (ref->sessionid == audioSession && ref->pid == caller) { 5205 ref->cnt--; 5206 ALOGV(" decremented refcount to %d", ref->cnt); 5207 if (ref->cnt == 0) { 5208 mAudioSessionRefs.removeAt(i); 5209 delete ref; 5210 purgeStaleEffects_l(); 5211 } 5212 return; 5213 } 5214 } 5215 LOGW("session id %d not found for pid %d", audioSession, caller); 5216} 5217 5218void AudioFlinger::purgeStaleEffects_l() { 5219 5220 ALOGV("purging stale effects"); 5221 5222 Vector< sp<EffectChain> > chains; 5223 5224 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5225 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5226 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5227 sp<EffectChain> ec = t->mEffectChains[j]; 5228 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5229 chains.push(ec); 5230 } 5231 } 5232 } 5233 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5234 sp<RecordThread> t = mRecordThreads.valueAt(i); 5235 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5236 sp<EffectChain> ec = t->mEffectChains[j]; 5237 chains.push(ec); 5238 } 5239 } 5240 5241 for (size_t i = 0; i < chains.size(); i++) { 5242 sp<EffectChain> ec = chains[i]; 5243 int sessionid = ec->sessionId(); 5244 sp<ThreadBase> t = ec->mThread.promote(); 5245 if (t == 0) { 5246 continue; 5247 } 5248 size_t numsessionrefs = mAudioSessionRefs.size(); 5249 bool found = false; 5250 for (size_t k = 0; k < numsessionrefs; k++) { 5251 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5252 if (ref->sessionid == sessionid) { 5253 ALOGV(" session %d still exists for %d with %d refs", 5254 sessionid, ref->pid, ref->cnt); 5255 found = true; 5256 break; 5257 } 5258 } 5259 if (!found) { 5260 // remove all effects from the chain 5261 while (ec->mEffects.size()) { 5262 sp<EffectModule> effect = ec->mEffects[0]; 5263 effect->unPin(); 5264 Mutex::Autolock _l (t->mLock); 5265 t->removeEffect_l(effect); 5266 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5267 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5268 if (handle != 0) { 5269 handle->mEffect.clear(); 5270 if (handle->mHasControl && handle->mEnabled) { 5271 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5272 } 5273 } 5274 } 5275 AudioSystem::unregisterEffect(effect->id()); 5276 } 5277 } 5278 } 5279 return; 5280} 5281 5282// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5283AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5284{ 5285 PlaybackThread *thread = NULL; 5286 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5287 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5288 } 5289 return thread; 5290} 5291 5292// checkMixerThread_l() must be called with AudioFlinger::mLock held 5293AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5294{ 5295 PlaybackThread *thread = checkPlaybackThread_l(output); 5296 if (thread != NULL) { 5297 if (thread->type() == ThreadBase::DIRECT) { 5298 thread = NULL; 5299 } 5300 } 5301 return (MixerThread *)thread; 5302} 5303 5304// checkRecordThread_l() must be called with AudioFlinger::mLock held 5305AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5306{ 5307 RecordThread *thread = NULL; 5308 if (mRecordThreads.indexOfKey(input) >= 0) { 5309 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5310 } 5311 return thread; 5312} 5313 5314uint32_t AudioFlinger::nextUniqueId() 5315{ 5316 return android_atomic_inc(&mNextUniqueId); 5317} 5318 5319AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5320{ 5321 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5322 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5323 AudioStreamOut *output = thread->getOutput(); 5324 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5325 return thread; 5326 } 5327 } 5328 return NULL; 5329} 5330 5331uint32_t AudioFlinger::primaryOutputDevice_l() 5332{ 5333 PlaybackThread *thread = primaryPlaybackThread_l(); 5334 5335 if (thread == NULL) { 5336 return 0; 5337 } 5338 5339 return thread->device(); 5340} 5341 5342 5343// ---------------------------------------------------------------------------- 5344// Effect management 5345// ---------------------------------------------------------------------------- 5346 5347 5348status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5349{ 5350 Mutex::Autolock _l(mLock); 5351 return EffectQueryNumberEffects(numEffects); 5352} 5353 5354status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5355{ 5356 Mutex::Autolock _l(mLock); 5357 return EffectQueryEffect(index, descriptor); 5358} 5359 5360status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5361{ 5362 Mutex::Autolock _l(mLock); 5363 return EffectGetDescriptor(pUuid, descriptor); 5364} 5365 5366 5367sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5368 effect_descriptor_t *pDesc, 5369 const sp<IEffectClient>& effectClient, 5370 int32_t priority, 5371 int io, 5372 int sessionId, 5373 status_t *status, 5374 int *id, 5375 int *enabled) 5376{ 5377 status_t lStatus = NO_ERROR; 5378 sp<EffectHandle> handle; 5379 effect_descriptor_t desc; 5380 sp<Client> client; 5381 wp<Client> wclient; 5382 5383 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5384 pid, effectClient.get(), priority, sessionId, io); 5385 5386 if (pDesc == NULL) { 5387 lStatus = BAD_VALUE; 5388 goto Exit; 5389 } 5390 5391 // check audio settings permission for global effects 5392 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5393 lStatus = PERMISSION_DENIED; 5394 goto Exit; 5395 } 5396 5397 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5398 // that can only be created by audio policy manager (running in same process) 5399 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5400 lStatus = PERMISSION_DENIED; 5401 goto Exit; 5402 } 5403 5404 if (io == 0) { 5405 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5406 // output must be specified by AudioPolicyManager when using session 5407 // AUDIO_SESSION_OUTPUT_STAGE 5408 lStatus = BAD_VALUE; 5409 goto Exit; 5410 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5411 // if the output returned by getOutputForEffect() is removed before we lock the 5412 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5413 // and we will exit safely 5414 io = AudioSystem::getOutputForEffect(&desc); 5415 } 5416 } 5417 5418 { 5419 Mutex::Autolock _l(mLock); 5420 5421 5422 if (!EffectIsNullUuid(&pDesc->uuid)) { 5423 // if uuid is specified, request effect descriptor 5424 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5425 if (lStatus < 0) { 5426 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5427 goto Exit; 5428 } 5429 } else { 5430 // if uuid is not specified, look for an available implementation 5431 // of the required type in effect factory 5432 if (EffectIsNullUuid(&pDesc->type)) { 5433 LOGW("createEffect() no effect type"); 5434 lStatus = BAD_VALUE; 5435 goto Exit; 5436 } 5437 uint32_t numEffects = 0; 5438 effect_descriptor_t d; 5439 d.flags = 0; // prevent compiler warning 5440 bool found = false; 5441 5442 lStatus = EffectQueryNumberEffects(&numEffects); 5443 if (lStatus < 0) { 5444 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5445 goto Exit; 5446 } 5447 for (uint32_t i = 0; i < numEffects; i++) { 5448 lStatus = EffectQueryEffect(i, &desc); 5449 if (lStatus < 0) { 5450 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5451 continue; 5452 } 5453 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5454 // If matching type found save effect descriptor. If the session is 5455 // 0 and the effect is not auxiliary, continue enumeration in case 5456 // an auxiliary version of this effect type is available 5457 found = true; 5458 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5459 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5460 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5461 break; 5462 } 5463 } 5464 } 5465 if (!found) { 5466 lStatus = BAD_VALUE; 5467 LOGW("createEffect() effect not found"); 5468 goto Exit; 5469 } 5470 // For same effect type, chose auxiliary version over insert version if 5471 // connect to output mix (Compliance to OpenSL ES) 5472 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5473 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5474 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5475 } 5476 } 5477 5478 // Do not allow auxiliary effects on a session different from 0 (output mix) 5479 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5480 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5481 lStatus = INVALID_OPERATION; 5482 goto Exit; 5483 } 5484 5485 // check recording permission for visualizer 5486 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5487 !recordingAllowed()) { 5488 lStatus = PERMISSION_DENIED; 5489 goto Exit; 5490 } 5491 5492 // return effect descriptor 5493 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5494 5495 // If output is not specified try to find a matching audio session ID in one of the 5496 // output threads. 5497 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5498 // because of code checking output when entering the function. 5499 // Note: io is never 0 when creating an effect on an input 5500 if (io == 0) { 5501 // look for the thread where the specified audio session is present 5502 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5503 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5504 io = mPlaybackThreads.keyAt(i); 5505 break; 5506 } 5507 } 5508 if (io == 0) { 5509 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5510 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5511 io = mRecordThreads.keyAt(i); 5512 break; 5513 } 5514 } 5515 } 5516 // If no output thread contains the requested session ID, default to 5517 // first output. The effect chain will be moved to the correct output 5518 // thread when a track with the same session ID is created 5519 if (io == 0 && mPlaybackThreads.size()) { 5520 io = mPlaybackThreads.keyAt(0); 5521 } 5522 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5523 } 5524 ThreadBase *thread = checkRecordThread_l(io); 5525 if (thread == NULL) { 5526 thread = checkPlaybackThread_l(io); 5527 if (thread == NULL) { 5528 LOGE("createEffect() unknown output thread"); 5529 lStatus = BAD_VALUE; 5530 goto Exit; 5531 } 5532 } 5533 5534 wclient = mClients.valueFor(pid); 5535 5536 if (wclient != NULL) { 5537 client = wclient.promote(); 5538 } else { 5539 client = new Client(this, pid); 5540 mClients.add(pid, client); 5541 } 5542 5543 // create effect on selected output thread 5544 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5545 &desc, enabled, &lStatus); 5546 if (handle != 0 && id != NULL) { 5547 *id = handle->id(); 5548 } 5549 } 5550 5551Exit: 5552 if(status) { 5553 *status = lStatus; 5554 } 5555 return handle; 5556} 5557 5558status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5559{ 5560 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5561 sessionId, srcOutput, dstOutput); 5562 Mutex::Autolock _l(mLock); 5563 if (srcOutput == dstOutput) { 5564 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 5565 return NO_ERROR; 5566 } 5567 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5568 if (srcThread == NULL) { 5569 LOGW("moveEffects() bad srcOutput %d", srcOutput); 5570 return BAD_VALUE; 5571 } 5572 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5573 if (dstThread == NULL) { 5574 LOGW("moveEffects() bad dstOutput %d", dstOutput); 5575 return BAD_VALUE; 5576 } 5577 5578 Mutex::Autolock _dl(dstThread->mLock); 5579 Mutex::Autolock _sl(srcThread->mLock); 5580 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5581 5582 return NO_ERROR; 5583} 5584 5585// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5586status_t AudioFlinger::moveEffectChain_l(int sessionId, 5587 AudioFlinger::PlaybackThread *srcThread, 5588 AudioFlinger::PlaybackThread *dstThread, 5589 bool reRegister) 5590{ 5591 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5592 sessionId, srcThread, dstThread); 5593 5594 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5595 if (chain == 0) { 5596 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5597 sessionId, srcThread); 5598 return INVALID_OPERATION; 5599 } 5600 5601 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5602 // so that a new chain is created with correct parameters when first effect is added. This is 5603 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5604 // removed. 5605 srcThread->removeEffectChain_l(chain); 5606 5607 // transfer all effects one by one so that new effect chain is created on new thread with 5608 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5609 int dstOutput = dstThread->id(); 5610 sp<EffectChain> dstChain; 5611 uint32_t strategy = 0; // prevent compiler warning 5612 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5613 while (effect != 0) { 5614 srcThread->removeEffect_l(effect); 5615 dstThread->addEffect_l(effect); 5616 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5617 if (effect->state() == EffectModule::ACTIVE || 5618 effect->state() == EffectModule::STOPPING) { 5619 effect->start(); 5620 } 5621 // if the move request is not received from audio policy manager, the effect must be 5622 // re-registered with the new strategy and output 5623 if (dstChain == 0) { 5624 dstChain = effect->chain().promote(); 5625 if (dstChain == 0) { 5626 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5627 srcThread->addEffect_l(effect); 5628 return NO_INIT; 5629 } 5630 strategy = dstChain->strategy(); 5631 } 5632 if (reRegister) { 5633 AudioSystem::unregisterEffect(effect->id()); 5634 AudioSystem::registerEffect(&effect->desc(), 5635 dstOutput, 5636 strategy, 5637 sessionId, 5638 effect->id()); 5639 } 5640 effect = chain->getEffectFromId_l(0); 5641 } 5642 5643 return NO_ERROR; 5644} 5645 5646 5647// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5648sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5649 const sp<AudioFlinger::Client>& client, 5650 const sp<IEffectClient>& effectClient, 5651 int32_t priority, 5652 int sessionId, 5653 effect_descriptor_t *desc, 5654 int *enabled, 5655 status_t *status 5656 ) 5657{ 5658 sp<EffectModule> effect; 5659 sp<EffectHandle> handle; 5660 status_t lStatus; 5661 sp<EffectChain> chain; 5662 bool chainCreated = false; 5663 bool effectCreated = false; 5664 bool effectRegistered = false; 5665 5666 lStatus = initCheck(); 5667 if (lStatus != NO_ERROR) { 5668 LOGW("createEffect_l() Audio driver not initialized."); 5669 goto Exit; 5670 } 5671 5672 // Do not allow effects with session ID 0 on direct output or duplicating threads 5673 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5674 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5675 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5676 desc->name, sessionId); 5677 lStatus = BAD_VALUE; 5678 goto Exit; 5679 } 5680 // Only Pre processor effects are allowed on input threads and only on input threads 5681 if ((mType == RECORD && 5682 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5683 (mType != RECORD && 5684 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5685 LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5686 desc->name, desc->flags, mType); 5687 lStatus = BAD_VALUE; 5688 goto Exit; 5689 } 5690 5691 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5692 5693 { // scope for mLock 5694 Mutex::Autolock _l(mLock); 5695 5696 // check for existing effect chain with the requested audio session 5697 chain = getEffectChain_l(sessionId); 5698 if (chain == 0) { 5699 // create a new chain for this session 5700 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5701 chain = new EffectChain(this, sessionId); 5702 addEffectChain_l(chain); 5703 chain->setStrategy(getStrategyForSession_l(sessionId)); 5704 chainCreated = true; 5705 } else { 5706 effect = chain->getEffectFromDesc_l(desc); 5707 } 5708 5709 ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 5710 5711 if (effect == 0) { 5712 int id = mAudioFlinger->nextUniqueId(); 5713 // Check CPU and memory usage 5714 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5715 if (lStatus != NO_ERROR) { 5716 goto Exit; 5717 } 5718 effectRegistered = true; 5719 // create a new effect module if none present in the chain 5720 effect = new EffectModule(this, chain, desc, id, sessionId); 5721 lStatus = effect->status(); 5722 if (lStatus != NO_ERROR) { 5723 goto Exit; 5724 } 5725 lStatus = chain->addEffect_l(effect); 5726 if (lStatus != NO_ERROR) { 5727 goto Exit; 5728 } 5729 effectCreated = true; 5730 5731 effect->setDevice(mDevice); 5732 effect->setMode(mAudioFlinger->getMode()); 5733 } 5734 // create effect handle and connect it to effect module 5735 handle = new EffectHandle(effect, client, effectClient, priority); 5736 lStatus = effect->addHandle(handle); 5737 if (enabled) { 5738 *enabled = (int)effect->isEnabled(); 5739 } 5740 } 5741 5742Exit: 5743 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5744 Mutex::Autolock _l(mLock); 5745 if (effectCreated) { 5746 chain->removeEffect_l(effect); 5747 } 5748 if (effectRegistered) { 5749 AudioSystem::unregisterEffect(effect->id()); 5750 } 5751 if (chainCreated) { 5752 removeEffectChain_l(chain); 5753 } 5754 handle.clear(); 5755 } 5756 5757 if(status) { 5758 *status = lStatus; 5759 } 5760 return handle; 5761} 5762 5763sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5764{ 5765 sp<EffectModule> effect; 5766 5767 sp<EffectChain> chain = getEffectChain_l(sessionId); 5768 if (chain != 0) { 5769 effect = chain->getEffectFromId_l(effectId); 5770 } 5771 return effect; 5772} 5773 5774// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5775// PlaybackThread::mLock held 5776status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5777{ 5778 // check for existing effect chain with the requested audio session 5779 int sessionId = effect->sessionId(); 5780 sp<EffectChain> chain = getEffectChain_l(sessionId); 5781 bool chainCreated = false; 5782 5783 if (chain == 0) { 5784 // create a new chain for this session 5785 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5786 chain = new EffectChain(this, sessionId); 5787 addEffectChain_l(chain); 5788 chain->setStrategy(getStrategyForSession_l(sessionId)); 5789 chainCreated = true; 5790 } 5791 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5792 5793 if (chain->getEffectFromId_l(effect->id()) != 0) { 5794 LOGW("addEffect_l() %p effect %s already present in chain %p", 5795 this, effect->desc().name, chain.get()); 5796 return BAD_VALUE; 5797 } 5798 5799 status_t status = chain->addEffect_l(effect); 5800 if (status != NO_ERROR) { 5801 if (chainCreated) { 5802 removeEffectChain_l(chain); 5803 } 5804 return status; 5805 } 5806 5807 effect->setDevice(mDevice); 5808 effect->setMode(mAudioFlinger->getMode()); 5809 return NO_ERROR; 5810} 5811 5812void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5813 5814 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5815 effect_descriptor_t desc = effect->desc(); 5816 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5817 detachAuxEffect_l(effect->id()); 5818 } 5819 5820 sp<EffectChain> chain = effect->chain().promote(); 5821 if (chain != 0) { 5822 // remove effect chain if removing last effect 5823 if (chain->removeEffect_l(effect) == 0) { 5824 removeEffectChain_l(chain); 5825 } 5826 } else { 5827 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5828 } 5829} 5830 5831void AudioFlinger::ThreadBase::lockEffectChains_l( 5832 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5833{ 5834 effectChains = mEffectChains; 5835 for (size_t i = 0; i < mEffectChains.size(); i++) { 5836 mEffectChains[i]->lock(); 5837 } 5838} 5839 5840void AudioFlinger::ThreadBase::unlockEffectChains( 5841 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5842{ 5843 for (size_t i = 0; i < effectChains.size(); i++) { 5844 effectChains[i]->unlock(); 5845 } 5846} 5847 5848sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5849{ 5850 Mutex::Autolock _l(mLock); 5851 return getEffectChain_l(sessionId); 5852} 5853 5854sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5855{ 5856 sp<EffectChain> chain; 5857 5858 size_t size = mEffectChains.size(); 5859 for (size_t i = 0; i < size; i++) { 5860 if (mEffectChains[i]->sessionId() == sessionId) { 5861 chain = mEffectChains[i]; 5862 break; 5863 } 5864 } 5865 return chain; 5866} 5867 5868void AudioFlinger::ThreadBase::setMode(uint32_t mode) 5869{ 5870 Mutex::Autolock _l(mLock); 5871 size_t size = mEffectChains.size(); 5872 for (size_t i = 0; i < size; i++) { 5873 mEffectChains[i]->setMode_l(mode); 5874 } 5875} 5876 5877void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5878 const wp<EffectHandle>& handle, 5879 bool unpiniflast) { 5880 5881 Mutex::Autolock _l(mLock); 5882 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5883 // delete the effect module if removing last handle on it 5884 if (effect->removeHandle(handle) == 0) { 5885 if (!effect->isPinned() || unpiniflast) { 5886 removeEffect_l(effect); 5887 AudioSystem::unregisterEffect(effect->id()); 5888 } 5889 } 5890} 5891 5892status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5893{ 5894 int session = chain->sessionId(); 5895 int16_t *buffer = mMixBuffer; 5896 bool ownsBuffer = false; 5897 5898 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5899 if (session > 0) { 5900 // Only one effect chain can be present in direct output thread and it uses 5901 // the mix buffer as input 5902 if (mType != DIRECT) { 5903 size_t numSamples = mFrameCount * mChannelCount; 5904 buffer = new int16_t[numSamples]; 5905 memset(buffer, 0, numSamples * sizeof(int16_t)); 5906 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5907 ownsBuffer = true; 5908 } 5909 5910 // Attach all tracks with same session ID to this chain. 5911 for (size_t i = 0; i < mTracks.size(); ++i) { 5912 sp<Track> track = mTracks[i]; 5913 if (session == track->sessionId()) { 5914 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5915 track->setMainBuffer(buffer); 5916 chain->incTrackCnt(); 5917 } 5918 } 5919 5920 // indicate all active tracks in the chain 5921 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5922 sp<Track> track = mActiveTracks[i].promote(); 5923 if (track == 0) continue; 5924 if (session == track->sessionId()) { 5925 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5926 chain->incActiveTrackCnt(); 5927 } 5928 } 5929 } 5930 5931 chain->setInBuffer(buffer, ownsBuffer); 5932 chain->setOutBuffer(mMixBuffer); 5933 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5934 // chains list in order to be processed last as it contains output stage effects 5935 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5936 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5937 // after track specific effects and before output stage 5938 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5939 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5940 // Effect chain for other sessions are inserted at beginning of effect 5941 // chains list to be processed before output mix effects. Relative order between other 5942 // sessions is not important 5943 size_t size = mEffectChains.size(); 5944 size_t i = 0; 5945 for (i = 0; i < size; i++) { 5946 if (mEffectChains[i]->sessionId() < session) break; 5947 } 5948 mEffectChains.insertAt(chain, i); 5949 checkSuspendOnAddEffectChain_l(chain); 5950 5951 return NO_ERROR; 5952} 5953 5954size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5955{ 5956 int session = chain->sessionId(); 5957 5958 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5959 5960 for (size_t i = 0; i < mEffectChains.size(); i++) { 5961 if (chain == mEffectChains[i]) { 5962 mEffectChains.removeAt(i); 5963 // detach all active tracks from the chain 5964 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5965 sp<Track> track = mActiveTracks[i].promote(); 5966 if (track == 0) continue; 5967 if (session == track->sessionId()) { 5968 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5969 chain.get(), session); 5970 chain->decActiveTrackCnt(); 5971 } 5972 } 5973 5974 // detach all tracks with same session ID from this chain 5975 for (size_t i = 0; i < mTracks.size(); ++i) { 5976 sp<Track> track = mTracks[i]; 5977 if (session == track->sessionId()) { 5978 track->setMainBuffer(mMixBuffer); 5979 chain->decTrackCnt(); 5980 } 5981 } 5982 break; 5983 } 5984 } 5985 return mEffectChains.size(); 5986} 5987 5988status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5989 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5990{ 5991 Mutex::Autolock _l(mLock); 5992 return attachAuxEffect_l(track, EffectId); 5993} 5994 5995status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5996 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5997{ 5998 status_t status = NO_ERROR; 5999 6000 if (EffectId == 0) { 6001 track->setAuxBuffer(0, NULL); 6002 } else { 6003 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6004 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6005 if (effect != 0) { 6006 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6007 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6008 } else { 6009 status = INVALID_OPERATION; 6010 } 6011 } else { 6012 status = BAD_VALUE; 6013 } 6014 } 6015 return status; 6016} 6017 6018void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6019{ 6020 for (size_t i = 0; i < mTracks.size(); ++i) { 6021 sp<Track> track = mTracks[i]; 6022 if (track->auxEffectId() == effectId) { 6023 attachAuxEffect_l(track, 0); 6024 } 6025 } 6026} 6027 6028status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6029{ 6030 // only one chain per input thread 6031 if (mEffectChains.size() != 0) { 6032 return INVALID_OPERATION; 6033 } 6034 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6035 6036 chain->setInBuffer(NULL); 6037 chain->setOutBuffer(NULL); 6038 6039 checkSuspendOnAddEffectChain_l(chain); 6040 6041 mEffectChains.add(chain); 6042 6043 return NO_ERROR; 6044} 6045 6046size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6047{ 6048 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6049 LOGW_IF(mEffectChains.size() != 1, 6050 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6051 chain.get(), mEffectChains.size(), this); 6052 if (mEffectChains.size() == 1) { 6053 mEffectChains.removeAt(0); 6054 } 6055 return 0; 6056} 6057 6058// ---------------------------------------------------------------------------- 6059// EffectModule implementation 6060// ---------------------------------------------------------------------------- 6061 6062#undef LOG_TAG 6063#define LOG_TAG "AudioFlinger::EffectModule" 6064 6065AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6066 const wp<AudioFlinger::EffectChain>& chain, 6067 effect_descriptor_t *desc, 6068 int id, 6069 int sessionId) 6070 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6071 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6072{ 6073 ALOGV("Constructor %p", this); 6074 int lStatus; 6075 sp<ThreadBase> thread = mThread.promote(); 6076 if (thread == 0) { 6077 return; 6078 } 6079 6080 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6081 6082 // create effect engine from effect factory 6083 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6084 6085 if (mStatus != NO_ERROR) { 6086 return; 6087 } 6088 lStatus = init(); 6089 if (lStatus < 0) { 6090 mStatus = lStatus; 6091 goto Error; 6092 } 6093 6094 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6095 mPinned = true; 6096 } 6097 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6098 return; 6099Error: 6100 EffectRelease(mEffectInterface); 6101 mEffectInterface = NULL; 6102 ALOGV("Constructor Error %d", mStatus); 6103} 6104 6105AudioFlinger::EffectModule::~EffectModule() 6106{ 6107 ALOGV("Destructor %p", this); 6108 if (mEffectInterface != NULL) { 6109 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6110 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6111 sp<ThreadBase> thread = mThread.promote(); 6112 if (thread != 0) { 6113 audio_stream_t *stream = thread->stream(); 6114 if (stream != NULL) { 6115 stream->remove_audio_effect(stream, mEffectInterface); 6116 } 6117 } 6118 } 6119 // release effect engine 6120 EffectRelease(mEffectInterface); 6121 } 6122} 6123 6124status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6125{ 6126 status_t status; 6127 6128 Mutex::Autolock _l(mLock); 6129 // First handle in mHandles has highest priority and controls the effect module 6130 int priority = handle->priority(); 6131 size_t size = mHandles.size(); 6132 sp<EffectHandle> h; 6133 size_t i; 6134 for (i = 0; i < size; i++) { 6135 h = mHandles[i].promote(); 6136 if (h == 0) continue; 6137 if (h->priority() <= priority) break; 6138 } 6139 // if inserted in first place, move effect control from previous owner to this handle 6140 if (i == 0) { 6141 bool enabled = false; 6142 if (h != 0) { 6143 enabled = h->enabled(); 6144 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6145 } 6146 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6147 status = NO_ERROR; 6148 } else { 6149 status = ALREADY_EXISTS; 6150 } 6151 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6152 mHandles.insertAt(handle, i); 6153 return status; 6154} 6155 6156size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6157{ 6158 Mutex::Autolock _l(mLock); 6159 size_t size = mHandles.size(); 6160 size_t i; 6161 for (i = 0; i < size; i++) { 6162 if (mHandles[i] == handle) break; 6163 } 6164 if (i == size) { 6165 return size; 6166 } 6167 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6168 6169 bool enabled = false; 6170 EffectHandle *hdl = handle.unsafe_get(); 6171 if (hdl) { 6172 ALOGV("removeHandle() unsafe_get OK"); 6173 enabled = hdl->enabled(); 6174 } 6175 mHandles.removeAt(i); 6176 size = mHandles.size(); 6177 // if removed from first place, move effect control from this handle to next in line 6178 if (i == 0 && size != 0) { 6179 sp<EffectHandle> h = mHandles[0].promote(); 6180 if (h != 0) { 6181 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6182 } 6183 } 6184 6185 // Prevent calls to process() and other functions on effect interface from now on. 6186 // The effect engine will be released by the destructor when the last strong reference on 6187 // this object is released which can happen after next process is called. 6188 if (size == 0 && !mPinned) { 6189 mState = DESTROYED; 6190 } 6191 6192 return size; 6193} 6194 6195sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6196{ 6197 Mutex::Autolock _l(mLock); 6198 sp<EffectHandle> handle; 6199 if (mHandles.size() != 0) { 6200 handle = mHandles[0].promote(); 6201 } 6202 return handle; 6203} 6204 6205void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6206{ 6207 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6208 // keep a strong reference on this EffectModule to avoid calling the 6209 // destructor before we exit 6210 sp<EffectModule> keep(this); 6211 { 6212 sp<ThreadBase> thread = mThread.promote(); 6213 if (thread != 0) { 6214 thread->disconnectEffect(keep, handle, unpiniflast); 6215 } 6216 } 6217} 6218 6219void AudioFlinger::EffectModule::updateState() { 6220 Mutex::Autolock _l(mLock); 6221 6222 switch (mState) { 6223 case RESTART: 6224 reset_l(); 6225 // FALL THROUGH 6226 6227 case STARTING: 6228 // clear auxiliary effect input buffer for next accumulation 6229 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6230 memset(mConfig.inputCfg.buffer.raw, 6231 0, 6232 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6233 } 6234 start_l(); 6235 mState = ACTIVE; 6236 break; 6237 case STOPPING: 6238 stop_l(); 6239 mDisableWaitCnt = mMaxDisableWaitCnt; 6240 mState = STOPPED; 6241 break; 6242 case STOPPED: 6243 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6244 // turn off sequence. 6245 if (--mDisableWaitCnt == 0) { 6246 reset_l(); 6247 mState = IDLE; 6248 } 6249 break; 6250 default: //IDLE , ACTIVE, DESTROYED 6251 break; 6252 } 6253} 6254 6255void AudioFlinger::EffectModule::process() 6256{ 6257 Mutex::Autolock _l(mLock); 6258 6259 if (mState == DESTROYED || mEffectInterface == NULL || 6260 mConfig.inputCfg.buffer.raw == NULL || 6261 mConfig.outputCfg.buffer.raw == NULL) { 6262 return; 6263 } 6264 6265 if (isProcessEnabled()) { 6266 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6267 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6268 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6269 mConfig.inputCfg.buffer.s32, 6270 mConfig.inputCfg.buffer.frameCount/2); 6271 } 6272 6273 // do the actual processing in the effect engine 6274 int ret = (*mEffectInterface)->process(mEffectInterface, 6275 &mConfig.inputCfg.buffer, 6276 &mConfig.outputCfg.buffer); 6277 6278 // force transition to IDLE state when engine is ready 6279 if (mState == STOPPED && ret == -ENODATA) { 6280 mDisableWaitCnt = 1; 6281 } 6282 6283 // clear auxiliary effect input buffer for next accumulation 6284 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6285 memset(mConfig.inputCfg.buffer.raw, 0, 6286 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6287 } 6288 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6289 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6290 // If an insert effect is idle and input buffer is different from output buffer, 6291 // accumulate input onto output 6292 sp<EffectChain> chain = mChain.promote(); 6293 if (chain != 0 && chain->activeTrackCnt() != 0) { 6294 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6295 int16_t *in = mConfig.inputCfg.buffer.s16; 6296 int16_t *out = mConfig.outputCfg.buffer.s16; 6297 for (size_t i = 0; i < frameCnt; i++) { 6298 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6299 } 6300 } 6301 } 6302} 6303 6304void AudioFlinger::EffectModule::reset_l() 6305{ 6306 if (mEffectInterface == NULL) { 6307 return; 6308 } 6309 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6310} 6311 6312status_t AudioFlinger::EffectModule::configure() 6313{ 6314 uint32_t channels; 6315 if (mEffectInterface == NULL) { 6316 return NO_INIT; 6317 } 6318 6319 sp<ThreadBase> thread = mThread.promote(); 6320 if (thread == 0) { 6321 return DEAD_OBJECT; 6322 } 6323 6324 // TODO: handle configuration of effects replacing track process 6325 if (thread->channelCount() == 1) { 6326 channels = AUDIO_CHANNEL_OUT_MONO; 6327 } else { 6328 channels = AUDIO_CHANNEL_OUT_STEREO; 6329 } 6330 6331 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6332 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6333 } else { 6334 mConfig.inputCfg.channels = channels; 6335 } 6336 mConfig.outputCfg.channels = channels; 6337 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6338 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6339 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6340 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6341 mConfig.inputCfg.bufferProvider.cookie = NULL; 6342 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6343 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6344 mConfig.outputCfg.bufferProvider.cookie = NULL; 6345 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6346 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6347 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6348 // Insert effect: 6349 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6350 // always overwrites output buffer: input buffer == output buffer 6351 // - in other sessions: 6352 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6353 // other effect: overwrites output buffer: input buffer == output buffer 6354 // Auxiliary effect: 6355 // accumulates in output buffer: input buffer != output buffer 6356 // Therefore: accumulate <=> input buffer != output buffer 6357 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6358 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6359 } else { 6360 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6361 } 6362 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6363 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6364 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6365 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6366 6367 ALOGV("configure() %p thread %p buffer %p framecount %d", 6368 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6369 6370 status_t cmdStatus; 6371 uint32_t size = sizeof(int); 6372 status_t status = (*mEffectInterface)->command(mEffectInterface, 6373 EFFECT_CMD_CONFIGURE, 6374 sizeof(effect_config_t), 6375 &mConfig, 6376 &size, 6377 &cmdStatus); 6378 if (status == 0) { 6379 status = cmdStatus; 6380 } 6381 6382 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6383 (1000 * mConfig.outputCfg.buffer.frameCount); 6384 6385 return status; 6386} 6387 6388status_t AudioFlinger::EffectModule::init() 6389{ 6390 Mutex::Autolock _l(mLock); 6391 if (mEffectInterface == NULL) { 6392 return NO_INIT; 6393 } 6394 status_t cmdStatus; 6395 uint32_t size = sizeof(status_t); 6396 status_t status = (*mEffectInterface)->command(mEffectInterface, 6397 EFFECT_CMD_INIT, 6398 0, 6399 NULL, 6400 &size, 6401 &cmdStatus); 6402 if (status == 0) { 6403 status = cmdStatus; 6404 } 6405 return status; 6406} 6407 6408status_t AudioFlinger::EffectModule::start() 6409{ 6410 Mutex::Autolock _l(mLock); 6411 return start_l(); 6412} 6413 6414status_t AudioFlinger::EffectModule::start_l() 6415{ 6416 if (mEffectInterface == NULL) { 6417 return NO_INIT; 6418 } 6419 status_t cmdStatus; 6420 uint32_t size = sizeof(status_t); 6421 status_t status = (*mEffectInterface)->command(mEffectInterface, 6422 EFFECT_CMD_ENABLE, 6423 0, 6424 NULL, 6425 &size, 6426 &cmdStatus); 6427 if (status == 0) { 6428 status = cmdStatus; 6429 } 6430 if (status == 0 && 6431 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6432 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6433 sp<ThreadBase> thread = mThread.promote(); 6434 if (thread != 0) { 6435 audio_stream_t *stream = thread->stream(); 6436 if (stream != NULL) { 6437 stream->add_audio_effect(stream, mEffectInterface); 6438 } 6439 } 6440 } 6441 return status; 6442} 6443 6444status_t AudioFlinger::EffectModule::stop() 6445{ 6446 Mutex::Autolock _l(mLock); 6447 return stop_l(); 6448} 6449 6450status_t AudioFlinger::EffectModule::stop_l() 6451{ 6452 if (mEffectInterface == NULL) { 6453 return NO_INIT; 6454 } 6455 status_t cmdStatus; 6456 uint32_t size = sizeof(status_t); 6457 status_t status = (*mEffectInterface)->command(mEffectInterface, 6458 EFFECT_CMD_DISABLE, 6459 0, 6460 NULL, 6461 &size, 6462 &cmdStatus); 6463 if (status == 0) { 6464 status = cmdStatus; 6465 } 6466 if (status == 0 && 6467 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6468 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6469 sp<ThreadBase> thread = mThread.promote(); 6470 if (thread != 0) { 6471 audio_stream_t *stream = thread->stream(); 6472 if (stream != NULL) { 6473 stream->remove_audio_effect(stream, mEffectInterface); 6474 } 6475 } 6476 } 6477 return status; 6478} 6479 6480status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6481 uint32_t cmdSize, 6482 void *pCmdData, 6483 uint32_t *replySize, 6484 void *pReplyData) 6485{ 6486 Mutex::Autolock _l(mLock); 6487// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6488 6489 if (mState == DESTROYED || mEffectInterface == NULL) { 6490 return NO_INIT; 6491 } 6492 status_t status = (*mEffectInterface)->command(mEffectInterface, 6493 cmdCode, 6494 cmdSize, 6495 pCmdData, 6496 replySize, 6497 pReplyData); 6498 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6499 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6500 for (size_t i = 1; i < mHandles.size(); i++) { 6501 sp<EffectHandle> h = mHandles[i].promote(); 6502 if (h != 0) { 6503 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6504 } 6505 } 6506 } 6507 return status; 6508} 6509 6510status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6511{ 6512 6513 Mutex::Autolock _l(mLock); 6514 ALOGV("setEnabled %p enabled %d", this, enabled); 6515 6516 if (enabled != isEnabled()) { 6517 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6518 if (enabled && status != NO_ERROR) { 6519 return status; 6520 } 6521 6522 switch (mState) { 6523 // going from disabled to enabled 6524 case IDLE: 6525 mState = STARTING; 6526 break; 6527 case STOPPED: 6528 mState = RESTART; 6529 break; 6530 case STOPPING: 6531 mState = ACTIVE; 6532 break; 6533 6534 // going from enabled to disabled 6535 case RESTART: 6536 mState = STOPPED; 6537 break; 6538 case STARTING: 6539 mState = IDLE; 6540 break; 6541 case ACTIVE: 6542 mState = STOPPING; 6543 break; 6544 case DESTROYED: 6545 return NO_ERROR; // simply ignore as we are being destroyed 6546 } 6547 for (size_t i = 1; i < mHandles.size(); i++) { 6548 sp<EffectHandle> h = mHandles[i].promote(); 6549 if (h != 0) { 6550 h->setEnabled(enabled); 6551 } 6552 } 6553 } 6554 return NO_ERROR; 6555} 6556 6557bool AudioFlinger::EffectModule::isEnabled() 6558{ 6559 switch (mState) { 6560 case RESTART: 6561 case STARTING: 6562 case ACTIVE: 6563 return true; 6564 case IDLE: 6565 case STOPPING: 6566 case STOPPED: 6567 case DESTROYED: 6568 default: 6569 return false; 6570 } 6571} 6572 6573bool AudioFlinger::EffectModule::isProcessEnabled() 6574{ 6575 switch (mState) { 6576 case RESTART: 6577 case ACTIVE: 6578 case STOPPING: 6579 case STOPPED: 6580 return true; 6581 case IDLE: 6582 case STARTING: 6583 case DESTROYED: 6584 default: 6585 return false; 6586 } 6587} 6588 6589status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6590{ 6591 Mutex::Autolock _l(mLock); 6592 status_t status = NO_ERROR; 6593 6594 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6595 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6596 if (isProcessEnabled() && 6597 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6598 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6599 status_t cmdStatus; 6600 uint32_t volume[2]; 6601 uint32_t *pVolume = NULL; 6602 uint32_t size = sizeof(volume); 6603 volume[0] = *left; 6604 volume[1] = *right; 6605 if (controller) { 6606 pVolume = volume; 6607 } 6608 status = (*mEffectInterface)->command(mEffectInterface, 6609 EFFECT_CMD_SET_VOLUME, 6610 size, 6611 volume, 6612 &size, 6613 pVolume); 6614 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6615 *left = volume[0]; 6616 *right = volume[1]; 6617 } 6618 } 6619 return status; 6620} 6621 6622status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6623{ 6624 Mutex::Autolock _l(mLock); 6625 status_t status = NO_ERROR; 6626 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6627 // audio pre processing modules on RecordThread can receive both output and 6628 // input device indication in the same call 6629 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6630 if (dev) { 6631 status_t cmdStatus; 6632 uint32_t size = sizeof(status_t); 6633 6634 status = (*mEffectInterface)->command(mEffectInterface, 6635 EFFECT_CMD_SET_DEVICE, 6636 sizeof(uint32_t), 6637 &dev, 6638 &size, 6639 &cmdStatus); 6640 if (status == NO_ERROR) { 6641 status = cmdStatus; 6642 } 6643 } 6644 dev = device & AUDIO_DEVICE_IN_ALL; 6645 if (dev) { 6646 status_t cmdStatus; 6647 uint32_t size = sizeof(status_t); 6648 6649 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6650 EFFECT_CMD_SET_INPUT_DEVICE, 6651 sizeof(uint32_t), 6652 &dev, 6653 &size, 6654 &cmdStatus); 6655 if (status2 == NO_ERROR) { 6656 status2 = cmdStatus; 6657 } 6658 if (status == NO_ERROR) { 6659 status = status2; 6660 } 6661 } 6662 } 6663 return status; 6664} 6665 6666status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 6667{ 6668 Mutex::Autolock _l(mLock); 6669 status_t status = NO_ERROR; 6670 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6671 status_t cmdStatus; 6672 uint32_t size = sizeof(status_t); 6673 status = (*mEffectInterface)->command(mEffectInterface, 6674 EFFECT_CMD_SET_AUDIO_MODE, 6675 sizeof(int), 6676 &mode, 6677 &size, 6678 &cmdStatus); 6679 if (status == NO_ERROR) { 6680 status = cmdStatus; 6681 } 6682 } 6683 return status; 6684} 6685 6686void AudioFlinger::EffectModule::setSuspended(bool suspended) 6687{ 6688 Mutex::Autolock _l(mLock); 6689 mSuspended = suspended; 6690} 6691bool AudioFlinger::EffectModule::suspended() 6692{ 6693 Mutex::Autolock _l(mLock); 6694 return mSuspended; 6695} 6696 6697status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6698{ 6699 const size_t SIZE = 256; 6700 char buffer[SIZE]; 6701 String8 result; 6702 6703 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6704 result.append(buffer); 6705 6706 bool locked = tryLock(mLock); 6707 // failed to lock - AudioFlinger is probably deadlocked 6708 if (!locked) { 6709 result.append("\t\tCould not lock Fx mutex:\n"); 6710 } 6711 6712 result.append("\t\tSession Status State Engine:\n"); 6713 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6714 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6715 result.append(buffer); 6716 6717 result.append("\t\tDescriptor:\n"); 6718 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6719 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6720 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6721 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6722 result.append(buffer); 6723 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6724 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6725 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6726 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6727 result.append(buffer); 6728 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6729 mDescriptor.apiVersion, 6730 mDescriptor.flags); 6731 result.append(buffer); 6732 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6733 mDescriptor.name); 6734 result.append(buffer); 6735 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6736 mDescriptor.implementor); 6737 result.append(buffer); 6738 6739 result.append("\t\t- Input configuration:\n"); 6740 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6741 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6742 (uint32_t)mConfig.inputCfg.buffer.raw, 6743 mConfig.inputCfg.buffer.frameCount, 6744 mConfig.inputCfg.samplingRate, 6745 mConfig.inputCfg.channels, 6746 mConfig.inputCfg.format); 6747 result.append(buffer); 6748 6749 result.append("\t\t- Output configuration:\n"); 6750 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6751 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6752 (uint32_t)mConfig.outputCfg.buffer.raw, 6753 mConfig.outputCfg.buffer.frameCount, 6754 mConfig.outputCfg.samplingRate, 6755 mConfig.outputCfg.channels, 6756 mConfig.outputCfg.format); 6757 result.append(buffer); 6758 6759 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6760 result.append(buffer); 6761 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6762 for (size_t i = 0; i < mHandles.size(); ++i) { 6763 sp<EffectHandle> handle = mHandles[i].promote(); 6764 if (handle != 0) { 6765 handle->dump(buffer, SIZE); 6766 result.append(buffer); 6767 } 6768 } 6769 6770 result.append("\n"); 6771 6772 write(fd, result.string(), result.length()); 6773 6774 if (locked) { 6775 mLock.unlock(); 6776 } 6777 6778 return NO_ERROR; 6779} 6780 6781// ---------------------------------------------------------------------------- 6782// EffectHandle implementation 6783// ---------------------------------------------------------------------------- 6784 6785#undef LOG_TAG 6786#define LOG_TAG "AudioFlinger::EffectHandle" 6787 6788AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6789 const sp<AudioFlinger::Client>& client, 6790 const sp<IEffectClient>& effectClient, 6791 int32_t priority) 6792 : BnEffect(), 6793 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6794 mPriority(priority), mHasControl(false), mEnabled(false) 6795{ 6796 ALOGV("constructor %p", this); 6797 6798 if (client == 0) { 6799 return; 6800 } 6801 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6802 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6803 if (mCblkMemory != 0) { 6804 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6805 6806 if (mCblk) { 6807 new(mCblk) effect_param_cblk_t(); 6808 mBuffer = (uint8_t *)mCblk + bufOffset; 6809 } 6810 } else { 6811 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6812 return; 6813 } 6814} 6815 6816AudioFlinger::EffectHandle::~EffectHandle() 6817{ 6818 ALOGV("Destructor %p", this); 6819 disconnect(false); 6820 ALOGV("Destructor DONE %p", this); 6821} 6822 6823status_t AudioFlinger::EffectHandle::enable() 6824{ 6825 ALOGV("enable %p", this); 6826 if (!mHasControl) return INVALID_OPERATION; 6827 if (mEffect == 0) return DEAD_OBJECT; 6828 6829 if (mEnabled) { 6830 return NO_ERROR; 6831 } 6832 6833 mEnabled = true; 6834 6835 sp<ThreadBase> thread = mEffect->thread().promote(); 6836 if (thread != 0) { 6837 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6838 } 6839 6840 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6841 if (mEffect->suspended()) { 6842 return NO_ERROR; 6843 } 6844 6845 status_t status = mEffect->setEnabled(true); 6846 if (status != NO_ERROR) { 6847 if (thread != 0) { 6848 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6849 } 6850 mEnabled = false; 6851 } 6852 return status; 6853} 6854 6855status_t AudioFlinger::EffectHandle::disable() 6856{ 6857 ALOGV("disable %p", this); 6858 if (!mHasControl) return INVALID_OPERATION; 6859 if (mEffect == 0) return DEAD_OBJECT; 6860 6861 if (!mEnabled) { 6862 return NO_ERROR; 6863 } 6864 mEnabled = false; 6865 6866 if (mEffect->suspended()) { 6867 return NO_ERROR; 6868 } 6869 6870 status_t status = mEffect->setEnabled(false); 6871 6872 sp<ThreadBase> thread = mEffect->thread().promote(); 6873 if (thread != 0) { 6874 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6875 } 6876 6877 return status; 6878} 6879 6880void AudioFlinger::EffectHandle::disconnect() 6881{ 6882 disconnect(true); 6883} 6884 6885void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6886{ 6887 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6888 if (mEffect == 0) { 6889 return; 6890 } 6891 mEffect->disconnect(this, unpiniflast); 6892 6893 if (mHasControl && mEnabled) { 6894 sp<ThreadBase> thread = mEffect->thread().promote(); 6895 if (thread != 0) { 6896 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6897 } 6898 } 6899 6900 // release sp on module => module destructor can be called now 6901 mEffect.clear(); 6902 if (mClient != 0) { 6903 if (mCblk) { 6904 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6905 } 6906 mCblkMemory.clear(); // and free the shared memory 6907 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6908 mClient.clear(); 6909 } 6910} 6911 6912status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6913 uint32_t cmdSize, 6914 void *pCmdData, 6915 uint32_t *replySize, 6916 void *pReplyData) 6917{ 6918// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6919// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6920 6921 // only get parameter command is permitted for applications not controlling the effect 6922 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6923 return INVALID_OPERATION; 6924 } 6925 if (mEffect == 0) return DEAD_OBJECT; 6926 if (mClient == 0) return INVALID_OPERATION; 6927 6928 // handle commands that are not forwarded transparently to effect engine 6929 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6930 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6931 // no risk to block the whole media server process or mixer threads is we are stuck here 6932 Mutex::Autolock _l(mCblk->lock); 6933 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6934 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6935 mCblk->serverIndex = 0; 6936 mCblk->clientIndex = 0; 6937 return BAD_VALUE; 6938 } 6939 status_t status = NO_ERROR; 6940 while (mCblk->serverIndex < mCblk->clientIndex) { 6941 int reply; 6942 uint32_t rsize = sizeof(int); 6943 int *p = (int *)(mBuffer + mCblk->serverIndex); 6944 int size = *p++; 6945 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6946 LOGW("command(): invalid parameter block size"); 6947 break; 6948 } 6949 effect_param_t *param = (effect_param_t *)p; 6950 if (param->psize == 0 || param->vsize == 0) { 6951 LOGW("command(): null parameter or value size"); 6952 mCblk->serverIndex += size; 6953 continue; 6954 } 6955 uint32_t psize = sizeof(effect_param_t) + 6956 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6957 param->vsize; 6958 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6959 psize, 6960 p, 6961 &rsize, 6962 &reply); 6963 // stop at first error encountered 6964 if (ret != NO_ERROR) { 6965 status = ret; 6966 *(int *)pReplyData = reply; 6967 break; 6968 } else if (reply != NO_ERROR) { 6969 *(int *)pReplyData = reply; 6970 break; 6971 } 6972 mCblk->serverIndex += size; 6973 } 6974 mCblk->serverIndex = 0; 6975 mCblk->clientIndex = 0; 6976 return status; 6977 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6978 *(int *)pReplyData = NO_ERROR; 6979 return enable(); 6980 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6981 *(int *)pReplyData = NO_ERROR; 6982 return disable(); 6983 } 6984 6985 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6986} 6987 6988sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 6989 return mCblkMemory; 6990} 6991 6992void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 6993{ 6994 ALOGV("setControl %p control %d", this, hasControl); 6995 6996 mHasControl = hasControl; 6997 mEnabled = enabled; 6998 6999 if (signal && mEffectClient != 0) { 7000 mEffectClient->controlStatusChanged(hasControl); 7001 } 7002} 7003 7004void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7005 uint32_t cmdSize, 7006 void *pCmdData, 7007 uint32_t replySize, 7008 void *pReplyData) 7009{ 7010 if (mEffectClient != 0) { 7011 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7012 } 7013} 7014 7015 7016 7017void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7018{ 7019 if (mEffectClient != 0) { 7020 mEffectClient->enableStatusChanged(enabled); 7021 } 7022} 7023 7024status_t AudioFlinger::EffectHandle::onTransact( 7025 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7026{ 7027 return BnEffect::onTransact(code, data, reply, flags); 7028} 7029 7030 7031void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7032{ 7033 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7034 7035 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7036 (mClient == NULL) ? getpid() : mClient->pid(), 7037 mPriority, 7038 mHasControl, 7039 !locked, 7040 mCblk ? mCblk->clientIndex : 0, 7041 mCblk ? mCblk->serverIndex : 0 7042 ); 7043 7044 if (locked) { 7045 mCblk->lock.unlock(); 7046 } 7047} 7048 7049#undef LOG_TAG 7050#define LOG_TAG "AudioFlinger::EffectChain" 7051 7052AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7053 int sessionId) 7054 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7055 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7056 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7057{ 7058 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7059 sp<ThreadBase> thread = mThread.promote(); 7060 if (thread == 0) { 7061 return; 7062 } 7063 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7064 thread->frameCount(); 7065} 7066 7067AudioFlinger::EffectChain::~EffectChain() 7068{ 7069 if (mOwnInBuffer) { 7070 delete mInBuffer; 7071 } 7072 7073} 7074 7075// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7076sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7077{ 7078 sp<EffectModule> effect; 7079 size_t size = mEffects.size(); 7080 7081 for (size_t i = 0; i < size; i++) { 7082 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7083 effect = mEffects[i]; 7084 break; 7085 } 7086 } 7087 return effect; 7088} 7089 7090// getEffectFromId_l() must be called with ThreadBase::mLock held 7091sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7092{ 7093 sp<EffectModule> effect; 7094 size_t size = mEffects.size(); 7095 7096 for (size_t i = 0; i < size; i++) { 7097 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7098 if (id == 0 || mEffects[i]->id() == id) { 7099 effect = mEffects[i]; 7100 break; 7101 } 7102 } 7103 return effect; 7104} 7105 7106// getEffectFromType_l() must be called with ThreadBase::mLock held 7107sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7108 const effect_uuid_t *type) 7109{ 7110 sp<EffectModule> effect; 7111 size_t size = mEffects.size(); 7112 7113 for (size_t i = 0; i < size; i++) { 7114 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7115 effect = mEffects[i]; 7116 break; 7117 } 7118 } 7119 return effect; 7120} 7121 7122// Must be called with EffectChain::mLock locked 7123void AudioFlinger::EffectChain::process_l() 7124{ 7125 sp<ThreadBase> thread = mThread.promote(); 7126 if (thread == 0) { 7127 LOGW("process_l(): cannot promote mixer thread"); 7128 return; 7129 } 7130 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7131 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7132 // always process effects unless no more tracks are on the session and the effect tail 7133 // has been rendered 7134 bool doProcess = true; 7135 if (!isGlobalSession) { 7136 bool tracksOnSession = (trackCnt() != 0); 7137 7138 if (!tracksOnSession && mTailBufferCount == 0) { 7139 doProcess = false; 7140 } 7141 7142 if (activeTrackCnt() == 0) { 7143 // if no track is active and the effect tail has not been rendered, 7144 // the input buffer must be cleared here as the mixer process will not do it 7145 if (tracksOnSession || mTailBufferCount > 0) { 7146 size_t numSamples = thread->frameCount() * thread->channelCount(); 7147 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7148 if (mTailBufferCount > 0) { 7149 mTailBufferCount--; 7150 } 7151 } 7152 } 7153 } 7154 7155 size_t size = mEffects.size(); 7156 if (doProcess) { 7157 for (size_t i = 0; i < size; i++) { 7158 mEffects[i]->process(); 7159 } 7160 } 7161 for (size_t i = 0; i < size; i++) { 7162 mEffects[i]->updateState(); 7163 } 7164} 7165 7166// addEffect_l() must be called with PlaybackThread::mLock held 7167status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7168{ 7169 effect_descriptor_t desc = effect->desc(); 7170 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7171 7172 Mutex::Autolock _l(mLock); 7173 effect->setChain(this); 7174 sp<ThreadBase> thread = mThread.promote(); 7175 if (thread == 0) { 7176 return NO_INIT; 7177 } 7178 effect->setThread(thread); 7179 7180 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7181 // Auxiliary effects are inserted at the beginning of mEffects vector as 7182 // they are processed first and accumulated in chain input buffer 7183 mEffects.insertAt(effect, 0); 7184 7185 // the input buffer for auxiliary effect contains mono samples in 7186 // 32 bit format. This is to avoid saturation in AudoMixer 7187 // accumulation stage. Saturation is done in EffectModule::process() before 7188 // calling the process in effect engine 7189 size_t numSamples = thread->frameCount(); 7190 int32_t *buffer = new int32_t[numSamples]; 7191 memset(buffer, 0, numSamples * sizeof(int32_t)); 7192 effect->setInBuffer((int16_t *)buffer); 7193 // auxiliary effects output samples to chain input buffer for further processing 7194 // by insert effects 7195 effect->setOutBuffer(mInBuffer); 7196 } else { 7197 // Insert effects are inserted at the end of mEffects vector as they are processed 7198 // after track and auxiliary effects. 7199 // Insert effect order as a function of indicated preference: 7200 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7201 // another effect is present 7202 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7203 // last effect claiming first position 7204 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7205 // first effect claiming last position 7206 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7207 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7208 // already present 7209 7210 int size = (int)mEffects.size(); 7211 int idx_insert = size; 7212 int idx_insert_first = -1; 7213 int idx_insert_last = -1; 7214 7215 for (int i = 0; i < size; i++) { 7216 effect_descriptor_t d = mEffects[i]->desc(); 7217 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7218 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7219 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7220 // check invalid effect chaining combinations 7221 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7222 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7223 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7224 return INVALID_OPERATION; 7225 } 7226 // remember position of first insert effect and by default 7227 // select this as insert position for new effect 7228 if (idx_insert == size) { 7229 idx_insert = i; 7230 } 7231 // remember position of last insert effect claiming 7232 // first position 7233 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7234 idx_insert_first = i; 7235 } 7236 // remember position of first insert effect claiming 7237 // last position 7238 if (iPref == EFFECT_FLAG_INSERT_LAST && 7239 idx_insert_last == -1) { 7240 idx_insert_last = i; 7241 } 7242 } 7243 } 7244 7245 // modify idx_insert from first position if needed 7246 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7247 if (idx_insert_last != -1) { 7248 idx_insert = idx_insert_last; 7249 } else { 7250 idx_insert = size; 7251 } 7252 } else { 7253 if (idx_insert_first != -1) { 7254 idx_insert = idx_insert_first + 1; 7255 } 7256 } 7257 7258 // always read samples from chain input buffer 7259 effect->setInBuffer(mInBuffer); 7260 7261 // if last effect in the chain, output samples to chain 7262 // output buffer, otherwise to chain input buffer 7263 if (idx_insert == size) { 7264 if (idx_insert != 0) { 7265 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7266 mEffects[idx_insert-1]->configure(); 7267 } 7268 effect->setOutBuffer(mOutBuffer); 7269 } else { 7270 effect->setOutBuffer(mInBuffer); 7271 } 7272 mEffects.insertAt(effect, idx_insert); 7273 7274 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7275 } 7276 effect->configure(); 7277 return NO_ERROR; 7278} 7279 7280// removeEffect_l() must be called with PlaybackThread::mLock held 7281size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7282{ 7283 Mutex::Autolock _l(mLock); 7284 int size = (int)mEffects.size(); 7285 int i; 7286 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7287 7288 for (i = 0; i < size; i++) { 7289 if (effect == mEffects[i]) { 7290 // calling stop here will remove pre-processing effect from the audio HAL. 7291 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7292 // the middle of a read from audio HAL 7293 if (mEffects[i]->state() == EffectModule::ACTIVE || 7294 mEffects[i]->state() == EffectModule::STOPPING) { 7295 mEffects[i]->stop(); 7296 } 7297 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7298 delete[] effect->inBuffer(); 7299 } else { 7300 if (i == size - 1 && i != 0) { 7301 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7302 mEffects[i - 1]->configure(); 7303 } 7304 } 7305 mEffects.removeAt(i); 7306 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7307 break; 7308 } 7309 } 7310 7311 return mEffects.size(); 7312} 7313 7314// setDevice_l() must be called with PlaybackThread::mLock held 7315void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7316{ 7317 size_t size = mEffects.size(); 7318 for (size_t i = 0; i < size; i++) { 7319 mEffects[i]->setDevice(device); 7320 } 7321} 7322 7323// setMode_l() must be called with PlaybackThread::mLock held 7324void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 7325{ 7326 size_t size = mEffects.size(); 7327 for (size_t i = 0; i < size; i++) { 7328 mEffects[i]->setMode(mode); 7329 } 7330} 7331 7332// setVolume_l() must be called with PlaybackThread::mLock held 7333bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7334{ 7335 uint32_t newLeft = *left; 7336 uint32_t newRight = *right; 7337 bool hasControl = false; 7338 int ctrlIdx = -1; 7339 size_t size = mEffects.size(); 7340 7341 // first update volume controller 7342 for (size_t i = size; i > 0; i--) { 7343 if (mEffects[i - 1]->isProcessEnabled() && 7344 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7345 ctrlIdx = i - 1; 7346 hasControl = true; 7347 break; 7348 } 7349 } 7350 7351 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7352 if (hasControl) { 7353 *left = mNewLeftVolume; 7354 *right = mNewRightVolume; 7355 } 7356 return hasControl; 7357 } 7358 7359 mVolumeCtrlIdx = ctrlIdx; 7360 mLeftVolume = newLeft; 7361 mRightVolume = newRight; 7362 7363 // second get volume update from volume controller 7364 if (ctrlIdx >= 0) { 7365 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7366 mNewLeftVolume = newLeft; 7367 mNewRightVolume = newRight; 7368 } 7369 // then indicate volume to all other effects in chain. 7370 // Pass altered volume to effects before volume controller 7371 // and requested volume to effects after controller 7372 uint32_t lVol = newLeft; 7373 uint32_t rVol = newRight; 7374 7375 for (size_t i = 0; i < size; i++) { 7376 if ((int)i == ctrlIdx) continue; 7377 // this also works for ctrlIdx == -1 when there is no volume controller 7378 if ((int)i > ctrlIdx) { 7379 lVol = *left; 7380 rVol = *right; 7381 } 7382 mEffects[i]->setVolume(&lVol, &rVol, false); 7383 } 7384 *left = newLeft; 7385 *right = newRight; 7386 7387 return hasControl; 7388} 7389 7390status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7391{ 7392 const size_t SIZE = 256; 7393 char buffer[SIZE]; 7394 String8 result; 7395 7396 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7397 result.append(buffer); 7398 7399 bool locked = tryLock(mLock); 7400 // failed to lock - AudioFlinger is probably deadlocked 7401 if (!locked) { 7402 result.append("\tCould not lock mutex:\n"); 7403 } 7404 7405 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7406 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7407 mEffects.size(), 7408 (uint32_t)mInBuffer, 7409 (uint32_t)mOutBuffer, 7410 mActiveTrackCnt); 7411 result.append(buffer); 7412 write(fd, result.string(), result.size()); 7413 7414 for (size_t i = 0; i < mEffects.size(); ++i) { 7415 sp<EffectModule> effect = mEffects[i]; 7416 if (effect != 0) { 7417 effect->dump(fd, args); 7418 } 7419 } 7420 7421 if (locked) { 7422 mLock.unlock(); 7423 } 7424 7425 return NO_ERROR; 7426} 7427 7428// must be called with ThreadBase::mLock held 7429void AudioFlinger::EffectChain::setEffectSuspended_l( 7430 const effect_uuid_t *type, bool suspend) 7431{ 7432 sp<SuspendedEffectDesc> desc; 7433 // use effect type UUID timelow as key as there is no real risk of identical 7434 // timeLow fields among effect type UUIDs. 7435 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7436 if (suspend) { 7437 if (index >= 0) { 7438 desc = mSuspendedEffects.valueAt(index); 7439 } else { 7440 desc = new SuspendedEffectDesc(); 7441 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7442 mSuspendedEffects.add(type->timeLow, desc); 7443 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7444 } 7445 if (desc->mRefCount++ == 0) { 7446 sp<EffectModule> effect = getEffectIfEnabled(type); 7447 if (effect != 0) { 7448 desc->mEffect = effect; 7449 effect->setSuspended(true); 7450 effect->setEnabled(false); 7451 } 7452 } 7453 } else { 7454 if (index < 0) { 7455 return; 7456 } 7457 desc = mSuspendedEffects.valueAt(index); 7458 if (desc->mRefCount <= 0) { 7459 LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7460 desc->mRefCount = 1; 7461 } 7462 if (--desc->mRefCount == 0) { 7463 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7464 if (desc->mEffect != 0) { 7465 sp<EffectModule> effect = desc->mEffect.promote(); 7466 if (effect != 0) { 7467 effect->setSuspended(false); 7468 sp<EffectHandle> handle = effect->controlHandle(); 7469 if (handle != 0) { 7470 effect->setEnabled(handle->enabled()); 7471 } 7472 } 7473 desc->mEffect.clear(); 7474 } 7475 mSuspendedEffects.removeItemsAt(index); 7476 } 7477 } 7478} 7479 7480// must be called with ThreadBase::mLock held 7481void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7482{ 7483 sp<SuspendedEffectDesc> desc; 7484 7485 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7486 if (suspend) { 7487 if (index >= 0) { 7488 desc = mSuspendedEffects.valueAt(index); 7489 } else { 7490 desc = new SuspendedEffectDesc(); 7491 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7492 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7493 } 7494 if (desc->mRefCount++ == 0) { 7495 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7496 for (size_t i = 0; i < effects.size(); i++) { 7497 setEffectSuspended_l(&effects[i]->desc().type, true); 7498 } 7499 } 7500 } else { 7501 if (index < 0) { 7502 return; 7503 } 7504 desc = mSuspendedEffects.valueAt(index); 7505 if (desc->mRefCount <= 0) { 7506 LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7507 desc->mRefCount = 1; 7508 } 7509 if (--desc->mRefCount == 0) { 7510 Vector<const effect_uuid_t *> types; 7511 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7512 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7513 continue; 7514 } 7515 types.add(&mSuspendedEffects.valueAt(i)->mType); 7516 } 7517 for (size_t i = 0; i < types.size(); i++) { 7518 setEffectSuspended_l(types[i], false); 7519 } 7520 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7521 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7522 } 7523 } 7524} 7525 7526 7527// The volume effect is used for automated tests only 7528#ifndef OPENSL_ES_H_ 7529static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7530 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7531const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7532#endif //OPENSL_ES_H_ 7533 7534bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7535{ 7536 // auxiliary effects and visualizer are never suspended on output mix 7537 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7538 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7539 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7540 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7541 return false; 7542 } 7543 return true; 7544} 7545 7546Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7547{ 7548 Vector< sp<EffectModule> > effects; 7549 for (size_t i = 0; i < mEffects.size(); i++) { 7550 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7551 continue; 7552 } 7553 effects.add(mEffects[i]); 7554 } 7555 return effects; 7556} 7557 7558sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7559 const effect_uuid_t *type) 7560{ 7561 sp<EffectModule> effect; 7562 effect = getEffectFromType_l(type); 7563 if (effect != 0 && !effect->isEnabled()) { 7564 effect.clear(); 7565 } 7566 return effect; 7567} 7568 7569void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7570 bool enabled) 7571{ 7572 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7573 if (enabled) { 7574 if (index < 0) { 7575 // if the effect is not suspend check if all effects are suspended 7576 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7577 if (index < 0) { 7578 return; 7579 } 7580 if (!isEffectEligibleForSuspend(effect->desc())) { 7581 return; 7582 } 7583 setEffectSuspended_l(&effect->desc().type, enabled); 7584 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7585 if (index < 0) { 7586 LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7587 return; 7588 } 7589 } 7590 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7591 effect->desc().type.timeLow); 7592 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7593 // if effect is requested to suspended but was not yet enabled, supend it now. 7594 if (desc->mEffect == 0) { 7595 desc->mEffect = effect; 7596 effect->setEnabled(false); 7597 effect->setSuspended(true); 7598 } 7599 } else { 7600 if (index < 0) { 7601 return; 7602 } 7603 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7604 effect->desc().type.timeLow); 7605 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7606 desc->mEffect.clear(); 7607 effect->setSuspended(false); 7608 } 7609} 7610 7611#undef LOG_TAG 7612#define LOG_TAG "AudioFlinger" 7613 7614// ---------------------------------------------------------------------------- 7615 7616status_t AudioFlinger::onTransact( 7617 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7618{ 7619 return BnAudioFlinger::onTransact(code, data, reply, flags); 7620} 7621 7622}; // namespace android 7623