AudioFlinger.cpp revision 7378ca506e4e20c2b2d4e94a131cf1b95831adb5
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <audio_effects/effect_visualizer.h> 54#include <audio_effects/effect_ns.h> 55#include <audio_effects/effect_aec.h> 56 57#include <audio_utils/primitives.h> 58 59#include <cpustats/ThreadCpuUsage.h> 60#include <powermanager/PowerManager.h> 61// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 62 63// ---------------------------------------------------------------------------- 64 65 66namespace android { 67 68static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 69static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 70 71//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 72static const float MAX_GAIN = 4096.0f; 73static const uint32_t MAX_GAIN_INT = 0x1000; 74 75// retry counts for buffer fill timeout 76// 50 * ~20msecs = 1 second 77static const int8_t kMaxTrackRetries = 50; 78static const int8_t kMaxTrackStartupRetries = 50; 79// allow less retry attempts on direct output thread. 80// direct outputs can be a scarce resource in audio hardware and should 81// be released as quickly as possible. 82static const int8_t kMaxTrackRetriesDirect = 2; 83 84static const int kDumpLockRetries = 50; 85static const int kDumpLockSleepUs = 20000; 86 87// don't warn about blocked writes or record buffer overflows more often than this 88static const nsecs_t kWarningThrottleNs = seconds(5); 89 90// RecordThread loop sleep time upon application overrun or audio HAL read error 91static const int kRecordThreadSleepUs = 5000; 92 93// maximum time to wait for setParameters to complete 94static const nsecs_t kSetParametersTimeoutNs = seconds(2); 95 96// minimum sleep time for the mixer thread loop when tracks are active but in underrun 97static const uint32_t kMinThreadSleepTimeUs = 5000; 98// maximum divider applied to the active sleep time in the mixer thread loop 99static const uint32_t kMaxThreadSleepTimeShift = 2; 100 101 102// ---------------------------------------------------------------------------- 103 104static bool recordingAllowed() { 105 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 106 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 107 if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO"); 108 return ok; 109} 110 111static bool settingsAllowed() { 112 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 113 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 114 if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 115 return ok; 116} 117 118// To collect the amplifier usage 119static void addBatteryData(uint32_t params) { 120 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 121 if (service == NULL) { 122 // it already logged 123 return; 124 } 125 126 service->addBatteryData(params); 127} 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 165 mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 166 mMode(AUDIO_MODE_INVALID), 167 mBtNrecIsOff(false) 168{ 169} 170 171void AudioFlinger::onFirstRef() 172{ 173 int rc = 0; 174 175 Mutex::Autolock _l(mLock); 176 177 /* TODO: move all this work into an Init() function */ 178 179 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 180 const hw_module_t *mod; 181 audio_hw_device_t *dev; 182 183 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 184 if (rc) 185 continue; 186 187 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 188 mod->name, mod->id); 189 mAudioHwDevs.push(dev); 190 191 if (!mPrimaryHardwareDev) { 192 mPrimaryHardwareDev = dev; 193 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 194 mod->name, mod->id, audio_interfaces[i]); 195 } 196 } 197 198 mHardwareStatus = AUDIO_HW_INIT; 199 200 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 201 ALOGE("Primary audio interface not found"); 202 return; 203 } 204 205 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 206 audio_hw_device_t *dev = mAudioHwDevs[i]; 207 208 mHardwareStatus = AUDIO_HW_INIT; 209 rc = dev->init_check(dev); 210 if (rc == 0) { 211 AutoMutex lock(mHardwareLock); 212 213 mMode = AUDIO_MODE_NORMAL; 214 mHardwareStatus = AUDIO_HW_SET_MODE; 215 dev->set_mode(dev, mMode); 216 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 217 dev->set_master_volume(dev, 1.0f); 218 mHardwareStatus = AUDIO_HW_IDLE; 219 } 220 } 221} 222 223status_t AudioFlinger::initCheck() const 224{ 225 Mutex::Autolock _l(mLock); 226 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 227 return NO_INIT; 228 return NO_ERROR; 229} 230 231AudioFlinger::~AudioFlinger() 232{ 233 int num_devs = mAudioHwDevs.size(); 234 235 while (!mRecordThreads.isEmpty()) { 236 // closeInput() will remove first entry from mRecordThreads 237 closeInput(mRecordThreads.keyAt(0)); 238 } 239 while (!mPlaybackThreads.isEmpty()) { 240 // closeOutput() will remove first entry from mPlaybackThreads 241 closeOutput(mPlaybackThreads.keyAt(0)); 242 } 243 244 for (int i = 0; i < num_devs; i++) { 245 audio_hw_device_t *dev = mAudioHwDevs[i]; 246 audio_hw_device_close(dev); 247 } 248 mAudioHwDevs.clear(); 249} 250 251audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 252{ 253 /* first matching HW device is returned */ 254 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 255 audio_hw_device_t *dev = mAudioHwDevs[i]; 256 if ((dev->get_supported_devices(dev) & devices) == devices) 257 return dev; 258 } 259 return NULL; 260} 261 262status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 263{ 264 const size_t SIZE = 256; 265 char buffer[SIZE]; 266 String8 result; 267 268 result.append("Clients:\n"); 269 for (size_t i = 0; i < mClients.size(); ++i) { 270 sp<Client> client = mClients.valueAt(i).promote(); 271 if (client != 0) { 272 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 273 result.append(buffer); 274 } 275 } 276 277 result.append("Global session refs:\n"); 278 result.append(" session pid cnt\n"); 279 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 280 AudioSessionRef *r = mAudioSessionRefs[i]; 281 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 282 result.append(buffer); 283 } 284 write(fd, result.string(), result.size()); 285 return NO_ERROR; 286} 287 288 289status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 290{ 291 const size_t SIZE = 256; 292 char buffer[SIZE]; 293 String8 result; 294 hardware_call_state hardwareStatus = mHardwareStatus; 295 296 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 297 result.append(buffer); 298 write(fd, result.string(), result.size()); 299 return NO_ERROR; 300} 301 302status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 303{ 304 const size_t SIZE = 256; 305 char buffer[SIZE]; 306 String8 result; 307 snprintf(buffer, SIZE, "Permission Denial: " 308 "can't dump AudioFlinger from pid=%d, uid=%d\n", 309 IPCThreadState::self()->getCallingPid(), 310 IPCThreadState::self()->getCallingUid()); 311 result.append(buffer); 312 write(fd, result.string(), result.size()); 313 return NO_ERROR; 314} 315 316static bool tryLock(Mutex& mutex) 317{ 318 bool locked = false; 319 for (int i = 0; i < kDumpLockRetries; ++i) { 320 if (mutex.tryLock() == NO_ERROR) { 321 locked = true; 322 break; 323 } 324 usleep(kDumpLockSleepUs); 325 } 326 return locked; 327} 328 329status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 330{ 331 if (!checkCallingPermission(String16("android.permission.DUMP"))) { 332 dumpPermissionDenial(fd, args); 333 } else { 334 // get state of hardware lock 335 bool hardwareLocked = tryLock(mHardwareLock); 336 if (!hardwareLocked) { 337 String8 result(kHardwareLockedString); 338 write(fd, result.string(), result.size()); 339 } else { 340 mHardwareLock.unlock(); 341 } 342 343 bool locked = tryLock(mLock); 344 345 // failed to lock - AudioFlinger is probably deadlocked 346 if (!locked) { 347 String8 result(kDeadlockedString); 348 write(fd, result.string(), result.size()); 349 } 350 351 dumpClients(fd, args); 352 dumpInternals(fd, args); 353 354 // dump playback threads 355 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 356 mPlaybackThreads.valueAt(i)->dump(fd, args); 357 } 358 359 // dump record threads 360 for (size_t i = 0; i < mRecordThreads.size(); i++) { 361 mRecordThreads.valueAt(i)->dump(fd, args); 362 } 363 364 // dump all hardware devs 365 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 366 audio_hw_device_t *dev = mAudioHwDevs[i]; 367 dev->dump(dev, fd); 368 } 369 if (locked) mLock.unlock(); 370 } 371 return NO_ERROR; 372} 373 374 375// IAudioFlinger interface 376 377 378sp<IAudioTrack> AudioFlinger::createTrack( 379 pid_t pid, 380 audio_stream_type_t streamType, 381 uint32_t sampleRate, 382 audio_format_t format, 383 uint32_t channelMask, 384 int frameCount, 385 uint32_t flags, 386 const sp<IMemory>& sharedBuffer, 387 int output, 388 int *sessionId, 389 status_t *status) 390{ 391 sp<PlaybackThread::Track> track; 392 sp<TrackHandle> trackHandle; 393 sp<Client> client; 394 wp<Client> wclient; 395 status_t lStatus; 396 int lSessionId; 397 398 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 399 // but if someone uses binder directly they could bypass that and cause us to crash 400 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 401 ALOGE("createTrack() invalid stream type %d", streamType); 402 lStatus = BAD_VALUE; 403 goto Exit; 404 } 405 406 { 407 Mutex::Autolock _l(mLock); 408 PlaybackThread *thread = checkPlaybackThread_l(output); 409 PlaybackThread *effectThread = NULL; 410 if (thread == NULL) { 411 ALOGE("unknown output thread"); 412 lStatus = BAD_VALUE; 413 goto Exit; 414 } 415 416 wclient = mClients.valueFor(pid); 417 418 if (wclient != NULL) { 419 client = wclient.promote(); 420 } else { 421 client = new Client(this, pid); 422 mClients.add(pid, client); 423 } 424 425 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 426 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 427 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 428 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 429 if (mPlaybackThreads.keyAt(i) != output) { 430 // prevent same audio session on different output threads 431 uint32_t sessions = t->hasAudioSession(*sessionId); 432 if (sessions & PlaybackThread::TRACK_SESSION) { 433 ALOGE("createTrack() session ID %d already in use", *sessionId); 434 lStatus = BAD_VALUE; 435 goto Exit; 436 } 437 // check if an effect with same session ID is waiting for a track to be created 438 if (sessions & PlaybackThread::EFFECT_SESSION) { 439 effectThread = t.get(); 440 } 441 } 442 } 443 lSessionId = *sessionId; 444 } else { 445 // if no audio session id is provided, create one here 446 lSessionId = nextUniqueId(); 447 if (sessionId != NULL) { 448 *sessionId = lSessionId; 449 } 450 } 451 ALOGV("createTrack() lSessionId: %d", lSessionId); 452 453 track = thread->createTrack_l(client, streamType, sampleRate, format, 454 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 455 456 // move effect chain to this output thread if an effect on same session was waiting 457 // for a track to be created 458 if (lStatus == NO_ERROR && effectThread != NULL) { 459 Mutex::Autolock _dl(thread->mLock); 460 Mutex::Autolock _sl(effectThread->mLock); 461 moveEffectChain_l(lSessionId, effectThread, thread, true); 462 } 463 } 464 if (lStatus == NO_ERROR) { 465 trackHandle = new TrackHandle(track); 466 } else { 467 // remove local strong reference to Client before deleting the Track so that the Client 468 // destructor is called by the TrackBase destructor with mLock held 469 client.clear(); 470 track.clear(); 471 } 472 473Exit: 474 if(status) { 475 *status = lStatus; 476 } 477 return trackHandle; 478} 479 480uint32_t AudioFlinger::sampleRate(int output) const 481{ 482 Mutex::Autolock _l(mLock); 483 PlaybackThread *thread = checkPlaybackThread_l(output); 484 if (thread == NULL) { 485 ALOGW("sampleRate() unknown thread %d", output); 486 return 0; 487 } 488 return thread->sampleRate(); 489} 490 491int AudioFlinger::channelCount(int output) const 492{ 493 Mutex::Autolock _l(mLock); 494 PlaybackThread *thread = checkPlaybackThread_l(output); 495 if (thread == NULL) { 496 ALOGW("channelCount() unknown thread %d", output); 497 return 0; 498 } 499 return thread->channelCount(); 500} 501 502audio_format_t AudioFlinger::format(int output) const 503{ 504 Mutex::Autolock _l(mLock); 505 PlaybackThread *thread = checkPlaybackThread_l(output); 506 if (thread == NULL) { 507 ALOGW("format() unknown thread %d", output); 508 return AUDIO_FORMAT_INVALID; 509 } 510 return thread->format(); 511} 512 513size_t AudioFlinger::frameCount(int output) const 514{ 515 Mutex::Autolock _l(mLock); 516 PlaybackThread *thread = checkPlaybackThread_l(output); 517 if (thread == NULL) { 518 ALOGW("frameCount() unknown thread %d", output); 519 return 0; 520 } 521 return thread->frameCount(); 522} 523 524uint32_t AudioFlinger::latency(int output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("latency() unknown thread %d", output); 530 return 0; 531 } 532 return thread->latency(); 533} 534 535status_t AudioFlinger::setMasterVolume(float value) 536{ 537 status_t ret = initCheck(); 538 if (ret != NO_ERROR) { 539 return ret; 540 } 541 542 // check calling permissions 543 if (!settingsAllowed()) { 544 return PERMISSION_DENIED; 545 } 546 547 // when hw supports master volume, don't scale in sw mixer 548 { // scope for the lock 549 AutoMutex lock(mHardwareLock); 550 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 551 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 552 value = 1.0f; 553 } 554 mHardwareStatus = AUDIO_HW_IDLE; 555 } 556 557 Mutex::Autolock _l(mLock); 558 mMasterVolume = value; 559 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 560 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 561 562 return NO_ERROR; 563} 564 565status_t AudioFlinger::setMode(audio_mode_t mode) 566{ 567 status_t ret = initCheck(); 568 if (ret != NO_ERROR) { 569 return ret; 570 } 571 572 // check calling permissions 573 if (!settingsAllowed()) { 574 return PERMISSION_DENIED; 575 } 576 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 577 ALOGW("Illegal value: setMode(%d)", mode); 578 return BAD_VALUE; 579 } 580 581 { // scope for the lock 582 AutoMutex lock(mHardwareLock); 583 mHardwareStatus = AUDIO_HW_SET_MODE; 584 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 585 mHardwareStatus = AUDIO_HW_IDLE; 586 } 587 588 if (NO_ERROR == ret) { 589 Mutex::Autolock _l(mLock); 590 mMode = mode; 591 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 592 mPlaybackThreads.valueAt(i)->setMode(mode); 593 } 594 595 return ret; 596} 597 598status_t AudioFlinger::setMicMute(bool state) 599{ 600 status_t ret = initCheck(); 601 if (ret != NO_ERROR) { 602 return ret; 603 } 604 605 // check calling permissions 606 if (!settingsAllowed()) { 607 return PERMISSION_DENIED; 608 } 609 610 AutoMutex lock(mHardwareLock); 611 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 612 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 613 mHardwareStatus = AUDIO_HW_IDLE; 614 return ret; 615} 616 617bool AudioFlinger::getMicMute() const 618{ 619 status_t ret = initCheck(); 620 if (ret != NO_ERROR) { 621 return false; 622 } 623 624 bool state = AUDIO_MODE_INVALID; 625 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 626 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 627 mHardwareStatus = AUDIO_HW_IDLE; 628 return state; 629} 630 631status_t AudioFlinger::setMasterMute(bool muted) 632{ 633 // check calling permissions 634 if (!settingsAllowed()) { 635 return PERMISSION_DENIED; 636 } 637 638 Mutex::Autolock _l(mLock); 639 mMasterMute = muted; 640 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 641 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 642 643 return NO_ERROR; 644} 645 646float AudioFlinger::masterVolume() const 647{ 648 Mutex::Autolock _l(mLock); 649 return masterVolume_l(); 650} 651 652bool AudioFlinger::masterMute() const 653{ 654 Mutex::Autolock _l(mLock); 655 return masterMute_l(); 656} 657 658status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output) 659{ 660 // check calling permissions 661 if (!settingsAllowed()) { 662 return PERMISSION_DENIED; 663 } 664 665 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 666 ALOGE("setStreamVolume() invalid stream %d", stream); 667 return BAD_VALUE; 668 } 669 670 AutoMutex lock(mLock); 671 PlaybackThread *thread = NULL; 672 if (output) { 673 thread = checkPlaybackThread_l(output); 674 if (thread == NULL) { 675 return BAD_VALUE; 676 } 677 } 678 679 mStreamTypes[stream].volume = value; 680 681 if (thread == NULL) { 682 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 683 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 684 } 685 } else { 686 thread->setStreamVolume(stream, value); 687 } 688 689 return NO_ERROR; 690} 691 692status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 693{ 694 // check calling permissions 695 if (!settingsAllowed()) { 696 return PERMISSION_DENIED; 697 } 698 699 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 700 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 701 ALOGE("setStreamMute() invalid stream %d", stream); 702 return BAD_VALUE; 703 } 704 705 AutoMutex lock(mLock); 706 mStreamTypes[stream].mute = muted; 707 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 708 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 709 710 return NO_ERROR; 711} 712 713float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const 714{ 715 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 716 return 0.0f; 717 } 718 719 AutoMutex lock(mLock); 720 float volume; 721 if (output) { 722 PlaybackThread *thread = checkPlaybackThread_l(output); 723 if (thread == NULL) { 724 return 0.0f; 725 } 726 volume = thread->streamVolume(stream); 727 } else { 728 volume = mStreamTypes[stream].volume; 729 } 730 731 return volume; 732} 733 734bool AudioFlinger::streamMute(audio_stream_type_t stream) const 735{ 736 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 737 return true; 738 } 739 740 return mStreamTypes[stream].mute; 741} 742 743status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 744{ 745 status_t result; 746 747 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 748 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 749 // check calling permissions 750 if (!settingsAllowed()) { 751 return PERMISSION_DENIED; 752 } 753 754 // ioHandle == 0 means the parameters are global to the audio hardware interface 755 if (ioHandle == 0) { 756 AutoMutex lock(mHardwareLock); 757 mHardwareStatus = AUDIO_SET_PARAMETER; 758 status_t final_result = NO_ERROR; 759 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 760 audio_hw_device_t *dev = mAudioHwDevs[i]; 761 result = dev->set_parameters(dev, keyValuePairs.string()); 762 final_result = result ?: final_result; 763 } 764 mHardwareStatus = AUDIO_HW_IDLE; 765 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 766 AudioParameter param = AudioParameter(keyValuePairs); 767 String8 value; 768 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 769 Mutex::Autolock _l(mLock); 770 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 771 if (mBtNrecIsOff != btNrecIsOff) { 772 for (size_t i = 0; i < mRecordThreads.size(); i++) { 773 sp<RecordThread> thread = mRecordThreads.valueAt(i); 774 RecordThread::RecordTrack *track = thread->track(); 775 if (track != NULL) { 776 audio_devices_t device = (audio_devices_t)( 777 thread->device() & AUDIO_DEVICE_IN_ALL); 778 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 779 thread->setEffectSuspended(FX_IID_AEC, 780 suspend, 781 track->sessionId()); 782 thread->setEffectSuspended(FX_IID_NS, 783 suspend, 784 track->sessionId()); 785 } 786 } 787 mBtNrecIsOff = btNrecIsOff; 788 } 789 } 790 return final_result; 791 } 792 793 // hold a strong ref on thread in case closeOutput() or closeInput() is called 794 // and the thread is exited once the lock is released 795 sp<ThreadBase> thread; 796 { 797 Mutex::Autolock _l(mLock); 798 thread = checkPlaybackThread_l(ioHandle); 799 if (thread == NULL) { 800 thread = checkRecordThread_l(ioHandle); 801 } else if (thread == primaryPlaybackThread_l()) { 802 // indicate output device change to all input threads for pre processing 803 AudioParameter param = AudioParameter(keyValuePairs); 804 int value; 805 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 806 for (size_t i = 0; i < mRecordThreads.size(); i++) { 807 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 808 } 809 } 810 } 811 } 812 if (thread != 0) { 813 return thread->setParameters(keyValuePairs); 814 } 815 return BAD_VALUE; 816} 817 818String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 819{ 820// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 821// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 822 823 if (ioHandle == 0) { 824 String8 out_s8; 825 826 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 827 audio_hw_device_t *dev = mAudioHwDevs[i]; 828 char *s = dev->get_parameters(dev, keys.string()); 829 out_s8 += String8(s); 830 free(s); 831 } 832 return out_s8; 833 } 834 835 Mutex::Autolock _l(mLock); 836 837 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 838 if (playbackThread != NULL) { 839 return playbackThread->getParameters(keys); 840 } 841 RecordThread *recordThread = checkRecordThread_l(ioHandle); 842 if (recordThread != NULL) { 843 return recordThread->getParameters(keys); 844 } 845 return String8(""); 846} 847 848size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) 849{ 850 status_t ret = initCheck(); 851 if (ret != NO_ERROR) { 852 return 0; 853 } 854 855 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 856} 857 858unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 859{ 860 if (ioHandle == 0) { 861 return 0; 862 } 863 864 Mutex::Autolock _l(mLock); 865 866 RecordThread *recordThread = checkRecordThread_l(ioHandle); 867 if (recordThread != NULL) { 868 return recordThread->getInputFramesLost(); 869 } 870 return 0; 871} 872 873status_t AudioFlinger::setVoiceVolume(float value) 874{ 875 status_t ret = initCheck(); 876 if (ret != NO_ERROR) { 877 return ret; 878 } 879 880 // check calling permissions 881 if (!settingsAllowed()) { 882 return PERMISSION_DENIED; 883 } 884 885 AutoMutex lock(mHardwareLock); 886 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 887 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 888 mHardwareStatus = AUDIO_HW_IDLE; 889 890 return ret; 891} 892 893status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 894{ 895 status_t status; 896 897 Mutex::Autolock _l(mLock); 898 899 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 900 if (playbackThread != NULL) { 901 return playbackThread->getRenderPosition(halFrames, dspFrames); 902 } 903 904 return BAD_VALUE; 905} 906 907void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 908{ 909 910 Mutex::Autolock _l(mLock); 911 912 int pid = IPCThreadState::self()->getCallingPid(); 913 if (mNotificationClients.indexOfKey(pid) < 0) { 914 sp<NotificationClient> notificationClient = new NotificationClient(this, 915 client, 916 pid); 917 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 918 919 mNotificationClients.add(pid, notificationClient); 920 921 sp<IBinder> binder = client->asBinder(); 922 binder->linkToDeath(notificationClient); 923 924 // the config change is always sent from playback or record threads to avoid deadlock 925 // with AudioSystem::gLock 926 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 927 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 928 } 929 930 for (size_t i = 0; i < mRecordThreads.size(); i++) { 931 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 932 } 933 } 934} 935 936void AudioFlinger::removeNotificationClient(pid_t pid) 937{ 938 Mutex::Autolock _l(mLock); 939 940 int index = mNotificationClients.indexOfKey(pid); 941 if (index >= 0) { 942 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 943 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 944 mNotificationClients.removeItem(pid); 945 } 946 947 ALOGV("%d died, releasing its sessions", pid); 948 int num = mAudioSessionRefs.size(); 949 bool removed = false; 950 for (int i = 0; i< num; i++) { 951 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 952 ALOGV(" pid %d @ %d", ref->pid, i); 953 if (ref->pid == pid) { 954 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 955 mAudioSessionRefs.removeAt(i); 956 delete ref; 957 removed = true; 958 i--; 959 num--; 960 } 961 } 962 if (removed) { 963 purgeStaleEffects_l(); 964 } 965} 966 967// audioConfigChanged_l() must be called with AudioFlinger::mLock held 968void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 969{ 970 size_t size = mNotificationClients.size(); 971 for (size_t i = 0; i < size; i++) { 972 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 973 param2); 974 } 975} 976 977// removeClient_l() must be called with AudioFlinger::mLock held 978void AudioFlinger::removeClient_l(pid_t pid) 979{ 980 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 981 mClients.removeItem(pid); 982} 983 984 985// ---------------------------------------------------------------------------- 986 987AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device, 988 type_t type) 989 : Thread(false), 990 mType(type), 991 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 992 // mChannelMask 993 mChannelCount(0), 994 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 995 mParamStatus(NO_ERROR), 996 mStandby(false), mId(id), mExiting(false), 997 mDevice(device), 998 mDeathRecipient(new PMDeathRecipient(this)) 999{ 1000} 1001 1002AudioFlinger::ThreadBase::~ThreadBase() 1003{ 1004 mParamCond.broadcast(); 1005 // do not lock the mutex in destructor 1006 releaseWakeLock_l(); 1007 if (mPowerManager != 0) { 1008 sp<IBinder> binder = mPowerManager->asBinder(); 1009 binder->unlinkToDeath(mDeathRecipient); 1010 } 1011} 1012 1013void AudioFlinger::ThreadBase::exit() 1014{ 1015 // keep a strong ref on ourself so that we won't get 1016 // destroyed in the middle of requestExitAndWait() 1017 sp <ThreadBase> strongMe = this; 1018 1019 ALOGV("ThreadBase::exit"); 1020 { 1021 AutoMutex lock(mLock); 1022 mExiting = true; 1023 requestExit(); 1024 mWaitWorkCV.signal(); 1025 } 1026 requestExitAndWait(); 1027} 1028 1029uint32_t AudioFlinger::ThreadBase::sampleRate() const 1030{ 1031 return mSampleRate; 1032} 1033 1034int AudioFlinger::ThreadBase::channelCount() const 1035{ 1036 return (int)mChannelCount; 1037} 1038 1039audio_format_t AudioFlinger::ThreadBase::format() const 1040{ 1041 return mFormat; 1042} 1043 1044size_t AudioFlinger::ThreadBase::frameCount() const 1045{ 1046 return mFrameCount; 1047} 1048 1049status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1050{ 1051 status_t status; 1052 1053 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1054 Mutex::Autolock _l(mLock); 1055 1056 mNewParameters.add(keyValuePairs); 1057 mWaitWorkCV.signal(); 1058 // wait condition with timeout in case the thread loop has exited 1059 // before the request could be processed 1060 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1061 status = mParamStatus; 1062 mWaitWorkCV.signal(); 1063 } else { 1064 status = TIMED_OUT; 1065 } 1066 return status; 1067} 1068 1069void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1070{ 1071 Mutex::Autolock _l(mLock); 1072 sendConfigEvent_l(event, param); 1073} 1074 1075// sendConfigEvent_l() must be called with ThreadBase::mLock held 1076void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1077{ 1078 ConfigEvent configEvent; 1079 configEvent.mEvent = event; 1080 configEvent.mParam = param; 1081 mConfigEvents.add(configEvent); 1082 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1083 mWaitWorkCV.signal(); 1084} 1085 1086void AudioFlinger::ThreadBase::processConfigEvents() 1087{ 1088 mLock.lock(); 1089 while(!mConfigEvents.isEmpty()) { 1090 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1091 ConfigEvent configEvent = mConfigEvents[0]; 1092 mConfigEvents.removeAt(0); 1093 // release mLock before locking AudioFlinger mLock: lock order is always 1094 // AudioFlinger then ThreadBase to avoid cross deadlock 1095 mLock.unlock(); 1096 mAudioFlinger->mLock.lock(); 1097 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1098 mAudioFlinger->mLock.unlock(); 1099 mLock.lock(); 1100 } 1101 mLock.unlock(); 1102} 1103 1104status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1105{ 1106 const size_t SIZE = 256; 1107 char buffer[SIZE]; 1108 String8 result; 1109 1110 bool locked = tryLock(mLock); 1111 if (!locked) { 1112 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1113 write(fd, buffer, strlen(buffer)); 1114 } 1115 1116 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1117 result.append(buffer); 1118 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1119 result.append(buffer); 1120 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1121 result.append(buffer); 1122 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1123 result.append(buffer); 1124 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1125 result.append(buffer); 1126 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1127 result.append(buffer); 1128 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1129 result.append(buffer); 1130 1131 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1132 result.append(buffer); 1133 result.append(" Index Command"); 1134 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1135 snprintf(buffer, SIZE, "\n %02d ", i); 1136 result.append(buffer); 1137 result.append(mNewParameters[i]); 1138 } 1139 1140 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1141 result.append(buffer); 1142 snprintf(buffer, SIZE, " Index event param\n"); 1143 result.append(buffer); 1144 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1145 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1146 result.append(buffer); 1147 } 1148 result.append("\n"); 1149 1150 write(fd, result.string(), result.size()); 1151 1152 if (locked) { 1153 mLock.unlock(); 1154 } 1155 return NO_ERROR; 1156} 1157 1158status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1159{ 1160 const size_t SIZE = 256; 1161 char buffer[SIZE]; 1162 String8 result; 1163 1164 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1165 write(fd, buffer, strlen(buffer)); 1166 1167 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1168 sp<EffectChain> chain = mEffectChains[i]; 1169 if (chain != 0) { 1170 chain->dump(fd, args); 1171 } 1172 } 1173 return NO_ERROR; 1174} 1175 1176void AudioFlinger::ThreadBase::acquireWakeLock() 1177{ 1178 Mutex::Autolock _l(mLock); 1179 acquireWakeLock_l(); 1180} 1181 1182void AudioFlinger::ThreadBase::acquireWakeLock_l() 1183{ 1184 if (mPowerManager == 0) { 1185 // use checkService() to avoid blocking if power service is not up yet 1186 sp<IBinder> binder = 1187 defaultServiceManager()->checkService(String16("power")); 1188 if (binder == 0) { 1189 ALOGW("Thread %s cannot connect to the power manager service", mName); 1190 } else { 1191 mPowerManager = interface_cast<IPowerManager>(binder); 1192 binder->linkToDeath(mDeathRecipient); 1193 } 1194 } 1195 if (mPowerManager != 0) { 1196 sp<IBinder> binder = new BBinder(); 1197 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1198 binder, 1199 String16(mName)); 1200 if (status == NO_ERROR) { 1201 mWakeLockToken = binder; 1202 } 1203 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1204 } 1205} 1206 1207void AudioFlinger::ThreadBase::releaseWakeLock() 1208{ 1209 Mutex::Autolock _l(mLock); 1210 releaseWakeLock_l(); 1211} 1212 1213void AudioFlinger::ThreadBase::releaseWakeLock_l() 1214{ 1215 if (mWakeLockToken != 0) { 1216 ALOGV("releaseWakeLock_l() %s", mName); 1217 if (mPowerManager != 0) { 1218 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1219 } 1220 mWakeLockToken.clear(); 1221 } 1222} 1223 1224void AudioFlinger::ThreadBase::clearPowerManager() 1225{ 1226 Mutex::Autolock _l(mLock); 1227 releaseWakeLock_l(); 1228 mPowerManager.clear(); 1229} 1230 1231void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1232{ 1233 sp<ThreadBase> thread = mThread.promote(); 1234 if (thread != 0) { 1235 thread->clearPowerManager(); 1236 } 1237 ALOGW("power manager service died !!!"); 1238} 1239 1240void AudioFlinger::ThreadBase::setEffectSuspended( 1241 const effect_uuid_t *type, bool suspend, int sessionId) 1242{ 1243 Mutex::Autolock _l(mLock); 1244 setEffectSuspended_l(type, suspend, sessionId); 1245} 1246 1247void AudioFlinger::ThreadBase::setEffectSuspended_l( 1248 const effect_uuid_t *type, bool suspend, int sessionId) 1249{ 1250 sp<EffectChain> chain; 1251 chain = getEffectChain_l(sessionId); 1252 if (chain != 0) { 1253 if (type != NULL) { 1254 chain->setEffectSuspended_l(type, suspend); 1255 } else { 1256 chain->setEffectSuspendedAll_l(suspend); 1257 } 1258 } 1259 1260 updateSuspendedSessions_l(type, suspend, sessionId); 1261} 1262 1263void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1264{ 1265 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1266 if (index < 0) { 1267 return; 1268 } 1269 1270 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1271 mSuspendedSessions.editValueAt(index); 1272 1273 for (size_t i = 0; i < sessionEffects.size(); i++) { 1274 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1275 for (int j = 0; j < desc->mRefCount; j++) { 1276 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1277 chain->setEffectSuspendedAll_l(true); 1278 } else { 1279 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1280 desc->mType.timeLow); 1281 chain->setEffectSuspended_l(&desc->mType, true); 1282 } 1283 } 1284 } 1285} 1286 1287void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1288 bool suspend, 1289 int sessionId) 1290{ 1291 int index = mSuspendedSessions.indexOfKey(sessionId); 1292 1293 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1294 1295 if (suspend) { 1296 if (index >= 0) { 1297 sessionEffects = mSuspendedSessions.editValueAt(index); 1298 } else { 1299 mSuspendedSessions.add(sessionId, sessionEffects); 1300 } 1301 } else { 1302 if (index < 0) { 1303 return; 1304 } 1305 sessionEffects = mSuspendedSessions.editValueAt(index); 1306 } 1307 1308 1309 int key = EffectChain::kKeyForSuspendAll; 1310 if (type != NULL) { 1311 key = type->timeLow; 1312 } 1313 index = sessionEffects.indexOfKey(key); 1314 1315 sp <SuspendedSessionDesc> desc; 1316 if (suspend) { 1317 if (index >= 0) { 1318 desc = sessionEffects.valueAt(index); 1319 } else { 1320 desc = new SuspendedSessionDesc(); 1321 if (type != NULL) { 1322 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1323 } 1324 sessionEffects.add(key, desc); 1325 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1326 } 1327 desc->mRefCount++; 1328 } else { 1329 if (index < 0) { 1330 return; 1331 } 1332 desc = sessionEffects.valueAt(index); 1333 if (--desc->mRefCount == 0) { 1334 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1335 sessionEffects.removeItemsAt(index); 1336 if (sessionEffects.isEmpty()) { 1337 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1338 sessionId); 1339 mSuspendedSessions.removeItem(sessionId); 1340 } 1341 } 1342 } 1343 if (!sessionEffects.isEmpty()) { 1344 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1345 } 1346} 1347 1348void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1349 bool enabled, 1350 int sessionId) 1351{ 1352 Mutex::Autolock _l(mLock); 1353 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1354} 1355 1356void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1357 bool enabled, 1358 int sessionId) 1359{ 1360 if (mType != RECORD) { 1361 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1362 // another session. This gives the priority to well behaved effect control panels 1363 // and applications not using global effects. 1364 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1365 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1366 } 1367 } 1368 1369 sp<EffectChain> chain = getEffectChain_l(sessionId); 1370 if (chain != 0) { 1371 chain->checkSuspendOnEffectEnabled(effect, enabled); 1372 } 1373} 1374 1375// ---------------------------------------------------------------------------- 1376 1377AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1378 AudioStreamOut* output, 1379 int id, 1380 uint32_t device, 1381 type_t type) 1382 : ThreadBase(audioFlinger, id, device, type), 1383 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1384 // Assumes constructor is called by AudioFlinger with it's mLock held, 1385 // but it would be safer to explicitly pass initial masterMute as parameter 1386 mMasterMute(audioFlinger->masterMute_l()), 1387 // mStreamTypes[] initialized in constructor body 1388 mOutput(output), 1389 // Assumes constructor is called by AudioFlinger with it's mLock held, 1390 // but it would be safer to explicitly pass initial masterVolume as parameter 1391 mMasterVolume(audioFlinger->masterVolume_l()), 1392 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1393{ 1394 snprintf(mName, kNameLength, "AudioOut_%d", id); 1395 1396 readOutputParameters(); 1397 1398 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1399 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1400 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1401 stream = (audio_stream_type_t) (stream + 1)) { 1402 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1403 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1404 // initialized by stream_type_t default constructor 1405 // mStreamTypes[stream].valid = true; 1406 } 1407} 1408 1409AudioFlinger::PlaybackThread::~PlaybackThread() 1410{ 1411 delete [] mMixBuffer; 1412} 1413 1414status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1415{ 1416 dumpInternals(fd, args); 1417 dumpTracks(fd, args); 1418 dumpEffectChains(fd, args); 1419 return NO_ERROR; 1420} 1421 1422status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1423{ 1424 const size_t SIZE = 256; 1425 char buffer[SIZE]; 1426 String8 result; 1427 1428 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1429 result.append(buffer); 1430 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1431 for (size_t i = 0; i < mTracks.size(); ++i) { 1432 sp<Track> track = mTracks[i]; 1433 if (track != 0) { 1434 track->dump(buffer, SIZE); 1435 result.append(buffer); 1436 } 1437 } 1438 1439 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1440 result.append(buffer); 1441 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1442 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1443 sp<Track> track = mActiveTracks[i].promote(); 1444 if (track != 0) { 1445 track->dump(buffer, SIZE); 1446 result.append(buffer); 1447 } 1448 } 1449 write(fd, result.string(), result.size()); 1450 return NO_ERROR; 1451} 1452 1453status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1454{ 1455 const size_t SIZE = 256; 1456 char buffer[SIZE]; 1457 String8 result; 1458 1459 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1460 result.append(buffer); 1461 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1462 result.append(buffer); 1463 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1464 result.append(buffer); 1465 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1466 result.append(buffer); 1467 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1468 result.append(buffer); 1469 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1470 result.append(buffer); 1471 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1472 result.append(buffer); 1473 write(fd, result.string(), result.size()); 1474 1475 dumpBase(fd, args); 1476 1477 return NO_ERROR; 1478} 1479 1480// Thread virtuals 1481status_t AudioFlinger::PlaybackThread::readyToRun() 1482{ 1483 status_t status = initCheck(); 1484 if (status == NO_ERROR) { 1485 ALOGI("AudioFlinger's thread %p ready to run", this); 1486 } else { 1487 ALOGE("No working audio driver found."); 1488 } 1489 return status; 1490} 1491 1492void AudioFlinger::PlaybackThread::onFirstRef() 1493{ 1494 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1495} 1496 1497// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1498sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1499 const sp<AudioFlinger::Client>& client, 1500 audio_stream_type_t streamType, 1501 uint32_t sampleRate, 1502 audio_format_t format, 1503 uint32_t channelMask, 1504 int frameCount, 1505 const sp<IMemory>& sharedBuffer, 1506 int sessionId, 1507 status_t *status) 1508{ 1509 sp<Track> track; 1510 status_t lStatus; 1511 1512 if (mType == DIRECT) { 1513 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1514 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1515 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1516 "for output %p with format %d", 1517 sampleRate, format, channelMask, mOutput, mFormat); 1518 lStatus = BAD_VALUE; 1519 goto Exit; 1520 } 1521 } 1522 } else { 1523 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1524 if (sampleRate > mSampleRate*2) { 1525 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1526 lStatus = BAD_VALUE; 1527 goto Exit; 1528 } 1529 } 1530 1531 lStatus = initCheck(); 1532 if (lStatus != NO_ERROR) { 1533 ALOGE("Audio driver not initialized."); 1534 goto Exit; 1535 } 1536 1537 { // scope for mLock 1538 Mutex::Autolock _l(mLock); 1539 1540 // all tracks in same audio session must share the same routing strategy otherwise 1541 // conflicts will happen when tracks are moved from one output to another by audio policy 1542 // manager 1543 uint32_t strategy = 1544 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1545 for (size_t i = 0; i < mTracks.size(); ++i) { 1546 sp<Track> t = mTracks[i]; 1547 if (t != 0) { 1548 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1549 if (sessionId == t->sessionId() && strategy != actual) { 1550 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1551 strategy, actual); 1552 lStatus = BAD_VALUE; 1553 goto Exit; 1554 } 1555 } 1556 } 1557 1558 track = new Track(this, client, streamType, sampleRate, format, 1559 channelMask, frameCount, sharedBuffer, sessionId); 1560 if (track->getCblk() == NULL || track->name() < 0) { 1561 lStatus = NO_MEMORY; 1562 goto Exit; 1563 } 1564 mTracks.add(track); 1565 1566 sp<EffectChain> chain = getEffectChain_l(sessionId); 1567 if (chain != 0) { 1568 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1569 track->setMainBuffer(chain->inBuffer()); 1570 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1571 chain->incTrackCnt(); 1572 } 1573 1574 // invalidate track immediately if the stream type was moved to another thread since 1575 // createTrack() was called by the client process. 1576 if (!mStreamTypes[streamType].valid) { 1577 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1578 this, streamType); 1579 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1580 } 1581 } 1582 lStatus = NO_ERROR; 1583 1584Exit: 1585 if(status) { 1586 *status = lStatus; 1587 } 1588 return track; 1589} 1590 1591uint32_t AudioFlinger::PlaybackThread::latency() const 1592{ 1593 Mutex::Autolock _l(mLock); 1594 if (initCheck() == NO_ERROR) { 1595 return mOutput->stream->get_latency(mOutput->stream); 1596 } else { 1597 return 0; 1598 } 1599} 1600 1601status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1602{ 1603 mMasterVolume = value; 1604 return NO_ERROR; 1605} 1606 1607status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1608{ 1609 mMasterMute = muted; 1610 return NO_ERROR; 1611} 1612 1613float AudioFlinger::PlaybackThread::masterVolume() const 1614{ 1615 return mMasterVolume; 1616} 1617 1618bool AudioFlinger::PlaybackThread::masterMute() const 1619{ 1620 return mMasterMute; 1621} 1622 1623status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1624{ 1625 mStreamTypes[stream].volume = value; 1626 return NO_ERROR; 1627} 1628 1629status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1630{ 1631 mStreamTypes[stream].mute = muted; 1632 return NO_ERROR; 1633} 1634 1635float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1636{ 1637 return mStreamTypes[stream].volume; 1638} 1639 1640bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1641{ 1642 return mStreamTypes[stream].mute; 1643} 1644 1645// addTrack_l() must be called with ThreadBase::mLock held 1646status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1647{ 1648 status_t status = ALREADY_EXISTS; 1649 1650 // set retry count for buffer fill 1651 track->mRetryCount = kMaxTrackStartupRetries; 1652 if (mActiveTracks.indexOf(track) < 0) { 1653 // the track is newly added, make sure it fills up all its 1654 // buffers before playing. This is to ensure the client will 1655 // effectively get the latency it requested. 1656 track->mFillingUpStatus = Track::FS_FILLING; 1657 track->mResetDone = false; 1658 mActiveTracks.add(track); 1659 if (track->mainBuffer() != mMixBuffer) { 1660 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1661 if (chain != 0) { 1662 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1663 chain->incActiveTrackCnt(); 1664 } 1665 } 1666 1667 status = NO_ERROR; 1668 } 1669 1670 ALOGV("mWaitWorkCV.broadcast"); 1671 mWaitWorkCV.broadcast(); 1672 1673 return status; 1674} 1675 1676// destroyTrack_l() must be called with ThreadBase::mLock held 1677void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1678{ 1679 track->mState = TrackBase::TERMINATED; 1680 if (mActiveTracks.indexOf(track) < 0) { 1681 removeTrack_l(track); 1682 } 1683} 1684 1685void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1686{ 1687 mTracks.remove(track); 1688 deleteTrackName_l(track->name()); 1689 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1690 if (chain != 0) { 1691 chain->decTrackCnt(); 1692 } 1693} 1694 1695String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1696{ 1697 String8 out_s8 = String8(""); 1698 char *s; 1699 1700 Mutex::Autolock _l(mLock); 1701 if (initCheck() != NO_ERROR) { 1702 return out_s8; 1703 } 1704 1705 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1706 out_s8 = String8(s); 1707 free(s); 1708 return out_s8; 1709} 1710 1711// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1712void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1713 AudioSystem::OutputDescriptor desc; 1714 void *param2 = NULL; 1715 1716 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1717 1718 switch (event) { 1719 case AudioSystem::OUTPUT_OPENED: 1720 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1721 desc.channels = mChannelMask; 1722 desc.samplingRate = mSampleRate; 1723 desc.format = mFormat; 1724 desc.frameCount = mFrameCount; 1725 desc.latency = latency(); 1726 param2 = &desc; 1727 break; 1728 1729 case AudioSystem::STREAM_CONFIG_CHANGED: 1730 param2 = ¶m; 1731 case AudioSystem::OUTPUT_CLOSED: 1732 default: 1733 break; 1734 } 1735 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1736} 1737 1738void AudioFlinger::PlaybackThread::readOutputParameters() 1739{ 1740 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1741 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1742 mChannelCount = (uint16_t)popcount(mChannelMask); 1743 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1744 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1745 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1746 1747 // FIXME - Current mixer implementation only supports stereo output: Always 1748 // Allocate a stereo buffer even if HW output is mono. 1749 delete[] mMixBuffer; 1750 mMixBuffer = new int16_t[mFrameCount * 2]; 1751 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1752 1753 // force reconfiguration of effect chains and engines to take new buffer size and audio 1754 // parameters into account 1755 // Note that mLock is not held when readOutputParameters() is called from the constructor 1756 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1757 // matter. 1758 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1759 Vector< sp<EffectChain> > effectChains = mEffectChains; 1760 for (size_t i = 0; i < effectChains.size(); i ++) { 1761 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1762 } 1763} 1764 1765status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1766{ 1767 if (halFrames == NULL || dspFrames == NULL) { 1768 return BAD_VALUE; 1769 } 1770 Mutex::Autolock _l(mLock); 1771 if (initCheck() != NO_ERROR) { 1772 return INVALID_OPERATION; 1773 } 1774 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1775 1776 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1777} 1778 1779uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1780{ 1781 Mutex::Autolock _l(mLock); 1782 uint32_t result = 0; 1783 if (getEffectChain_l(sessionId) != 0) { 1784 result = EFFECT_SESSION; 1785 } 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 result |= TRACK_SESSION; 1792 break; 1793 } 1794 } 1795 1796 return result; 1797} 1798 1799uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1800{ 1801 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1802 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1803 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1804 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1805 } 1806 for (size_t i = 0; i < mTracks.size(); i++) { 1807 sp<Track> track = mTracks[i]; 1808 if (sessionId == track->sessionId() && 1809 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1810 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1811 } 1812 } 1813 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1814} 1815 1816 1817AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1818{ 1819 Mutex::Autolock _l(mLock); 1820 return mOutput; 1821} 1822 1823AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1824{ 1825 Mutex::Autolock _l(mLock); 1826 AudioStreamOut *output = mOutput; 1827 mOutput = NULL; 1828 return output; 1829} 1830 1831// this method must always be called either with ThreadBase mLock held or inside the thread loop 1832audio_stream_t* AudioFlinger::PlaybackThread::stream() 1833{ 1834 if (mOutput == NULL) { 1835 return NULL; 1836 } 1837 return &mOutput->stream->common; 1838} 1839 1840uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1841{ 1842 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1843 // decoding and transfer time. So sleeping for half of the latency would likely cause 1844 // underruns 1845 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1846 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1847 } else { 1848 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1849 } 1850} 1851 1852// ---------------------------------------------------------------------------- 1853 1854AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1855 int id, uint32_t device, type_t type) 1856 : PlaybackThread(audioFlinger, output, id, device, type), 1857 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1858 mPrevMixerStatus(MIXER_IDLE) 1859{ 1860 // FIXME - Current mixer implementation only supports stereo output 1861 if (mChannelCount == 1) { 1862 ALOGE("Invalid audio hardware channel count"); 1863 } 1864} 1865 1866AudioFlinger::MixerThread::~MixerThread() 1867{ 1868 delete mAudioMixer; 1869} 1870 1871bool AudioFlinger::MixerThread::threadLoop() 1872{ 1873 Vector< sp<Track> > tracksToRemove; 1874 mixer_state mixerStatus = MIXER_IDLE; 1875 nsecs_t standbyTime = systemTime(); 1876 size_t mixBufferSize = mFrameCount * mFrameSize; 1877 // FIXME: Relaxed timing because of a certain device that can't meet latency 1878 // Should be reduced to 2x after the vendor fixes the driver issue 1879 // increase threshold again due to low power audio mode. The way this warning threshold is 1880 // calculated and its usefulness should be reconsidered anyway. 1881 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1882 nsecs_t lastWarning = 0; 1883 bool longStandbyExit = false; 1884 uint32_t activeSleepTime = activeSleepTimeUs(); 1885 uint32_t idleSleepTime = idleSleepTimeUs(); 1886 uint32_t sleepTime = idleSleepTime; 1887 uint32_t sleepTimeShift = 0; 1888 Vector< sp<EffectChain> > effectChains; 1889#ifdef DEBUG_CPU_USAGE 1890 ThreadCpuUsage cpu; 1891 const CentralTendencyStatistics& stats = cpu.statistics(); 1892#endif 1893 1894 acquireWakeLock(); 1895 1896 while (!exitPending()) 1897 { 1898#ifdef DEBUG_CPU_USAGE 1899 cpu.sampleAndEnable(); 1900 unsigned n = stats.n(); 1901 // cpu.elapsed() is expensive, so don't call it every loop 1902 if ((n & 127) == 1) { 1903 long long elapsed = cpu.elapsed(); 1904 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1905 double perLoop = elapsed / (double) n; 1906 double perLoop100 = perLoop * 0.01; 1907 double mean = stats.mean(); 1908 double stddev = stats.stddev(); 1909 double minimum = stats.minimum(); 1910 double maximum = stats.maximum(); 1911 cpu.resetStatistics(); 1912 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1913 elapsed * .000000001, n, perLoop * .000001, 1914 mean * .001, 1915 stddev * .001, 1916 minimum * .001, 1917 maximum * .001, 1918 mean / perLoop100, 1919 stddev / perLoop100, 1920 minimum / perLoop100, 1921 maximum / perLoop100); 1922 } 1923 } 1924#endif 1925 processConfigEvents(); 1926 1927 mixerStatus = MIXER_IDLE; 1928 { // scope for mLock 1929 1930 Mutex::Autolock _l(mLock); 1931 1932 if (checkForNewParameters_l()) { 1933 mixBufferSize = mFrameCount * mFrameSize; 1934 // FIXME: Relaxed timing because of a certain device that can't meet latency 1935 // Should be reduced to 2x after the vendor fixes the driver issue 1936 // increase threshold again due to low power audio mode. The way this warning 1937 // threshold is calculated and its usefulness should be reconsidered anyway. 1938 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1939 activeSleepTime = activeSleepTimeUs(); 1940 idleSleepTime = idleSleepTimeUs(); 1941 } 1942 1943 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1944 1945 // put audio hardware into standby after short delay 1946 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1947 mSuspended)) { 1948 if (!mStandby) { 1949 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1950 mOutput->stream->common.standby(&mOutput->stream->common); 1951 mStandby = true; 1952 mBytesWritten = 0; 1953 } 1954 1955 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1956 // we're about to wait, flush the binder command buffer 1957 IPCThreadState::self()->flushCommands(); 1958 1959 if (exitPending()) break; 1960 1961 releaseWakeLock_l(); 1962 // wait until we have something to do... 1963 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1964 mWaitWorkCV.wait(mLock); 1965 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1966 acquireWakeLock_l(); 1967 1968 mPrevMixerStatus = MIXER_IDLE; 1969 if (!mMasterMute) { 1970 char value[PROPERTY_VALUE_MAX]; 1971 property_get("ro.audio.silent", value, "0"); 1972 if (atoi(value)) { 1973 ALOGD("Silence is golden"); 1974 setMasterMute(true); 1975 } 1976 } 1977 1978 standbyTime = systemTime() + kStandbyTimeInNsecs; 1979 sleepTime = idleSleepTime; 1980 sleepTimeShift = 0; 1981 continue; 1982 } 1983 } 1984 1985 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1986 1987 // prevent any changes in effect chain list and in each effect chain 1988 // during mixing and effect process as the audio buffers could be deleted 1989 // or modified if an effect is created or deleted 1990 lockEffectChains_l(effectChains); 1991 } 1992 1993 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1994 // mix buffers... 1995 mAudioMixer->process(); 1996 // increase sleep time progressively when application underrun condition clears. 1997 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1998 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1999 // such that we would underrun the audio HAL. 2000 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2001 sleepTimeShift--; 2002 } 2003 sleepTime = 0; 2004 standbyTime = systemTime() + kStandbyTimeInNsecs; 2005 //TODO: delay standby when effects have a tail 2006 } else { 2007 // If no tracks are ready, sleep once for the duration of an output 2008 // buffer size, then write 0s to the output 2009 if (sleepTime == 0) { 2010 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2011 sleepTime = activeSleepTime >> sleepTimeShift; 2012 if (sleepTime < kMinThreadSleepTimeUs) { 2013 sleepTime = kMinThreadSleepTimeUs; 2014 } 2015 // reduce sleep time in case of consecutive application underruns to avoid 2016 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2017 // duration we would end up writing less data than needed by the audio HAL if 2018 // the condition persists. 2019 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2020 sleepTimeShift++; 2021 } 2022 } else { 2023 sleepTime = idleSleepTime; 2024 } 2025 } else if (mBytesWritten != 0 || 2026 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2027 memset (mMixBuffer, 0, mixBufferSize); 2028 sleepTime = 0; 2029 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2030 } 2031 // TODO add standby time extension fct of effect tail 2032 } 2033 2034 if (mSuspended) { 2035 sleepTime = suspendSleepTimeUs(); 2036 } 2037 // sleepTime == 0 means we must write to audio hardware 2038 if (sleepTime == 0) { 2039 for (size_t i = 0; i < effectChains.size(); i ++) { 2040 effectChains[i]->process_l(); 2041 } 2042 // enable changes in effect chain 2043 unlockEffectChains(effectChains); 2044 mLastWriteTime = systemTime(); 2045 mInWrite = true; 2046 mBytesWritten += mixBufferSize; 2047 2048 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2049 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2050 mNumWrites++; 2051 mInWrite = false; 2052 nsecs_t now = systemTime(); 2053 nsecs_t delta = now - mLastWriteTime; 2054 if (!mStandby && delta > maxPeriod) { 2055 mNumDelayedWrites++; 2056 if ((now - lastWarning) > kWarningThrottleNs) { 2057 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2058 ns2ms(delta), mNumDelayedWrites, this); 2059 lastWarning = now; 2060 } 2061 if (mStandby) { 2062 longStandbyExit = true; 2063 } 2064 } 2065 mStandby = false; 2066 } else { 2067 // enable changes in effect chain 2068 unlockEffectChains(effectChains); 2069 usleep(sleepTime); 2070 } 2071 2072 // finally let go of all our tracks, without the lock held 2073 // since we can't guarantee the destructors won't acquire that 2074 // same lock. 2075 tracksToRemove.clear(); 2076 2077 // Effect chains will be actually deleted here if they were removed from 2078 // mEffectChains list during mixing or effects processing 2079 effectChains.clear(); 2080 } 2081 2082 if (!mStandby) { 2083 mOutput->stream->common.standby(&mOutput->stream->common); 2084 } 2085 2086 releaseWakeLock(); 2087 2088 ALOGV("MixerThread %p exiting", this); 2089 return false; 2090} 2091 2092// prepareTracks_l() must be called with ThreadBase::mLock held 2093AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2094 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2095{ 2096 2097 mixer_state mixerStatus = MIXER_IDLE; 2098 // find out which tracks need to be processed 2099 size_t count = activeTracks.size(); 2100 size_t mixedTracks = 0; 2101 size_t tracksWithEffect = 0; 2102 2103 float masterVolume = mMasterVolume; 2104 bool masterMute = mMasterMute; 2105 2106 if (masterMute) { 2107 masterVolume = 0; 2108 } 2109 // Delegate master volume control to effect in output mix effect chain if needed 2110 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2111 if (chain != 0) { 2112 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2113 chain->setVolume_l(&v, &v); 2114 masterVolume = (float)((v + (1 << 23)) >> 24); 2115 chain.clear(); 2116 } 2117 2118 for (size_t i=0 ; i<count ; i++) { 2119 sp<Track> t = activeTracks[i].promote(); 2120 if (t == 0) continue; 2121 2122 // this const just means the local variable doesn't change 2123 Track* const track = t.get(); 2124 audio_track_cblk_t* cblk = track->cblk(); 2125 2126 // The first time a track is added we wait 2127 // for all its buffers to be filled before processing it 2128 int name = track->name(); 2129 // make sure that we have enough frames to mix one full buffer. 2130 // enforce this condition only once to enable draining the buffer in case the client 2131 // app does not call stop() and relies on underrun to stop: 2132 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2133 // during last round 2134 uint32_t minFrames = 1; 2135 if (!track->isStopped() && !track->isPausing() && 2136 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2137 if (t->sampleRate() == (int)mSampleRate) { 2138 minFrames = mFrameCount; 2139 } else { 2140 // +1 for rounding and +1 for additional sample needed for interpolation 2141 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2142 // add frames already consumed but not yet released by the resampler 2143 // because cblk->framesReady() will include these frames 2144 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2145 // the minimum track buffer size is normally twice the number of frames necessary 2146 // to fill one buffer and the resampler should not leave more than one buffer worth 2147 // of unreleased frames after each pass, but just in case... 2148 ALOG_ASSERT(minFrames <= cblk->frameCount); 2149 } 2150 } 2151 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2152 !track->isPaused() && !track->isTerminated()) 2153 { 2154 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2155 2156 mixedTracks++; 2157 2158 // track->mainBuffer() != mMixBuffer means there is an effect chain 2159 // connected to the track 2160 chain.clear(); 2161 if (track->mainBuffer() != mMixBuffer) { 2162 chain = getEffectChain_l(track->sessionId()); 2163 // Delegate volume control to effect in track effect chain if needed 2164 if (chain != 0) { 2165 tracksWithEffect++; 2166 } else { 2167 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2168 name, track->sessionId()); 2169 } 2170 } 2171 2172 2173 int param = AudioMixer::VOLUME; 2174 if (track->mFillingUpStatus == Track::FS_FILLED) { 2175 // no ramp for the first volume setting 2176 track->mFillingUpStatus = Track::FS_ACTIVE; 2177 if (track->mState == TrackBase::RESUMING) { 2178 track->mState = TrackBase::ACTIVE; 2179 param = AudioMixer::RAMP_VOLUME; 2180 } 2181 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2182 } else if (cblk->server != 0) { 2183 // If the track is stopped before the first frame was mixed, 2184 // do not apply ramp 2185 param = AudioMixer::RAMP_VOLUME; 2186 } 2187 2188 // compute volume for this track 2189 uint32_t vl, vr, va; 2190 if (track->isMuted() || track->isPausing() || 2191 mStreamTypes[track->type()].mute) { 2192 vl = vr = va = 0; 2193 if (track->isPausing()) { 2194 track->setPaused(); 2195 } 2196 } else { 2197 2198 // read original volumes with volume control 2199 float typeVolume = mStreamTypes[track->type()].volume; 2200 float v = masterVolume * typeVolume; 2201 uint32_t vlr = cblk->getVolumeLR(); 2202 vl = vlr & 0xFFFF; 2203 vr = vlr >> 16; 2204 // track volumes come from shared memory, so can't be trusted and must be clamped 2205 if (vl > MAX_GAIN_INT) { 2206 ALOGV("Track left volume out of range: %04X", vl); 2207 vl = MAX_GAIN_INT; 2208 } 2209 if (vr > MAX_GAIN_INT) { 2210 ALOGV("Track right volume out of range: %04X", vr); 2211 vr = MAX_GAIN_INT; 2212 } 2213 // now apply the master volume and stream type volume 2214 vl = (uint32_t)(v * vl) << 12; 2215 vr = (uint32_t)(v * vr) << 12; 2216 // assuming master volume and stream type volume each go up to 1.0, 2217 // vl and vr are now in 8.24 format 2218 2219 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2220 // send level comes from shared memory and so may be corrupt 2221 if (sendLevel >= MAX_GAIN_INT) { 2222 ALOGV("Track send level out of range: %04X", sendLevel); 2223 sendLevel = MAX_GAIN_INT; 2224 } 2225 va = (uint32_t)(v * sendLevel); 2226 } 2227 // Delegate volume control to effect in track effect chain if needed 2228 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2229 // Do not ramp volume if volume is controlled by effect 2230 param = AudioMixer::VOLUME; 2231 track->mHasVolumeController = true; 2232 } else { 2233 // force no volume ramp when volume controller was just disabled or removed 2234 // from effect chain to avoid volume spike 2235 if (track->mHasVolumeController) { 2236 param = AudioMixer::VOLUME; 2237 } 2238 track->mHasVolumeController = false; 2239 } 2240 2241 // Convert volumes from 8.24 to 4.12 format 2242 int16_t left, right, aux; 2243 // This additional clamping is needed in case chain->setVolume_l() overshot 2244 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2245 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2246 left = int16_t(v_clamped); 2247 v_clamped = (vr + (1 << 11)) >> 12; 2248 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2249 right = int16_t(v_clamped); 2250 2251 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2252 aux = int16_t(va); 2253 2254 // XXX: these things DON'T need to be done each time 2255 mAudioMixer->setBufferProvider(name, track); 2256 mAudioMixer->enable(name); 2257 2258 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2259 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2260 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2261 mAudioMixer->setParameter( 2262 name, 2263 AudioMixer::TRACK, 2264 AudioMixer::FORMAT, (void *)track->format()); 2265 mAudioMixer->setParameter( 2266 name, 2267 AudioMixer::TRACK, 2268 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2269 mAudioMixer->setParameter( 2270 name, 2271 AudioMixer::RESAMPLE, 2272 AudioMixer::SAMPLE_RATE, 2273 (void *)(cblk->sampleRate)); 2274 mAudioMixer->setParameter( 2275 name, 2276 AudioMixer::TRACK, 2277 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2278 mAudioMixer->setParameter( 2279 name, 2280 AudioMixer::TRACK, 2281 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2282 2283 // reset retry count 2284 track->mRetryCount = kMaxTrackRetries; 2285 // If one track is ready, set the mixer ready if: 2286 // - the mixer was not ready during previous round OR 2287 // - no other track is not ready 2288 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2289 mixerStatus != MIXER_TRACKS_ENABLED) { 2290 mixerStatus = MIXER_TRACKS_READY; 2291 } 2292 } else { 2293 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2294 if (track->isStopped()) { 2295 track->reset(); 2296 } 2297 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2298 // We have consumed all the buffers of this track. 2299 // Remove it from the list of active tracks. 2300 tracksToRemove->add(track); 2301 } else { 2302 // No buffers for this track. Give it a few chances to 2303 // fill a buffer, then remove it from active list. 2304 if (--(track->mRetryCount) <= 0) { 2305 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2306 tracksToRemove->add(track); 2307 // indicate to client process that the track was disabled because of underrun 2308 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2309 // If one track is not ready, mark the mixer also not ready if: 2310 // - the mixer was ready during previous round OR 2311 // - no other track is ready 2312 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2313 mixerStatus != MIXER_TRACKS_READY) { 2314 mixerStatus = MIXER_TRACKS_ENABLED; 2315 } 2316 } 2317 mAudioMixer->disable(name); 2318 } 2319 } 2320 2321 // remove all the tracks that need to be... 2322 count = tracksToRemove->size(); 2323 if (CC_UNLIKELY(count)) { 2324 for (size_t i=0 ; i<count ; i++) { 2325 const sp<Track>& track = tracksToRemove->itemAt(i); 2326 mActiveTracks.remove(track); 2327 if (track->mainBuffer() != mMixBuffer) { 2328 chain = getEffectChain_l(track->sessionId()); 2329 if (chain != 0) { 2330 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2331 chain->decActiveTrackCnt(); 2332 } 2333 } 2334 if (track->isTerminated()) { 2335 removeTrack_l(track); 2336 } 2337 } 2338 } 2339 2340 // mix buffer must be cleared if all tracks are connected to an 2341 // effect chain as in this case the mixer will not write to 2342 // mix buffer and track effects will accumulate into it 2343 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2344 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2345 } 2346 2347 mPrevMixerStatus = mixerStatus; 2348 return mixerStatus; 2349} 2350 2351void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2352{ 2353 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2354 this, streamType, mTracks.size()); 2355 Mutex::Autolock _l(mLock); 2356 2357 size_t size = mTracks.size(); 2358 for (size_t i = 0; i < size; i++) { 2359 sp<Track> t = mTracks[i]; 2360 if (t->type() == streamType) { 2361 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2362 t->mCblk->cv.signal(); 2363 } 2364 } 2365} 2366 2367void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2368{ 2369 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2370 this, streamType, valid); 2371 Mutex::Autolock _l(mLock); 2372 2373 mStreamTypes[streamType].valid = valid; 2374} 2375 2376// getTrackName_l() must be called with ThreadBase::mLock held 2377int AudioFlinger::MixerThread::getTrackName_l() 2378{ 2379 return mAudioMixer->getTrackName(); 2380} 2381 2382// deleteTrackName_l() must be called with ThreadBase::mLock held 2383void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2384{ 2385 ALOGV("remove track (%d) and delete from mixer", name); 2386 mAudioMixer->deleteTrackName(name); 2387} 2388 2389// checkForNewParameters_l() must be called with ThreadBase::mLock held 2390bool AudioFlinger::MixerThread::checkForNewParameters_l() 2391{ 2392 bool reconfig = false; 2393 2394 while (!mNewParameters.isEmpty()) { 2395 status_t status = NO_ERROR; 2396 String8 keyValuePair = mNewParameters[0]; 2397 AudioParameter param = AudioParameter(keyValuePair); 2398 int value; 2399 2400 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2401 reconfig = true; 2402 } 2403 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2404 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2405 status = BAD_VALUE; 2406 } else { 2407 reconfig = true; 2408 } 2409 } 2410 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2411 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2412 status = BAD_VALUE; 2413 } else { 2414 reconfig = true; 2415 } 2416 } 2417 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2418 // do not accept frame count changes if tracks are open as the track buffer 2419 // size depends on frame count and correct behavior would not be guaranteed 2420 // if frame count is changed after track creation 2421 if (!mTracks.isEmpty()) { 2422 status = INVALID_OPERATION; 2423 } else { 2424 reconfig = true; 2425 } 2426 } 2427 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2428 // when changing the audio output device, call addBatteryData to notify 2429 // the change 2430 if ((int)mDevice != value) { 2431 uint32_t params = 0; 2432 // check whether speaker is on 2433 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2434 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2435 } 2436 2437 int deviceWithoutSpeaker 2438 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2439 // check if any other device (except speaker) is on 2440 if (value & deviceWithoutSpeaker ) { 2441 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2442 } 2443 2444 if (params != 0) { 2445 addBatteryData(params); 2446 } 2447 } 2448 2449 // forward device change to effects that have requested to be 2450 // aware of attached audio device. 2451 mDevice = (uint32_t)value; 2452 for (size_t i = 0; i < mEffectChains.size(); i++) { 2453 mEffectChains[i]->setDevice_l(mDevice); 2454 } 2455 } 2456 2457 if (status == NO_ERROR) { 2458 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2459 keyValuePair.string()); 2460 if (!mStandby && status == INVALID_OPERATION) { 2461 mOutput->stream->common.standby(&mOutput->stream->common); 2462 mStandby = true; 2463 mBytesWritten = 0; 2464 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2465 keyValuePair.string()); 2466 } 2467 if (status == NO_ERROR && reconfig) { 2468 delete mAudioMixer; 2469 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2470 mAudioMixer = NULL; 2471 readOutputParameters(); 2472 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2473 for (size_t i = 0; i < mTracks.size() ; i++) { 2474 int name = getTrackName_l(); 2475 if (name < 0) break; 2476 mTracks[i]->mName = name; 2477 // limit track sample rate to 2 x new output sample rate 2478 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2479 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2480 } 2481 } 2482 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2483 } 2484 } 2485 2486 mNewParameters.removeAt(0); 2487 2488 mParamStatus = status; 2489 mParamCond.signal(); 2490 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2491 // already timed out waiting for the status and will never signal the condition. 2492 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2493 } 2494 return reconfig; 2495} 2496 2497status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2498{ 2499 const size_t SIZE = 256; 2500 char buffer[SIZE]; 2501 String8 result; 2502 2503 PlaybackThread::dumpInternals(fd, args); 2504 2505 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2506 result.append(buffer); 2507 write(fd, result.string(), result.size()); 2508 return NO_ERROR; 2509} 2510 2511uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2512{ 2513 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2514} 2515 2516uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2517{ 2518 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2519} 2520 2521// ---------------------------------------------------------------------------- 2522AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2523 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2524 // mLeftVolFloat, mRightVolFloat 2525 // mLeftVolShort, mRightVolShort 2526{ 2527} 2528 2529AudioFlinger::DirectOutputThread::~DirectOutputThread() 2530{ 2531} 2532 2533static inline 2534int32_t mul(int16_t in, int16_t v) 2535{ 2536#if defined(__arm__) && !defined(__thumb__) 2537 int32_t out; 2538 asm( "smulbb %[out], %[in], %[v] \n" 2539 : [out]"=r"(out) 2540 : [in]"%r"(in), [v]"r"(v) 2541 : ); 2542 return out; 2543#else 2544 return in * int32_t(v); 2545#endif 2546} 2547 2548void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2549{ 2550 // Do not apply volume on compressed audio 2551 if (!audio_is_linear_pcm(mFormat)) { 2552 return; 2553 } 2554 2555 // convert to signed 16 bit before volume calculation 2556 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2557 size_t count = mFrameCount * mChannelCount; 2558 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2559 int16_t *dst = mMixBuffer + count-1; 2560 while(count--) { 2561 *dst-- = (int16_t)(*src--^0x80) << 8; 2562 } 2563 } 2564 2565 size_t frameCount = mFrameCount; 2566 int16_t *out = mMixBuffer; 2567 if (ramp) { 2568 if (mChannelCount == 1) { 2569 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2570 int32_t vlInc = d / (int32_t)frameCount; 2571 int32_t vl = ((int32_t)mLeftVolShort << 16); 2572 do { 2573 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2574 out++; 2575 vl += vlInc; 2576 } while (--frameCount); 2577 2578 } else { 2579 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2580 int32_t vlInc = d / (int32_t)frameCount; 2581 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2582 int32_t vrInc = d / (int32_t)frameCount; 2583 int32_t vl = ((int32_t)mLeftVolShort << 16); 2584 int32_t vr = ((int32_t)mRightVolShort << 16); 2585 do { 2586 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2587 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2588 out += 2; 2589 vl += vlInc; 2590 vr += vrInc; 2591 } while (--frameCount); 2592 } 2593 } else { 2594 if (mChannelCount == 1) { 2595 do { 2596 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2597 out++; 2598 } while (--frameCount); 2599 } else { 2600 do { 2601 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2602 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2603 out += 2; 2604 } while (--frameCount); 2605 } 2606 } 2607 2608 // convert back to unsigned 8 bit after volume calculation 2609 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2610 size_t count = mFrameCount * mChannelCount; 2611 int16_t *src = mMixBuffer; 2612 uint8_t *dst = (uint8_t *)mMixBuffer; 2613 while(count--) { 2614 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2615 } 2616 } 2617 2618 mLeftVolShort = leftVol; 2619 mRightVolShort = rightVol; 2620} 2621 2622bool AudioFlinger::DirectOutputThread::threadLoop() 2623{ 2624 mixer_state mixerStatus = MIXER_IDLE; 2625 sp<Track> trackToRemove; 2626 sp<Track> activeTrack; 2627 nsecs_t standbyTime = systemTime(); 2628 int8_t *curBuf; 2629 size_t mixBufferSize = mFrameCount*mFrameSize; 2630 uint32_t activeSleepTime = activeSleepTimeUs(); 2631 uint32_t idleSleepTime = idleSleepTimeUs(); 2632 uint32_t sleepTime = idleSleepTime; 2633 // use shorter standby delay as on normal output to release 2634 // hardware resources as soon as possible 2635 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2636 2637 acquireWakeLock(); 2638 2639 while (!exitPending()) 2640 { 2641 bool rampVolume; 2642 uint16_t leftVol; 2643 uint16_t rightVol; 2644 Vector< sp<EffectChain> > effectChains; 2645 2646 processConfigEvents(); 2647 2648 mixerStatus = MIXER_IDLE; 2649 2650 { // scope for the mLock 2651 2652 Mutex::Autolock _l(mLock); 2653 2654 if (checkForNewParameters_l()) { 2655 mixBufferSize = mFrameCount*mFrameSize; 2656 activeSleepTime = activeSleepTimeUs(); 2657 idleSleepTime = idleSleepTimeUs(); 2658 standbyDelay = microseconds(activeSleepTime*2); 2659 } 2660 2661 // put audio hardware into standby after short delay 2662 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2663 mSuspended)) { 2664 // wait until we have something to do... 2665 if (!mStandby) { 2666 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2667 mOutput->stream->common.standby(&mOutput->stream->common); 2668 mStandby = true; 2669 mBytesWritten = 0; 2670 } 2671 2672 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2673 // we're about to wait, flush the binder command buffer 2674 IPCThreadState::self()->flushCommands(); 2675 2676 if (exitPending()) break; 2677 2678 releaseWakeLock_l(); 2679 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2680 mWaitWorkCV.wait(mLock); 2681 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2682 acquireWakeLock_l(); 2683 2684 if (!mMasterMute) { 2685 char value[PROPERTY_VALUE_MAX]; 2686 property_get("ro.audio.silent", value, "0"); 2687 if (atoi(value)) { 2688 ALOGD("Silence is golden"); 2689 setMasterMute(true); 2690 } 2691 } 2692 2693 standbyTime = systemTime() + standbyDelay; 2694 sleepTime = idleSleepTime; 2695 continue; 2696 } 2697 } 2698 2699 effectChains = mEffectChains; 2700 2701 // find out which tracks need to be processed 2702 if (mActiveTracks.size() != 0) { 2703 sp<Track> t = mActiveTracks[0].promote(); 2704 if (t == 0) continue; 2705 2706 Track* const track = t.get(); 2707 audio_track_cblk_t* cblk = track->cblk(); 2708 2709 // The first time a track is added we wait 2710 // for all its buffers to be filled before processing it 2711 if (cblk->framesReady() && track->isReady() && 2712 !track->isPaused() && !track->isTerminated()) 2713 { 2714 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2715 2716 if (track->mFillingUpStatus == Track::FS_FILLED) { 2717 track->mFillingUpStatus = Track::FS_ACTIVE; 2718 mLeftVolFloat = mRightVolFloat = 0; 2719 mLeftVolShort = mRightVolShort = 0; 2720 if (track->mState == TrackBase::RESUMING) { 2721 track->mState = TrackBase::ACTIVE; 2722 rampVolume = true; 2723 } 2724 } else if (cblk->server != 0) { 2725 // If the track is stopped before the first frame was mixed, 2726 // do not apply ramp 2727 rampVolume = true; 2728 } 2729 // compute volume for this track 2730 float left, right; 2731 if (track->isMuted() || mMasterMute || track->isPausing() || 2732 mStreamTypes[track->type()].mute) { 2733 left = right = 0; 2734 if (track->isPausing()) { 2735 track->setPaused(); 2736 } 2737 } else { 2738 float typeVolume = mStreamTypes[track->type()].volume; 2739 float v = mMasterVolume * typeVolume; 2740 uint32_t vlr = cblk->getVolumeLR(); 2741 float v_clamped = v * (vlr & 0xFFFF); 2742 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2743 left = v_clamped/MAX_GAIN; 2744 v_clamped = v * (vlr >> 16); 2745 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2746 right = v_clamped/MAX_GAIN; 2747 } 2748 2749 if (left != mLeftVolFloat || right != mRightVolFloat) { 2750 mLeftVolFloat = left; 2751 mRightVolFloat = right; 2752 2753 // If audio HAL implements volume control, 2754 // force software volume to nominal value 2755 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2756 left = 1.0f; 2757 right = 1.0f; 2758 } 2759 2760 // Convert volumes from float to 8.24 2761 uint32_t vl = (uint32_t)(left * (1 << 24)); 2762 uint32_t vr = (uint32_t)(right * (1 << 24)); 2763 2764 // Delegate volume control to effect in track effect chain if needed 2765 // only one effect chain can be present on DirectOutputThread, so if 2766 // there is one, the track is connected to it 2767 if (!effectChains.isEmpty()) { 2768 // Do not ramp volume if volume is controlled by effect 2769 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2770 rampVolume = false; 2771 } 2772 } 2773 2774 // Convert volumes from 8.24 to 4.12 format 2775 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2776 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2777 leftVol = (uint16_t)v_clamped; 2778 v_clamped = (vr + (1 << 11)) >> 12; 2779 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2780 rightVol = (uint16_t)v_clamped; 2781 } else { 2782 leftVol = mLeftVolShort; 2783 rightVol = mRightVolShort; 2784 rampVolume = false; 2785 } 2786 2787 // reset retry count 2788 track->mRetryCount = kMaxTrackRetriesDirect; 2789 activeTrack = t; 2790 mixerStatus = MIXER_TRACKS_READY; 2791 } else { 2792 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2793 if (track->isStopped()) { 2794 track->reset(); 2795 } 2796 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2797 // We have consumed all the buffers of this track. 2798 // Remove it from the list of active tracks. 2799 trackToRemove = track; 2800 } else { 2801 // No buffers for this track. Give it a few chances to 2802 // fill a buffer, then remove it from active list. 2803 if (--(track->mRetryCount) <= 0) { 2804 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2805 trackToRemove = track; 2806 } else { 2807 mixerStatus = MIXER_TRACKS_ENABLED; 2808 } 2809 } 2810 } 2811 } 2812 2813 // remove all the tracks that need to be... 2814 if (CC_UNLIKELY(trackToRemove != 0)) { 2815 mActiveTracks.remove(trackToRemove); 2816 if (!effectChains.isEmpty()) { 2817 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2818 trackToRemove->sessionId()); 2819 effectChains[0]->decActiveTrackCnt(); 2820 } 2821 if (trackToRemove->isTerminated()) { 2822 removeTrack_l(trackToRemove); 2823 } 2824 } 2825 2826 lockEffectChains_l(effectChains); 2827 } 2828 2829 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2830 AudioBufferProvider::Buffer buffer; 2831 size_t frameCount = mFrameCount; 2832 curBuf = (int8_t *)mMixBuffer; 2833 // output audio to hardware 2834 while (frameCount) { 2835 buffer.frameCount = frameCount; 2836 activeTrack->getNextBuffer(&buffer); 2837 if (CC_UNLIKELY(buffer.raw == NULL)) { 2838 memset(curBuf, 0, frameCount * mFrameSize); 2839 break; 2840 } 2841 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2842 frameCount -= buffer.frameCount; 2843 curBuf += buffer.frameCount * mFrameSize; 2844 activeTrack->releaseBuffer(&buffer); 2845 } 2846 sleepTime = 0; 2847 standbyTime = systemTime() + standbyDelay; 2848 } else { 2849 if (sleepTime == 0) { 2850 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2851 sleepTime = activeSleepTime; 2852 } else { 2853 sleepTime = idleSleepTime; 2854 } 2855 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2856 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2857 sleepTime = 0; 2858 } 2859 } 2860 2861 if (mSuspended) { 2862 sleepTime = suspendSleepTimeUs(); 2863 } 2864 // sleepTime == 0 means we must write to audio hardware 2865 if (sleepTime == 0) { 2866 if (mixerStatus == MIXER_TRACKS_READY) { 2867 applyVolume(leftVol, rightVol, rampVolume); 2868 } 2869 for (size_t i = 0; i < effectChains.size(); i ++) { 2870 effectChains[i]->process_l(); 2871 } 2872 unlockEffectChains(effectChains); 2873 2874 mLastWriteTime = systemTime(); 2875 mInWrite = true; 2876 mBytesWritten += mixBufferSize; 2877 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2878 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2879 mNumWrites++; 2880 mInWrite = false; 2881 mStandby = false; 2882 } else { 2883 unlockEffectChains(effectChains); 2884 usleep(sleepTime); 2885 } 2886 2887 // finally let go of removed track, without the lock held 2888 // since we can't guarantee the destructors won't acquire that 2889 // same lock. 2890 trackToRemove.clear(); 2891 activeTrack.clear(); 2892 2893 // Effect chains will be actually deleted here if they were removed from 2894 // mEffectChains list during mixing or effects processing 2895 effectChains.clear(); 2896 } 2897 2898 if (!mStandby) { 2899 mOutput->stream->common.standby(&mOutput->stream->common); 2900 } 2901 2902 releaseWakeLock(); 2903 2904 ALOGV("DirectOutputThread %p exiting", this); 2905 return false; 2906} 2907 2908// getTrackName_l() must be called with ThreadBase::mLock held 2909int AudioFlinger::DirectOutputThread::getTrackName_l() 2910{ 2911 return 0; 2912} 2913 2914// deleteTrackName_l() must be called with ThreadBase::mLock held 2915void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2916{ 2917} 2918 2919// checkForNewParameters_l() must be called with ThreadBase::mLock held 2920bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2921{ 2922 bool reconfig = false; 2923 2924 while (!mNewParameters.isEmpty()) { 2925 status_t status = NO_ERROR; 2926 String8 keyValuePair = mNewParameters[0]; 2927 AudioParameter param = AudioParameter(keyValuePair); 2928 int value; 2929 2930 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2931 // do not accept frame count changes if tracks are open as the track buffer 2932 // size depends on frame count and correct behavior would not be garantied 2933 // if frame count is changed after track creation 2934 if (!mTracks.isEmpty()) { 2935 status = INVALID_OPERATION; 2936 } else { 2937 reconfig = true; 2938 } 2939 } 2940 if (status == NO_ERROR) { 2941 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2942 keyValuePair.string()); 2943 if (!mStandby && status == INVALID_OPERATION) { 2944 mOutput->stream->common.standby(&mOutput->stream->common); 2945 mStandby = true; 2946 mBytesWritten = 0; 2947 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2948 keyValuePair.string()); 2949 } 2950 if (status == NO_ERROR && reconfig) { 2951 readOutputParameters(); 2952 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2953 } 2954 } 2955 2956 mNewParameters.removeAt(0); 2957 2958 mParamStatus = status; 2959 mParamCond.signal(); 2960 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2961 // already timed out waiting for the status and will never signal the condition. 2962 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2963 } 2964 return reconfig; 2965} 2966 2967uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2968{ 2969 uint32_t time; 2970 if (audio_is_linear_pcm(mFormat)) { 2971 time = PlaybackThread::activeSleepTimeUs(); 2972 } else { 2973 time = 10000; 2974 } 2975 return time; 2976} 2977 2978uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2979{ 2980 uint32_t time; 2981 if (audio_is_linear_pcm(mFormat)) { 2982 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2983 } else { 2984 time = 10000; 2985 } 2986 return time; 2987} 2988 2989uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2990{ 2991 uint32_t time; 2992 if (audio_is_linear_pcm(mFormat)) { 2993 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2994 } else { 2995 time = 10000; 2996 } 2997 return time; 2998} 2999 3000 3001// ---------------------------------------------------------------------------- 3002 3003AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3004 AudioFlinger::MixerThread* mainThread, int id) 3005 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3006 mWaitTimeMs(UINT_MAX) 3007{ 3008 addOutputTrack(mainThread); 3009} 3010 3011AudioFlinger::DuplicatingThread::~DuplicatingThread() 3012{ 3013 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3014 mOutputTracks[i]->destroy(); 3015 } 3016 mOutputTracks.clear(); 3017} 3018 3019bool AudioFlinger::DuplicatingThread::threadLoop() 3020{ 3021 Vector< sp<Track> > tracksToRemove; 3022 mixer_state mixerStatus = MIXER_IDLE; 3023 nsecs_t standbyTime = systemTime(); 3024 size_t mixBufferSize = mFrameCount*mFrameSize; 3025 SortedVector< sp<OutputTrack> > outputTracks; 3026 uint32_t writeFrames = 0; 3027 uint32_t activeSleepTime = activeSleepTimeUs(); 3028 uint32_t idleSleepTime = idleSleepTimeUs(); 3029 uint32_t sleepTime = idleSleepTime; 3030 Vector< sp<EffectChain> > effectChains; 3031 3032 acquireWakeLock(); 3033 3034 while (!exitPending()) 3035 { 3036 processConfigEvents(); 3037 3038 mixerStatus = MIXER_IDLE; 3039 { // scope for the mLock 3040 3041 Mutex::Autolock _l(mLock); 3042 3043 if (checkForNewParameters_l()) { 3044 mixBufferSize = mFrameCount*mFrameSize; 3045 updateWaitTime(); 3046 activeSleepTime = activeSleepTimeUs(); 3047 idleSleepTime = idleSleepTimeUs(); 3048 } 3049 3050 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3051 3052 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3053 outputTracks.add(mOutputTracks[i]); 3054 } 3055 3056 // put audio hardware into standby after short delay 3057 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3058 mSuspended)) { 3059 if (!mStandby) { 3060 for (size_t i = 0; i < outputTracks.size(); i++) { 3061 outputTracks[i]->stop(); 3062 } 3063 mStandby = true; 3064 mBytesWritten = 0; 3065 } 3066 3067 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3068 // we're about to wait, flush the binder command buffer 3069 IPCThreadState::self()->flushCommands(); 3070 outputTracks.clear(); 3071 3072 if (exitPending()) break; 3073 3074 releaseWakeLock_l(); 3075 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3076 mWaitWorkCV.wait(mLock); 3077 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3078 acquireWakeLock_l(); 3079 3080 mPrevMixerStatus = MIXER_IDLE; 3081 if (!mMasterMute) { 3082 char value[PROPERTY_VALUE_MAX]; 3083 property_get("ro.audio.silent", value, "0"); 3084 if (atoi(value)) { 3085 ALOGD("Silence is golden"); 3086 setMasterMute(true); 3087 } 3088 } 3089 3090 standbyTime = systemTime() + kStandbyTimeInNsecs; 3091 sleepTime = idleSleepTime; 3092 continue; 3093 } 3094 } 3095 3096 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3097 3098 // prevent any changes in effect chain list and in each effect chain 3099 // during mixing and effect process as the audio buffers could be deleted 3100 // or modified if an effect is created or deleted 3101 lockEffectChains_l(effectChains); 3102 } 3103 3104 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3105 // mix buffers... 3106 if (outputsReady(outputTracks)) { 3107 mAudioMixer->process(); 3108 } else { 3109 memset(mMixBuffer, 0, mixBufferSize); 3110 } 3111 sleepTime = 0; 3112 writeFrames = mFrameCount; 3113 } else { 3114 if (sleepTime == 0) { 3115 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3116 sleepTime = activeSleepTime; 3117 } else { 3118 sleepTime = idleSleepTime; 3119 } 3120 } else if (mBytesWritten != 0) { 3121 // flush remaining overflow buffers in output tracks 3122 for (size_t i = 0; i < outputTracks.size(); i++) { 3123 if (outputTracks[i]->isActive()) { 3124 sleepTime = 0; 3125 writeFrames = 0; 3126 memset(mMixBuffer, 0, mixBufferSize); 3127 break; 3128 } 3129 } 3130 } 3131 } 3132 3133 if (mSuspended) { 3134 sleepTime = suspendSleepTimeUs(); 3135 } 3136 // sleepTime == 0 means we must write to audio hardware 3137 if (sleepTime == 0) { 3138 for (size_t i = 0; i < effectChains.size(); i ++) { 3139 effectChains[i]->process_l(); 3140 } 3141 // enable changes in effect chain 3142 unlockEffectChains(effectChains); 3143 3144 standbyTime = systemTime() + kStandbyTimeInNsecs; 3145 for (size_t i = 0; i < outputTracks.size(); i++) { 3146 outputTracks[i]->write(mMixBuffer, writeFrames); 3147 } 3148 mStandby = false; 3149 mBytesWritten += mixBufferSize; 3150 } else { 3151 // enable changes in effect chain 3152 unlockEffectChains(effectChains); 3153 usleep(sleepTime); 3154 } 3155 3156 // finally let go of all our tracks, without the lock held 3157 // since we can't guarantee the destructors won't acquire that 3158 // same lock. 3159 tracksToRemove.clear(); 3160 outputTracks.clear(); 3161 3162 // Effect chains will be actually deleted here if they were removed from 3163 // mEffectChains list during mixing or effects processing 3164 effectChains.clear(); 3165 } 3166 3167 releaseWakeLock(); 3168 3169 return false; 3170} 3171 3172void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3173{ 3174 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3175 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3176 this, 3177 mSampleRate, 3178 mFormat, 3179 mChannelMask, 3180 frameCount); 3181 if (outputTrack->cblk() != NULL) { 3182 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3183 mOutputTracks.add(outputTrack); 3184 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3185 updateWaitTime(); 3186 } 3187} 3188 3189void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3190{ 3191 Mutex::Autolock _l(mLock); 3192 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3193 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3194 mOutputTracks[i]->destroy(); 3195 mOutputTracks.removeAt(i); 3196 updateWaitTime(); 3197 return; 3198 } 3199 } 3200 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3201} 3202 3203void AudioFlinger::DuplicatingThread::updateWaitTime() 3204{ 3205 mWaitTimeMs = UINT_MAX; 3206 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3207 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3208 if (strong != 0) { 3209 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3210 if (waitTimeMs < mWaitTimeMs) { 3211 mWaitTimeMs = waitTimeMs; 3212 } 3213 } 3214 } 3215} 3216 3217 3218bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3219{ 3220 for (size_t i = 0; i < outputTracks.size(); i++) { 3221 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3222 if (thread == 0) { 3223 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3224 return false; 3225 } 3226 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3227 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3228 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3229 return false; 3230 } 3231 } 3232 return true; 3233} 3234 3235uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3236{ 3237 return (mWaitTimeMs * 1000) / 2; 3238} 3239 3240// ---------------------------------------------------------------------------- 3241 3242// TrackBase constructor must be called with AudioFlinger::mLock held 3243AudioFlinger::ThreadBase::TrackBase::TrackBase( 3244 const wp<ThreadBase>& thread, 3245 const sp<Client>& client, 3246 uint32_t sampleRate, 3247 audio_format_t format, 3248 uint32_t channelMask, 3249 int frameCount, 3250 uint32_t flags, 3251 const sp<IMemory>& sharedBuffer, 3252 int sessionId) 3253 : RefBase(), 3254 mThread(thread), 3255 mClient(client), 3256 mCblk(NULL), 3257 // mBuffer 3258 // mBufferEnd 3259 mFrameCount(0), 3260 mState(IDLE), 3261 mClientTid(-1), 3262 mFormat(format), 3263 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3264 mSessionId(sessionId) 3265 // mChannelCount 3266 // mChannelMask 3267{ 3268 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3269 3270 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3271 size_t size = sizeof(audio_track_cblk_t); 3272 uint8_t channelCount = popcount(channelMask); 3273 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3274 if (sharedBuffer == 0) { 3275 size += bufferSize; 3276 } 3277 3278 if (client != NULL) { 3279 mCblkMemory = client->heap()->allocate(size); 3280 if (mCblkMemory != 0) { 3281 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3282 if (mCblk != NULL) { // construct the shared structure in-place. 3283 new(mCblk) audio_track_cblk_t(); 3284 // clear all buffers 3285 mCblk->frameCount = frameCount; 3286 mCblk->sampleRate = sampleRate; 3287 mChannelCount = channelCount; 3288 mChannelMask = channelMask; 3289 if (sharedBuffer == 0) { 3290 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3291 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3292 // Force underrun condition to avoid false underrun callback until first data is 3293 // written to buffer (other flags are cleared) 3294 mCblk->flags = CBLK_UNDERRUN_ON; 3295 } else { 3296 mBuffer = sharedBuffer->pointer(); 3297 } 3298 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3299 } 3300 } else { 3301 ALOGE("not enough memory for AudioTrack size=%u", size); 3302 client->heap()->dump("AudioTrack"); 3303 return; 3304 } 3305 } else { 3306 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3307 // construct the shared structure in-place. 3308 new(mCblk) audio_track_cblk_t(); 3309 // clear all buffers 3310 mCblk->frameCount = frameCount; 3311 mCblk->sampleRate = sampleRate; 3312 mChannelCount = channelCount; 3313 mChannelMask = channelMask; 3314 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3315 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3316 // Force underrun condition to avoid false underrun callback until first data is 3317 // written to buffer (other flags are cleared) 3318 mCblk->flags = CBLK_UNDERRUN_ON; 3319 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3320 } 3321} 3322 3323AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3324{ 3325 if (mCblk != NULL) { 3326 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3327 if (mClient == NULL) { 3328 delete mCblk; 3329 } 3330 } 3331 mCblkMemory.clear(); // and free the shared memory 3332 if (mClient != 0) { 3333 // Client destructor must run with AudioFlinger mutex locked 3334 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3335 // If the client's reference count drops to zero, the associated destructor 3336 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3337 // relying on the automatic clear() at end of scope. 3338 mClient.clear(); 3339 } 3340} 3341 3342void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3343{ 3344 buffer->raw = NULL; 3345 mFrameCount = buffer->frameCount; 3346 step(); 3347 buffer->frameCount = 0; 3348} 3349 3350bool AudioFlinger::ThreadBase::TrackBase::step() { 3351 bool result; 3352 audio_track_cblk_t* cblk = this->cblk(); 3353 3354 result = cblk->stepServer(mFrameCount); 3355 if (!result) { 3356 ALOGV("stepServer failed acquiring cblk mutex"); 3357 mFlags |= STEPSERVER_FAILED; 3358 } 3359 return result; 3360} 3361 3362void AudioFlinger::ThreadBase::TrackBase::reset() { 3363 audio_track_cblk_t* cblk = this->cblk(); 3364 3365 cblk->user = 0; 3366 cblk->server = 0; 3367 cblk->userBase = 0; 3368 cblk->serverBase = 0; 3369 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3370 ALOGV("TrackBase::reset"); 3371} 3372 3373sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3374{ 3375 return mCblkMemory; 3376} 3377 3378int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3379 return (int)mCblk->sampleRate; 3380} 3381 3382int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3383 return (const int)mChannelCount; 3384} 3385 3386uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3387 return mChannelMask; 3388} 3389 3390void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3391 audio_track_cblk_t* cblk = this->cblk(); 3392 size_t frameSize = cblk->frameSize; 3393 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3394 int8_t *bufferEnd = bufferStart + frames * frameSize; 3395 3396 // Check validity of returned pointer in case the track control block would have been corrupted. 3397 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3398 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3399 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3400 server %d, serverBase %d, user %d, userBase %d", 3401 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3402 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3403 return NULL; 3404 } 3405 3406 return bufferStart; 3407} 3408 3409// ---------------------------------------------------------------------------- 3410 3411// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3412AudioFlinger::PlaybackThread::Track::Track( 3413 const wp<ThreadBase>& thread, 3414 const sp<Client>& client, 3415 audio_stream_type_t streamType, 3416 uint32_t sampleRate, 3417 audio_format_t format, 3418 uint32_t channelMask, 3419 int frameCount, 3420 const sp<IMemory>& sharedBuffer, 3421 int sessionId) 3422 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3423 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3424 mAuxEffectId(0), mHasVolumeController(false) 3425{ 3426 if (mCblk != NULL) { 3427 sp<ThreadBase> baseThread = thread.promote(); 3428 if (baseThread != 0) { 3429 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3430 mName = playbackThread->getTrackName_l(); 3431 mMainBuffer = playbackThread->mixBuffer(); 3432 } 3433 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3434 if (mName < 0) { 3435 ALOGE("no more track names available"); 3436 } 3437 mStreamType = streamType; 3438 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3439 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3440 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3441 } 3442} 3443 3444AudioFlinger::PlaybackThread::Track::~Track() 3445{ 3446 ALOGV("PlaybackThread::Track destructor"); 3447 sp<ThreadBase> thread = mThread.promote(); 3448 if (thread != 0) { 3449 Mutex::Autolock _l(thread->mLock); 3450 mState = TERMINATED; 3451 } 3452} 3453 3454void AudioFlinger::PlaybackThread::Track::destroy() 3455{ 3456 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3457 // by removing it from mTracks vector, so there is a risk that this Tracks's 3458 // desctructor is called. As the destructor needs to lock mLock, 3459 // we must acquire a strong reference on this Track before locking mLock 3460 // here so that the destructor is called only when exiting this function. 3461 // On the other hand, as long as Track::destroy() is only called by 3462 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3463 // this Track with its member mTrack. 3464 sp<Track> keep(this); 3465 { // scope for mLock 3466 sp<ThreadBase> thread = mThread.promote(); 3467 if (thread != 0) { 3468 if (!isOutputTrack()) { 3469 if (mState == ACTIVE || mState == RESUMING) { 3470 AudioSystem::stopOutput(thread->id(), 3471 (audio_stream_type_t)mStreamType, 3472 mSessionId); 3473 3474 // to track the speaker usage 3475 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3476 } 3477 AudioSystem::releaseOutput(thread->id()); 3478 } 3479 Mutex::Autolock _l(thread->mLock); 3480 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3481 playbackThread->destroyTrack_l(this); 3482 } 3483 } 3484} 3485 3486void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3487{ 3488 uint32_t vlr = mCblk->getVolumeLR(); 3489 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", 3490 mName - AudioMixer::TRACK0, 3491 (mClient == 0) ? getpid() : mClient->pid(), 3492 mStreamType, 3493 mFormat, 3494 mChannelMask, 3495 mSessionId, 3496 mFrameCount, 3497 mState, 3498 mMute, 3499 mFillingUpStatus, 3500 mCblk->sampleRate, 3501 vlr & 0xFFFF, 3502 vlr >> 16, 3503 mCblk->server, 3504 mCblk->user, 3505 (int)mMainBuffer, 3506 (int)mAuxBuffer); 3507} 3508 3509status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3510{ 3511 audio_track_cblk_t* cblk = this->cblk(); 3512 uint32_t framesReady; 3513 uint32_t framesReq = buffer->frameCount; 3514 3515 // Check if last stepServer failed, try to step now 3516 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3517 if (!step()) goto getNextBuffer_exit; 3518 ALOGV("stepServer recovered"); 3519 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3520 } 3521 3522 framesReady = cblk->framesReady(); 3523 3524 if (CC_LIKELY(framesReady)) { 3525 uint32_t s = cblk->server; 3526 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3527 3528 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3529 if (framesReq > framesReady) { 3530 framesReq = framesReady; 3531 } 3532 if (s + framesReq > bufferEnd) { 3533 framesReq = bufferEnd - s; 3534 } 3535 3536 buffer->raw = getBuffer(s, framesReq); 3537 if (buffer->raw == NULL) goto getNextBuffer_exit; 3538 3539 buffer->frameCount = framesReq; 3540 return NO_ERROR; 3541 } 3542 3543getNextBuffer_exit: 3544 buffer->raw = NULL; 3545 buffer->frameCount = 0; 3546 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3547 return NOT_ENOUGH_DATA; 3548} 3549 3550bool AudioFlinger::PlaybackThread::Track::isReady() const { 3551 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3552 3553 if (mCblk->framesReady() >= mCblk->frameCount || 3554 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3555 mFillingUpStatus = FS_FILLED; 3556 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3557 return true; 3558 } 3559 return false; 3560} 3561 3562status_t AudioFlinger::PlaybackThread::Track::start() 3563{ 3564 status_t status = NO_ERROR; 3565 ALOGV("start(%d), calling thread %d session %d", 3566 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3567 sp<ThreadBase> thread = mThread.promote(); 3568 if (thread != 0) { 3569 Mutex::Autolock _l(thread->mLock); 3570 track_state state = mState; 3571 // here the track could be either new, or restarted 3572 // in both cases "unstop" the track 3573 if (mState == PAUSED) { 3574 mState = TrackBase::RESUMING; 3575 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3576 } else { 3577 mState = TrackBase::ACTIVE; 3578 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3579 } 3580 3581 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3582 thread->mLock.unlock(); 3583 status = AudioSystem::startOutput(thread->id(), 3584 (audio_stream_type_t)mStreamType, 3585 mSessionId); 3586 thread->mLock.lock(); 3587 3588 // to track the speaker usage 3589 if (status == NO_ERROR) { 3590 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3591 } 3592 } 3593 if (status == NO_ERROR) { 3594 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3595 playbackThread->addTrack_l(this); 3596 } else { 3597 mState = state; 3598 } 3599 } else { 3600 status = BAD_VALUE; 3601 } 3602 return status; 3603} 3604 3605void AudioFlinger::PlaybackThread::Track::stop() 3606{ 3607 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3608 sp<ThreadBase> thread = mThread.promote(); 3609 if (thread != 0) { 3610 Mutex::Autolock _l(thread->mLock); 3611 track_state state = mState; 3612 if (mState > STOPPED) { 3613 mState = STOPPED; 3614 // If the track is not active (PAUSED and buffers full), flush buffers 3615 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3616 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3617 reset(); 3618 } 3619 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3620 } 3621 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3622 thread->mLock.unlock(); 3623 AudioSystem::stopOutput(thread->id(), 3624 (audio_stream_type_t)mStreamType, 3625 mSessionId); 3626 thread->mLock.lock(); 3627 3628 // to track the speaker usage 3629 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3630 } 3631 } 3632} 3633 3634void AudioFlinger::PlaybackThread::Track::pause() 3635{ 3636 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3637 sp<ThreadBase> thread = mThread.promote(); 3638 if (thread != 0) { 3639 Mutex::Autolock _l(thread->mLock); 3640 if (mState == ACTIVE || mState == RESUMING) { 3641 mState = PAUSING; 3642 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3643 if (!isOutputTrack()) { 3644 thread->mLock.unlock(); 3645 AudioSystem::stopOutput(thread->id(), 3646 (audio_stream_type_t)mStreamType, 3647 mSessionId); 3648 thread->mLock.lock(); 3649 3650 // to track the speaker usage 3651 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3652 } 3653 } 3654 } 3655} 3656 3657void AudioFlinger::PlaybackThread::Track::flush() 3658{ 3659 ALOGV("flush(%d)", mName); 3660 sp<ThreadBase> thread = mThread.promote(); 3661 if (thread != 0) { 3662 Mutex::Autolock _l(thread->mLock); 3663 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3664 return; 3665 } 3666 // No point remaining in PAUSED state after a flush => go to 3667 // STOPPED state 3668 mState = STOPPED; 3669 3670 // do not reset the track if it is still in the process of being stopped or paused. 3671 // this will be done by prepareTracks_l() when the track is stopped. 3672 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3673 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3674 reset(); 3675 } 3676 } 3677} 3678 3679void AudioFlinger::PlaybackThread::Track::reset() 3680{ 3681 // Do not reset twice to avoid discarding data written just after a flush and before 3682 // the audioflinger thread detects the track is stopped. 3683 if (!mResetDone) { 3684 TrackBase::reset(); 3685 // Force underrun condition to avoid false underrun callback until first data is 3686 // written to buffer 3687 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3688 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3689 mFillingUpStatus = FS_FILLING; 3690 mResetDone = true; 3691 } 3692} 3693 3694void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3695{ 3696 mMute = muted; 3697} 3698 3699status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3700{ 3701 status_t status = DEAD_OBJECT; 3702 sp<ThreadBase> thread = mThread.promote(); 3703 if (thread != 0) { 3704 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3705 status = playbackThread->attachAuxEffect(this, EffectId); 3706 } 3707 return status; 3708} 3709 3710void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3711{ 3712 mAuxEffectId = EffectId; 3713 mAuxBuffer = buffer; 3714} 3715 3716// ---------------------------------------------------------------------------- 3717 3718// RecordTrack constructor must be called with AudioFlinger::mLock held 3719AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3720 const wp<ThreadBase>& thread, 3721 const sp<Client>& client, 3722 uint32_t sampleRate, 3723 audio_format_t format, 3724 uint32_t channelMask, 3725 int frameCount, 3726 uint32_t flags, 3727 int sessionId) 3728 : TrackBase(thread, client, sampleRate, format, 3729 channelMask, frameCount, flags, 0, sessionId), 3730 mOverflow(false) 3731{ 3732 if (mCblk != NULL) { 3733 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3734 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3735 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3736 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3737 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3738 } else { 3739 mCblk->frameSize = sizeof(int8_t); 3740 } 3741 } 3742} 3743 3744AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3745{ 3746 sp<ThreadBase> thread = mThread.promote(); 3747 if (thread != 0) { 3748 AudioSystem::releaseInput(thread->id()); 3749 } 3750} 3751 3752status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3753{ 3754 audio_track_cblk_t* cblk = this->cblk(); 3755 uint32_t framesAvail; 3756 uint32_t framesReq = buffer->frameCount; 3757 3758 // Check if last stepServer failed, try to step now 3759 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3760 if (!step()) goto getNextBuffer_exit; 3761 ALOGV("stepServer recovered"); 3762 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3763 } 3764 3765 framesAvail = cblk->framesAvailable_l(); 3766 3767 if (CC_LIKELY(framesAvail)) { 3768 uint32_t s = cblk->server; 3769 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3770 3771 if (framesReq > framesAvail) { 3772 framesReq = framesAvail; 3773 } 3774 if (s + framesReq > bufferEnd) { 3775 framesReq = bufferEnd - s; 3776 } 3777 3778 buffer->raw = getBuffer(s, framesReq); 3779 if (buffer->raw == NULL) goto getNextBuffer_exit; 3780 3781 buffer->frameCount = framesReq; 3782 return NO_ERROR; 3783 } 3784 3785getNextBuffer_exit: 3786 buffer->raw = NULL; 3787 buffer->frameCount = 0; 3788 return NOT_ENOUGH_DATA; 3789} 3790 3791status_t AudioFlinger::RecordThread::RecordTrack::start() 3792{ 3793 sp<ThreadBase> thread = mThread.promote(); 3794 if (thread != 0) { 3795 RecordThread *recordThread = (RecordThread *)thread.get(); 3796 return recordThread->start(this); 3797 } else { 3798 return BAD_VALUE; 3799 } 3800} 3801 3802void AudioFlinger::RecordThread::RecordTrack::stop() 3803{ 3804 sp<ThreadBase> thread = mThread.promote(); 3805 if (thread != 0) { 3806 RecordThread *recordThread = (RecordThread *)thread.get(); 3807 recordThread->stop(this); 3808 TrackBase::reset(); 3809 // Force overerrun condition to avoid false overrun callback until first data is 3810 // read from buffer 3811 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3812 } 3813} 3814 3815void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3816{ 3817 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3818 (mClient == 0) ? getpid() : mClient->pid(), 3819 mFormat, 3820 mChannelMask, 3821 mSessionId, 3822 mFrameCount, 3823 mState, 3824 mCblk->sampleRate, 3825 mCblk->server, 3826 mCblk->user); 3827} 3828 3829 3830// ---------------------------------------------------------------------------- 3831 3832AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3833 const wp<ThreadBase>& thread, 3834 DuplicatingThread *sourceThread, 3835 uint32_t sampleRate, 3836 audio_format_t format, 3837 uint32_t channelMask, 3838 int frameCount) 3839 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3840 mActive(false), mSourceThread(sourceThread) 3841{ 3842 3843 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3844 if (mCblk != NULL) { 3845 mCblk->flags |= CBLK_DIRECTION_OUT; 3846 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3847 mOutBuffer.frameCount = 0; 3848 playbackThread->mTracks.add(this); 3849 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3850 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3851 mCblk, mBuffer, mCblk->buffers, 3852 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3853 } else { 3854 ALOGW("Error creating output track on thread %p", playbackThread); 3855 } 3856} 3857 3858AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3859{ 3860 clearBufferQueue(); 3861} 3862 3863status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3864{ 3865 status_t status = Track::start(); 3866 if (status != NO_ERROR) { 3867 return status; 3868 } 3869 3870 mActive = true; 3871 mRetryCount = 127; 3872 return status; 3873} 3874 3875void AudioFlinger::PlaybackThread::OutputTrack::stop() 3876{ 3877 Track::stop(); 3878 clearBufferQueue(); 3879 mOutBuffer.frameCount = 0; 3880 mActive = false; 3881} 3882 3883bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3884{ 3885 Buffer *pInBuffer; 3886 Buffer inBuffer; 3887 uint32_t channelCount = mChannelCount; 3888 bool outputBufferFull = false; 3889 inBuffer.frameCount = frames; 3890 inBuffer.i16 = data; 3891 3892 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3893 3894 if (!mActive && frames != 0) { 3895 start(); 3896 sp<ThreadBase> thread = mThread.promote(); 3897 if (thread != 0) { 3898 MixerThread *mixerThread = (MixerThread *)thread.get(); 3899 if (mCblk->frameCount > frames){ 3900 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3901 uint32_t startFrames = (mCblk->frameCount - frames); 3902 pInBuffer = new Buffer; 3903 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3904 pInBuffer->frameCount = startFrames; 3905 pInBuffer->i16 = pInBuffer->mBuffer; 3906 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3907 mBufferQueue.add(pInBuffer); 3908 } else { 3909 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3910 } 3911 } 3912 } 3913 } 3914 3915 while (waitTimeLeftMs) { 3916 // First write pending buffers, then new data 3917 if (mBufferQueue.size()) { 3918 pInBuffer = mBufferQueue.itemAt(0); 3919 } else { 3920 pInBuffer = &inBuffer; 3921 } 3922 3923 if (pInBuffer->frameCount == 0) { 3924 break; 3925 } 3926 3927 if (mOutBuffer.frameCount == 0) { 3928 mOutBuffer.frameCount = pInBuffer->frameCount; 3929 nsecs_t startTime = systemTime(); 3930 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3931 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3932 outputBufferFull = true; 3933 break; 3934 } 3935 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3936 if (waitTimeLeftMs >= waitTimeMs) { 3937 waitTimeLeftMs -= waitTimeMs; 3938 } else { 3939 waitTimeLeftMs = 0; 3940 } 3941 } 3942 3943 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3944 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3945 mCblk->stepUser(outFrames); 3946 pInBuffer->frameCount -= outFrames; 3947 pInBuffer->i16 += outFrames * channelCount; 3948 mOutBuffer.frameCount -= outFrames; 3949 mOutBuffer.i16 += outFrames * channelCount; 3950 3951 if (pInBuffer->frameCount == 0) { 3952 if (mBufferQueue.size()) { 3953 mBufferQueue.removeAt(0); 3954 delete [] pInBuffer->mBuffer; 3955 delete pInBuffer; 3956 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3957 } else { 3958 break; 3959 } 3960 } 3961 } 3962 3963 // If we could not write all frames, allocate a buffer and queue it for next time. 3964 if (inBuffer.frameCount) { 3965 sp<ThreadBase> thread = mThread.promote(); 3966 if (thread != 0 && !thread->standby()) { 3967 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3968 pInBuffer = new Buffer; 3969 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3970 pInBuffer->frameCount = inBuffer.frameCount; 3971 pInBuffer->i16 = pInBuffer->mBuffer; 3972 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3973 mBufferQueue.add(pInBuffer); 3974 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3975 } else { 3976 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3977 } 3978 } 3979 } 3980 3981 // Calling write() with a 0 length buffer, means that no more data will be written: 3982 // If no more buffers are pending, fill output track buffer to make sure it is started 3983 // by output mixer. 3984 if (frames == 0 && mBufferQueue.size() == 0) { 3985 if (mCblk->user < mCblk->frameCount) { 3986 frames = mCblk->frameCount - mCblk->user; 3987 pInBuffer = new Buffer; 3988 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3989 pInBuffer->frameCount = frames; 3990 pInBuffer->i16 = pInBuffer->mBuffer; 3991 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3992 mBufferQueue.add(pInBuffer); 3993 } else if (mActive) { 3994 stop(); 3995 } 3996 } 3997 3998 return outputBufferFull; 3999} 4000 4001status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4002{ 4003 int active; 4004 status_t result; 4005 audio_track_cblk_t* cblk = mCblk; 4006 uint32_t framesReq = buffer->frameCount; 4007 4008// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4009 buffer->frameCount = 0; 4010 4011 uint32_t framesAvail = cblk->framesAvailable(); 4012 4013 4014 if (framesAvail == 0) { 4015 Mutex::Autolock _l(cblk->lock); 4016 goto start_loop_here; 4017 while (framesAvail == 0) { 4018 active = mActive; 4019 if (CC_UNLIKELY(!active)) { 4020 ALOGV("Not active and NO_MORE_BUFFERS"); 4021 return NO_MORE_BUFFERS; 4022 } 4023 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4024 if (result != NO_ERROR) { 4025 return NO_MORE_BUFFERS; 4026 } 4027 // read the server count again 4028 start_loop_here: 4029 framesAvail = cblk->framesAvailable_l(); 4030 } 4031 } 4032 4033// if (framesAvail < framesReq) { 4034// return NO_MORE_BUFFERS; 4035// } 4036 4037 if (framesReq > framesAvail) { 4038 framesReq = framesAvail; 4039 } 4040 4041 uint32_t u = cblk->user; 4042 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4043 4044 if (u + framesReq > bufferEnd) { 4045 framesReq = bufferEnd - u; 4046 } 4047 4048 buffer->frameCount = framesReq; 4049 buffer->raw = (void *)cblk->buffer(u); 4050 return NO_ERROR; 4051} 4052 4053 4054void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4055{ 4056 size_t size = mBufferQueue.size(); 4057 Buffer *pBuffer; 4058 4059 for (size_t i = 0; i < size; i++) { 4060 pBuffer = mBufferQueue.itemAt(i); 4061 delete [] pBuffer->mBuffer; 4062 delete pBuffer; 4063 } 4064 mBufferQueue.clear(); 4065} 4066 4067// ---------------------------------------------------------------------------- 4068 4069AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4070 : RefBase(), 4071 mAudioFlinger(audioFlinger), 4072 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4073 mPid(pid) 4074{ 4075 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4076} 4077 4078// Client destructor must be called with AudioFlinger::mLock held 4079AudioFlinger::Client::~Client() 4080{ 4081 mAudioFlinger->removeClient_l(mPid); 4082} 4083 4084sp<MemoryDealer> AudioFlinger::Client::heap() const 4085{ 4086 return mMemoryDealer; 4087} 4088 4089// ---------------------------------------------------------------------------- 4090 4091AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4092 const sp<IAudioFlingerClient>& client, 4093 pid_t pid) 4094 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4095{ 4096} 4097 4098AudioFlinger::NotificationClient::~NotificationClient() 4099{ 4100} 4101 4102void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4103{ 4104 sp<NotificationClient> keep(this); 4105 { 4106 mAudioFlinger->removeNotificationClient(mPid); 4107 } 4108} 4109 4110// ---------------------------------------------------------------------------- 4111 4112AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4113 : BnAudioTrack(), 4114 mTrack(track) 4115{ 4116} 4117 4118AudioFlinger::TrackHandle::~TrackHandle() { 4119 // just stop the track on deletion, associated resources 4120 // will be freed from the main thread once all pending buffers have 4121 // been played. Unless it's not in the active track list, in which 4122 // case we free everything now... 4123 mTrack->destroy(); 4124} 4125 4126sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4127 return mTrack->getCblk(); 4128} 4129 4130status_t AudioFlinger::TrackHandle::start() { 4131 return mTrack->start(); 4132} 4133 4134void AudioFlinger::TrackHandle::stop() { 4135 mTrack->stop(); 4136} 4137 4138void AudioFlinger::TrackHandle::flush() { 4139 mTrack->flush(); 4140} 4141 4142void AudioFlinger::TrackHandle::mute(bool e) { 4143 mTrack->mute(e); 4144} 4145 4146void AudioFlinger::TrackHandle::pause() { 4147 mTrack->pause(); 4148} 4149 4150status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4151{ 4152 return mTrack->attachAuxEffect(EffectId); 4153} 4154 4155status_t AudioFlinger::TrackHandle::onTransact( 4156 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4157{ 4158 return BnAudioTrack::onTransact(code, data, reply, flags); 4159} 4160 4161// ---------------------------------------------------------------------------- 4162 4163sp<IAudioRecord> AudioFlinger::openRecord( 4164 pid_t pid, 4165 int input, 4166 uint32_t sampleRate, 4167 audio_format_t format, 4168 uint32_t channelMask, 4169 int frameCount, 4170 uint32_t flags, 4171 int *sessionId, 4172 status_t *status) 4173{ 4174 sp<RecordThread::RecordTrack> recordTrack; 4175 sp<RecordHandle> recordHandle; 4176 sp<Client> client; 4177 wp<Client> wclient; 4178 status_t lStatus; 4179 RecordThread *thread; 4180 size_t inFrameCount; 4181 int lSessionId; 4182 4183 // check calling permissions 4184 if (!recordingAllowed()) { 4185 lStatus = PERMISSION_DENIED; 4186 goto Exit; 4187 } 4188 4189 // add client to list 4190 { // scope for mLock 4191 Mutex::Autolock _l(mLock); 4192 thread = checkRecordThread_l(input); 4193 if (thread == NULL) { 4194 lStatus = BAD_VALUE; 4195 goto Exit; 4196 } 4197 4198 wclient = mClients.valueFor(pid); 4199 if (wclient != NULL) { 4200 client = wclient.promote(); 4201 } else { 4202 client = new Client(this, pid); 4203 mClients.add(pid, client); 4204 } 4205 4206 // If no audio session id is provided, create one here 4207 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4208 lSessionId = *sessionId; 4209 } else { 4210 lSessionId = nextUniqueId(); 4211 if (sessionId != NULL) { 4212 *sessionId = lSessionId; 4213 } 4214 } 4215 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4216 recordTrack = thread->createRecordTrack_l(client, 4217 sampleRate, 4218 format, 4219 channelMask, 4220 frameCount, 4221 flags, 4222 lSessionId, 4223 &lStatus); 4224 } 4225 if (lStatus != NO_ERROR) { 4226 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4227 // destructor is called by the TrackBase destructor with mLock held 4228 client.clear(); 4229 recordTrack.clear(); 4230 goto Exit; 4231 } 4232 4233 // return to handle to client 4234 recordHandle = new RecordHandle(recordTrack); 4235 lStatus = NO_ERROR; 4236 4237Exit: 4238 if (status) { 4239 *status = lStatus; 4240 } 4241 return recordHandle; 4242} 4243 4244// ---------------------------------------------------------------------------- 4245 4246AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4247 : BnAudioRecord(), 4248 mRecordTrack(recordTrack) 4249{ 4250} 4251 4252AudioFlinger::RecordHandle::~RecordHandle() { 4253 stop(); 4254} 4255 4256sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4257 return mRecordTrack->getCblk(); 4258} 4259 4260status_t AudioFlinger::RecordHandle::start() { 4261 ALOGV("RecordHandle::start()"); 4262 return mRecordTrack->start(); 4263} 4264 4265void AudioFlinger::RecordHandle::stop() { 4266 ALOGV("RecordHandle::stop()"); 4267 mRecordTrack->stop(); 4268} 4269 4270status_t AudioFlinger::RecordHandle::onTransact( 4271 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4272{ 4273 return BnAudioRecord::onTransact(code, data, reply, flags); 4274} 4275 4276// ---------------------------------------------------------------------------- 4277 4278AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4279 AudioStreamIn *input, 4280 uint32_t sampleRate, 4281 uint32_t channels, 4282 int id, 4283 uint32_t device) : 4284 ThreadBase(audioFlinger, id, device, RECORD), 4285 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4286 // mRsmpInIndex and mInputBytes set by readInputParameters() 4287 mReqChannelCount(popcount(channels)), 4288 mReqSampleRate(sampleRate) 4289 // mBytesRead is only meaningful while active, and so is cleared in start() 4290 // (but might be better to also clear here for dump?) 4291{ 4292 snprintf(mName, kNameLength, "AudioIn_%d", id); 4293 4294 readInputParameters(); 4295} 4296 4297 4298AudioFlinger::RecordThread::~RecordThread() 4299{ 4300 delete[] mRsmpInBuffer; 4301 delete mResampler; 4302 delete[] mRsmpOutBuffer; 4303} 4304 4305void AudioFlinger::RecordThread::onFirstRef() 4306{ 4307 run(mName, PRIORITY_URGENT_AUDIO); 4308} 4309 4310status_t AudioFlinger::RecordThread::readyToRun() 4311{ 4312 status_t status = initCheck(); 4313 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4314 return status; 4315} 4316 4317bool AudioFlinger::RecordThread::threadLoop() 4318{ 4319 AudioBufferProvider::Buffer buffer; 4320 sp<RecordTrack> activeTrack; 4321 Vector< sp<EffectChain> > effectChains; 4322 4323 nsecs_t lastWarning = 0; 4324 4325 acquireWakeLock(); 4326 4327 // start recording 4328 while (!exitPending()) { 4329 4330 processConfigEvents(); 4331 4332 { // scope for mLock 4333 Mutex::Autolock _l(mLock); 4334 checkForNewParameters_l(); 4335 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4336 if (!mStandby) { 4337 mInput->stream->common.standby(&mInput->stream->common); 4338 mStandby = true; 4339 } 4340 4341 if (exitPending()) break; 4342 4343 releaseWakeLock_l(); 4344 ALOGV("RecordThread: loop stopping"); 4345 // go to sleep 4346 mWaitWorkCV.wait(mLock); 4347 ALOGV("RecordThread: loop starting"); 4348 acquireWakeLock_l(); 4349 continue; 4350 } 4351 if (mActiveTrack != 0) { 4352 if (mActiveTrack->mState == TrackBase::PAUSING) { 4353 if (!mStandby) { 4354 mInput->stream->common.standby(&mInput->stream->common); 4355 mStandby = true; 4356 } 4357 mActiveTrack.clear(); 4358 mStartStopCond.broadcast(); 4359 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4360 if (mReqChannelCount != mActiveTrack->channelCount()) { 4361 mActiveTrack.clear(); 4362 mStartStopCond.broadcast(); 4363 } else if (mBytesRead != 0) { 4364 // record start succeeds only if first read from audio input 4365 // succeeds 4366 if (mBytesRead > 0) { 4367 mActiveTrack->mState = TrackBase::ACTIVE; 4368 } else { 4369 mActiveTrack.clear(); 4370 } 4371 mStartStopCond.broadcast(); 4372 } 4373 mStandby = false; 4374 } 4375 } 4376 lockEffectChains_l(effectChains); 4377 } 4378 4379 if (mActiveTrack != 0) { 4380 if (mActiveTrack->mState != TrackBase::ACTIVE && 4381 mActiveTrack->mState != TrackBase::RESUMING) { 4382 unlockEffectChains(effectChains); 4383 usleep(kRecordThreadSleepUs); 4384 continue; 4385 } 4386 for (size_t i = 0; i < effectChains.size(); i ++) { 4387 effectChains[i]->process_l(); 4388 } 4389 4390 buffer.frameCount = mFrameCount; 4391 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4392 size_t framesOut = buffer.frameCount; 4393 if (mResampler == NULL) { 4394 // no resampling 4395 while (framesOut) { 4396 size_t framesIn = mFrameCount - mRsmpInIndex; 4397 if (framesIn) { 4398 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4399 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4400 if (framesIn > framesOut) 4401 framesIn = framesOut; 4402 mRsmpInIndex += framesIn; 4403 framesOut -= framesIn; 4404 if ((int)mChannelCount == mReqChannelCount || 4405 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4406 memcpy(dst, src, framesIn * mFrameSize); 4407 } else { 4408 int16_t *src16 = (int16_t *)src; 4409 int16_t *dst16 = (int16_t *)dst; 4410 if (mChannelCount == 1) { 4411 while (framesIn--) { 4412 *dst16++ = *src16; 4413 *dst16++ = *src16++; 4414 } 4415 } else { 4416 while (framesIn--) { 4417 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4418 src16 += 2; 4419 } 4420 } 4421 } 4422 } 4423 if (framesOut && mFrameCount == mRsmpInIndex) { 4424 if (framesOut == mFrameCount && 4425 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4426 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4427 framesOut = 0; 4428 } else { 4429 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4430 mRsmpInIndex = 0; 4431 } 4432 if (mBytesRead < 0) { 4433 ALOGE("Error reading audio input"); 4434 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4435 // Force input into standby so that it tries to 4436 // recover at next read attempt 4437 mInput->stream->common.standby(&mInput->stream->common); 4438 usleep(kRecordThreadSleepUs); 4439 } 4440 mRsmpInIndex = mFrameCount; 4441 framesOut = 0; 4442 buffer.frameCount = 0; 4443 } 4444 } 4445 } 4446 } else { 4447 // resampling 4448 4449 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4450 // alter output frame count as if we were expecting stereo samples 4451 if (mChannelCount == 1 && mReqChannelCount == 1) { 4452 framesOut >>= 1; 4453 } 4454 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4455 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4456 // are 32 bit aligned which should be always true. 4457 if (mChannelCount == 2 && mReqChannelCount == 1) { 4458 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4459 // the resampler always outputs stereo samples: do post stereo to mono conversion 4460 int16_t *src = (int16_t *)mRsmpOutBuffer; 4461 int16_t *dst = buffer.i16; 4462 while (framesOut--) { 4463 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4464 src += 2; 4465 } 4466 } else { 4467 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4468 } 4469 4470 } 4471 mActiveTrack->releaseBuffer(&buffer); 4472 mActiveTrack->overflow(); 4473 } 4474 // client isn't retrieving buffers fast enough 4475 else { 4476 if (!mActiveTrack->setOverflow()) { 4477 nsecs_t now = systemTime(); 4478 if ((now - lastWarning) > kWarningThrottleNs) { 4479 ALOGW("RecordThread: buffer overflow"); 4480 lastWarning = now; 4481 } 4482 } 4483 // Release the processor for a while before asking for a new buffer. 4484 // This will give the application more chance to read from the buffer and 4485 // clear the overflow. 4486 usleep(kRecordThreadSleepUs); 4487 } 4488 } 4489 // enable changes in effect chain 4490 unlockEffectChains(effectChains); 4491 effectChains.clear(); 4492 } 4493 4494 if (!mStandby) { 4495 mInput->stream->common.standby(&mInput->stream->common); 4496 } 4497 mActiveTrack.clear(); 4498 4499 mStartStopCond.broadcast(); 4500 4501 releaseWakeLock(); 4502 4503 ALOGV("RecordThread %p exiting", this); 4504 return false; 4505} 4506 4507 4508sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4509 const sp<AudioFlinger::Client>& client, 4510 uint32_t sampleRate, 4511 audio_format_t format, 4512 int channelMask, 4513 int frameCount, 4514 uint32_t flags, 4515 int sessionId, 4516 status_t *status) 4517{ 4518 sp<RecordTrack> track; 4519 status_t lStatus; 4520 4521 lStatus = initCheck(); 4522 if (lStatus != NO_ERROR) { 4523 ALOGE("Audio driver not initialized."); 4524 goto Exit; 4525 } 4526 4527 { // scope for mLock 4528 Mutex::Autolock _l(mLock); 4529 4530 track = new RecordTrack(this, client, sampleRate, 4531 format, channelMask, frameCount, flags, sessionId); 4532 4533 if (track->getCblk() == 0) { 4534 lStatus = NO_MEMORY; 4535 goto Exit; 4536 } 4537 4538 mTrack = track.get(); 4539 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4540 bool suspend = audio_is_bluetooth_sco_device( 4541 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4542 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4543 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4544 } 4545 lStatus = NO_ERROR; 4546 4547Exit: 4548 if (status) { 4549 *status = lStatus; 4550 } 4551 return track; 4552} 4553 4554status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4555{ 4556 ALOGV("RecordThread::start"); 4557 sp <ThreadBase> strongMe = this; 4558 status_t status = NO_ERROR; 4559 { 4560 AutoMutex lock(mLock); 4561 if (mActiveTrack != 0) { 4562 if (recordTrack != mActiveTrack.get()) { 4563 status = -EBUSY; 4564 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4565 mActiveTrack->mState = TrackBase::ACTIVE; 4566 } 4567 return status; 4568 } 4569 4570 recordTrack->mState = TrackBase::IDLE; 4571 mActiveTrack = recordTrack; 4572 mLock.unlock(); 4573 status_t status = AudioSystem::startInput(mId); 4574 mLock.lock(); 4575 if (status != NO_ERROR) { 4576 mActiveTrack.clear(); 4577 return status; 4578 } 4579 mRsmpInIndex = mFrameCount; 4580 mBytesRead = 0; 4581 if (mResampler != NULL) { 4582 mResampler->reset(); 4583 } 4584 mActiveTrack->mState = TrackBase::RESUMING; 4585 // signal thread to start 4586 ALOGV("Signal record thread"); 4587 mWaitWorkCV.signal(); 4588 // do not wait for mStartStopCond if exiting 4589 if (mExiting) { 4590 mActiveTrack.clear(); 4591 status = INVALID_OPERATION; 4592 goto startError; 4593 } 4594 mStartStopCond.wait(mLock); 4595 if (mActiveTrack == 0) { 4596 ALOGV("Record failed to start"); 4597 status = BAD_VALUE; 4598 goto startError; 4599 } 4600 ALOGV("Record started OK"); 4601 return status; 4602 } 4603startError: 4604 AudioSystem::stopInput(mId); 4605 return status; 4606} 4607 4608void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4609 ALOGV("RecordThread::stop"); 4610 sp <ThreadBase> strongMe = this; 4611 { 4612 AutoMutex lock(mLock); 4613 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4614 mActiveTrack->mState = TrackBase::PAUSING; 4615 // do not wait for mStartStopCond if exiting 4616 if (mExiting) { 4617 return; 4618 } 4619 mStartStopCond.wait(mLock); 4620 // if we have been restarted, recordTrack == mActiveTrack.get() here 4621 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4622 mLock.unlock(); 4623 AudioSystem::stopInput(mId); 4624 mLock.lock(); 4625 ALOGV("Record stopped OK"); 4626 } 4627 } 4628 } 4629} 4630 4631status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4632{ 4633 const size_t SIZE = 256; 4634 char buffer[SIZE]; 4635 String8 result; 4636 pid_t pid = 0; 4637 4638 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4639 result.append(buffer); 4640 4641 if (mActiveTrack != 0) { 4642 result.append("Active Track:\n"); 4643 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4644 mActiveTrack->dump(buffer, SIZE); 4645 result.append(buffer); 4646 4647 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4648 result.append(buffer); 4649 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4650 result.append(buffer); 4651 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4652 result.append(buffer); 4653 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4654 result.append(buffer); 4655 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4656 result.append(buffer); 4657 4658 4659 } else { 4660 result.append("No record client\n"); 4661 } 4662 write(fd, result.string(), result.size()); 4663 4664 dumpBase(fd, args); 4665 dumpEffectChains(fd, args); 4666 4667 return NO_ERROR; 4668} 4669 4670status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4671{ 4672 size_t framesReq = buffer->frameCount; 4673 size_t framesReady = mFrameCount - mRsmpInIndex; 4674 int channelCount; 4675 4676 if (framesReady == 0) { 4677 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4678 if (mBytesRead < 0) { 4679 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4680 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4681 // Force input into standby so that it tries to 4682 // recover at next read attempt 4683 mInput->stream->common.standby(&mInput->stream->common); 4684 usleep(kRecordThreadSleepUs); 4685 } 4686 buffer->raw = NULL; 4687 buffer->frameCount = 0; 4688 return NOT_ENOUGH_DATA; 4689 } 4690 mRsmpInIndex = 0; 4691 framesReady = mFrameCount; 4692 } 4693 4694 if (framesReq > framesReady) { 4695 framesReq = framesReady; 4696 } 4697 4698 if (mChannelCount == 1 && mReqChannelCount == 2) { 4699 channelCount = 1; 4700 } else { 4701 channelCount = 2; 4702 } 4703 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4704 buffer->frameCount = framesReq; 4705 return NO_ERROR; 4706} 4707 4708void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4709{ 4710 mRsmpInIndex += buffer->frameCount; 4711 buffer->frameCount = 0; 4712} 4713 4714bool AudioFlinger::RecordThread::checkForNewParameters_l() 4715{ 4716 bool reconfig = false; 4717 4718 while (!mNewParameters.isEmpty()) { 4719 status_t status = NO_ERROR; 4720 String8 keyValuePair = mNewParameters[0]; 4721 AudioParameter param = AudioParameter(keyValuePair); 4722 int value; 4723 audio_format_t reqFormat = mFormat; 4724 int reqSamplingRate = mReqSampleRate; 4725 int reqChannelCount = mReqChannelCount; 4726 4727 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4728 reqSamplingRate = value; 4729 reconfig = true; 4730 } 4731 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4732 reqFormat = (audio_format_t) value; 4733 reconfig = true; 4734 } 4735 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4736 reqChannelCount = popcount(value); 4737 reconfig = true; 4738 } 4739 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4740 // do not accept frame count changes if tracks are open as the track buffer 4741 // size depends on frame count and correct behavior would not be garantied 4742 // if frame count is changed after track creation 4743 if (mActiveTrack != 0) { 4744 status = INVALID_OPERATION; 4745 } else { 4746 reconfig = true; 4747 } 4748 } 4749 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4750 // forward device change to effects that have requested to be 4751 // aware of attached audio device. 4752 for (size_t i = 0; i < mEffectChains.size(); i++) { 4753 mEffectChains[i]->setDevice_l(value); 4754 } 4755 // store input device and output device but do not forward output device to audio HAL. 4756 // Note that status is ignored by the caller for output device 4757 // (see AudioFlinger::setParameters() 4758 if (value & AUDIO_DEVICE_OUT_ALL) { 4759 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4760 status = BAD_VALUE; 4761 } else { 4762 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4763 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4764 if (mTrack != NULL) { 4765 bool suspend = audio_is_bluetooth_sco_device( 4766 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4767 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4768 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4769 } 4770 } 4771 mDevice |= (uint32_t)value; 4772 } 4773 if (status == NO_ERROR) { 4774 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4775 if (status == INVALID_OPERATION) { 4776 mInput->stream->common.standby(&mInput->stream->common); 4777 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4778 } 4779 if (reconfig) { 4780 if (status == BAD_VALUE && 4781 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4782 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4783 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4784 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4785 (reqChannelCount < 3)) { 4786 status = NO_ERROR; 4787 } 4788 if (status == NO_ERROR) { 4789 readInputParameters(); 4790 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4791 } 4792 } 4793 } 4794 4795 mNewParameters.removeAt(0); 4796 4797 mParamStatus = status; 4798 mParamCond.signal(); 4799 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4800 // already timed out waiting for the status and will never signal the condition. 4801 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4802 } 4803 return reconfig; 4804} 4805 4806String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4807{ 4808 char *s; 4809 String8 out_s8 = String8(); 4810 4811 Mutex::Autolock _l(mLock); 4812 if (initCheck() != NO_ERROR) { 4813 return out_s8; 4814 } 4815 4816 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4817 out_s8 = String8(s); 4818 free(s); 4819 return out_s8; 4820} 4821 4822void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4823 AudioSystem::OutputDescriptor desc; 4824 void *param2 = NULL; 4825 4826 switch (event) { 4827 case AudioSystem::INPUT_OPENED: 4828 case AudioSystem::INPUT_CONFIG_CHANGED: 4829 desc.channels = mChannelMask; 4830 desc.samplingRate = mSampleRate; 4831 desc.format = mFormat; 4832 desc.frameCount = mFrameCount; 4833 desc.latency = 0; 4834 param2 = &desc; 4835 break; 4836 4837 case AudioSystem::INPUT_CLOSED: 4838 default: 4839 break; 4840 } 4841 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4842} 4843 4844void AudioFlinger::RecordThread::readInputParameters() 4845{ 4846 delete mRsmpInBuffer; 4847 // mRsmpInBuffer is always assigned a new[] below 4848 delete mRsmpOutBuffer; 4849 mRsmpOutBuffer = NULL; 4850 delete mResampler; 4851 mResampler = NULL; 4852 4853 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4854 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4855 mChannelCount = (uint16_t)popcount(mChannelMask); 4856 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4857 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4858 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4859 mFrameCount = mInputBytes / mFrameSize; 4860 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4861 4862 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4863 { 4864 int channelCount; 4865 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4866 // stereo to mono post process as the resampler always outputs stereo. 4867 if (mChannelCount == 1 && mReqChannelCount == 2) { 4868 channelCount = 1; 4869 } else { 4870 channelCount = 2; 4871 } 4872 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4873 mResampler->setSampleRate(mSampleRate); 4874 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4875 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4876 4877 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4878 if (mChannelCount == 1 && mReqChannelCount == 1) { 4879 mFrameCount >>= 1; 4880 } 4881 4882 } 4883 mRsmpInIndex = mFrameCount; 4884} 4885 4886unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4887{ 4888 Mutex::Autolock _l(mLock); 4889 if (initCheck() != NO_ERROR) { 4890 return 0; 4891 } 4892 4893 return mInput->stream->get_input_frames_lost(mInput->stream); 4894} 4895 4896uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4897{ 4898 Mutex::Autolock _l(mLock); 4899 uint32_t result = 0; 4900 if (getEffectChain_l(sessionId) != 0) { 4901 result = EFFECT_SESSION; 4902 } 4903 4904 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4905 result |= TRACK_SESSION; 4906 } 4907 4908 return result; 4909} 4910 4911AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4912{ 4913 Mutex::Autolock _l(mLock); 4914 return mTrack; 4915} 4916 4917AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4918{ 4919 Mutex::Autolock _l(mLock); 4920 return mInput; 4921} 4922 4923AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4924{ 4925 Mutex::Autolock _l(mLock); 4926 AudioStreamIn *input = mInput; 4927 mInput = NULL; 4928 return input; 4929} 4930 4931// this method must always be called either with ThreadBase mLock held or inside the thread loop 4932audio_stream_t* AudioFlinger::RecordThread::stream() 4933{ 4934 if (mInput == NULL) { 4935 return NULL; 4936 } 4937 return &mInput->stream->common; 4938} 4939 4940 4941// ---------------------------------------------------------------------------- 4942 4943int AudioFlinger::openOutput(uint32_t *pDevices, 4944 uint32_t *pSamplingRate, 4945 audio_format_t *pFormat, 4946 uint32_t *pChannels, 4947 uint32_t *pLatencyMs, 4948 uint32_t flags) 4949{ 4950 status_t status; 4951 PlaybackThread *thread = NULL; 4952 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4953 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4954 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4955 uint32_t channels = pChannels ? *pChannels : 0; 4956 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4957 audio_stream_out_t *outStream; 4958 audio_hw_device_t *outHwDev; 4959 4960 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4961 pDevices ? *pDevices : 0, 4962 samplingRate, 4963 format, 4964 channels, 4965 flags); 4966 4967 if (pDevices == NULL || *pDevices == 0) { 4968 return 0; 4969 } 4970 4971 Mutex::Autolock _l(mLock); 4972 4973 outHwDev = findSuitableHwDev_l(*pDevices); 4974 if (outHwDev == NULL) 4975 return 0; 4976 4977 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4978 &channels, &samplingRate, &outStream); 4979 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4980 outStream, 4981 samplingRate, 4982 format, 4983 channels, 4984 status); 4985 4986 mHardwareStatus = AUDIO_HW_IDLE; 4987 if (outStream != NULL) { 4988 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4989 int id = nextUniqueId(); 4990 4991 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4992 (format != AUDIO_FORMAT_PCM_16_BIT) || 4993 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4994 thread = new DirectOutputThread(this, output, id, *pDevices); 4995 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4996 } else { 4997 thread = new MixerThread(this, output, id, *pDevices); 4998 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4999 } 5000 mPlaybackThreads.add(id, thread); 5001 5002 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5003 if (pFormat != NULL) *pFormat = format; 5004 if (pChannels != NULL) *pChannels = channels; 5005 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5006 5007 // notify client processes of the new output creation 5008 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5009 return id; 5010 } 5011 5012 return 0; 5013} 5014 5015int AudioFlinger::openDuplicateOutput(int output1, int output2) 5016{ 5017 Mutex::Autolock _l(mLock); 5018 MixerThread *thread1 = checkMixerThread_l(output1); 5019 MixerThread *thread2 = checkMixerThread_l(output2); 5020 5021 if (thread1 == NULL || thread2 == NULL) { 5022 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5023 return 0; 5024 } 5025 5026 int id = nextUniqueId(); 5027 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5028 thread->addOutputTrack(thread2); 5029 mPlaybackThreads.add(id, thread); 5030 // notify client processes of the new output creation 5031 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5032 return id; 5033} 5034 5035status_t AudioFlinger::closeOutput(int output) 5036{ 5037 // keep strong reference on the playback thread so that 5038 // it is not destroyed while exit() is executed 5039 sp <PlaybackThread> thread; 5040 { 5041 Mutex::Autolock _l(mLock); 5042 thread = checkPlaybackThread_l(output); 5043 if (thread == NULL) { 5044 return BAD_VALUE; 5045 } 5046 5047 ALOGV("closeOutput() %d", output); 5048 5049 if (thread->type() == ThreadBase::MIXER) { 5050 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5051 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5052 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5053 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5054 } 5055 } 5056 } 5057 void *param2 = NULL; 5058 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5059 mPlaybackThreads.removeItem(output); 5060 } 5061 thread->exit(); 5062 5063 if (thread->type() != ThreadBase::DUPLICATING) { 5064 AudioStreamOut *out = thread->clearOutput(); 5065 assert(out != NULL); 5066 // from now on thread->mOutput is NULL 5067 out->hwDev->close_output_stream(out->hwDev, out->stream); 5068 delete out; 5069 } 5070 return NO_ERROR; 5071} 5072 5073status_t AudioFlinger::suspendOutput(int output) 5074{ 5075 Mutex::Autolock _l(mLock); 5076 PlaybackThread *thread = checkPlaybackThread_l(output); 5077 5078 if (thread == NULL) { 5079 return BAD_VALUE; 5080 } 5081 5082 ALOGV("suspendOutput() %d", output); 5083 thread->suspend(); 5084 5085 return NO_ERROR; 5086} 5087 5088status_t AudioFlinger::restoreOutput(int output) 5089{ 5090 Mutex::Autolock _l(mLock); 5091 PlaybackThread *thread = checkPlaybackThread_l(output); 5092 5093 if (thread == NULL) { 5094 return BAD_VALUE; 5095 } 5096 5097 ALOGV("restoreOutput() %d", output); 5098 5099 thread->restore(); 5100 5101 return NO_ERROR; 5102} 5103 5104int AudioFlinger::openInput(uint32_t *pDevices, 5105 uint32_t *pSamplingRate, 5106 audio_format_t *pFormat, 5107 uint32_t *pChannels, 5108 audio_in_acoustics_t acoustics) 5109{ 5110 status_t status; 5111 RecordThread *thread = NULL; 5112 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5113 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5114 uint32_t channels = pChannels ? *pChannels : 0; 5115 uint32_t reqSamplingRate = samplingRate; 5116 audio_format_t reqFormat = format; 5117 uint32_t reqChannels = channels; 5118 audio_stream_in_t *inStream; 5119 audio_hw_device_t *inHwDev; 5120 5121 if (pDevices == NULL || *pDevices == 0) { 5122 return 0; 5123 } 5124 5125 Mutex::Autolock _l(mLock); 5126 5127 inHwDev = findSuitableHwDev_l(*pDevices); 5128 if (inHwDev == NULL) 5129 return 0; 5130 5131 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5132 &channels, &samplingRate, 5133 acoustics, 5134 &inStream); 5135 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5136 inStream, 5137 samplingRate, 5138 format, 5139 channels, 5140 acoustics, 5141 status); 5142 5143 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5144 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5145 // or stereo to mono conversions on 16 bit PCM inputs. 5146 if (inStream == NULL && status == BAD_VALUE && 5147 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5148 (samplingRate <= 2 * reqSamplingRate) && 5149 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5150 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5151 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5152 &channels, &samplingRate, 5153 acoustics, 5154 &inStream); 5155 } 5156 5157 if (inStream != NULL) { 5158 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5159 5160 int id = nextUniqueId(); 5161 // Start record thread 5162 // RecorThread require both input and output device indication to forward to audio 5163 // pre processing modules 5164 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5165 thread = new RecordThread(this, 5166 input, 5167 reqSamplingRate, 5168 reqChannels, 5169 id, 5170 device); 5171 mRecordThreads.add(id, thread); 5172 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5173 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5174 if (pFormat != NULL) *pFormat = format; 5175 if (pChannels != NULL) *pChannels = reqChannels; 5176 5177 input->stream->common.standby(&input->stream->common); 5178 5179 // notify client processes of the new input creation 5180 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5181 return id; 5182 } 5183 5184 return 0; 5185} 5186 5187status_t AudioFlinger::closeInput(int input) 5188{ 5189 // keep strong reference on the record thread so that 5190 // it is not destroyed while exit() is executed 5191 sp <RecordThread> thread; 5192 { 5193 Mutex::Autolock _l(mLock); 5194 thread = checkRecordThread_l(input); 5195 if (thread == NULL) { 5196 return BAD_VALUE; 5197 } 5198 5199 ALOGV("closeInput() %d", input); 5200 void *param2 = NULL; 5201 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5202 mRecordThreads.removeItem(input); 5203 } 5204 thread->exit(); 5205 5206 AudioStreamIn *in = thread->clearInput(); 5207 assert(in != NULL); 5208 // from now on thread->mInput is NULL 5209 in->hwDev->close_input_stream(in->hwDev, in->stream); 5210 delete in; 5211 5212 return NO_ERROR; 5213} 5214 5215status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5216{ 5217 Mutex::Autolock _l(mLock); 5218 MixerThread *dstThread = checkMixerThread_l(output); 5219 if (dstThread == NULL) { 5220 ALOGW("setStreamOutput() bad output id %d", output); 5221 return BAD_VALUE; 5222 } 5223 5224 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5225 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5226 5227 dstThread->setStreamValid(stream, true); 5228 5229 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5230 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5231 if (thread != dstThread && 5232 thread->type() != ThreadBase::DIRECT) { 5233 MixerThread *srcThread = (MixerThread *)thread; 5234 srcThread->setStreamValid(stream, false); 5235 srcThread->invalidateTracks(stream); 5236 } 5237 } 5238 5239 return NO_ERROR; 5240} 5241 5242 5243int AudioFlinger::newAudioSessionId() 5244{ 5245 return nextUniqueId(); 5246} 5247 5248void AudioFlinger::acquireAudioSessionId(int audioSession) 5249{ 5250 Mutex::Autolock _l(mLock); 5251 int caller = IPCThreadState::self()->getCallingPid(); 5252 ALOGV("acquiring %d from %d", audioSession, caller); 5253 int num = mAudioSessionRefs.size(); 5254 for (int i = 0; i< num; i++) { 5255 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5256 if (ref->sessionid == audioSession && ref->pid == caller) { 5257 ref->cnt++; 5258 ALOGV(" incremented refcount to %d", ref->cnt); 5259 return; 5260 } 5261 } 5262 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5263 ALOGV(" added new entry for %d", audioSession); 5264} 5265 5266void AudioFlinger::releaseAudioSessionId(int audioSession) 5267{ 5268 Mutex::Autolock _l(mLock); 5269 int caller = IPCThreadState::self()->getCallingPid(); 5270 ALOGV("releasing %d from %d", audioSession, caller); 5271 int num = mAudioSessionRefs.size(); 5272 for (int i = 0; i< num; i++) { 5273 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5274 if (ref->sessionid == audioSession && ref->pid == caller) { 5275 ref->cnt--; 5276 ALOGV(" decremented refcount to %d", ref->cnt); 5277 if (ref->cnt == 0) { 5278 mAudioSessionRefs.removeAt(i); 5279 delete ref; 5280 purgeStaleEffects_l(); 5281 } 5282 return; 5283 } 5284 } 5285 ALOGW("session id %d not found for pid %d", audioSession, caller); 5286} 5287 5288void AudioFlinger::purgeStaleEffects_l() { 5289 5290 ALOGV("purging stale effects"); 5291 5292 Vector< sp<EffectChain> > chains; 5293 5294 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5295 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5296 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5297 sp<EffectChain> ec = t->mEffectChains[j]; 5298 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5299 chains.push(ec); 5300 } 5301 } 5302 } 5303 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5304 sp<RecordThread> t = mRecordThreads.valueAt(i); 5305 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5306 sp<EffectChain> ec = t->mEffectChains[j]; 5307 chains.push(ec); 5308 } 5309 } 5310 5311 for (size_t i = 0; i < chains.size(); i++) { 5312 sp<EffectChain> ec = chains[i]; 5313 int sessionid = ec->sessionId(); 5314 sp<ThreadBase> t = ec->mThread.promote(); 5315 if (t == 0) { 5316 continue; 5317 } 5318 size_t numsessionrefs = mAudioSessionRefs.size(); 5319 bool found = false; 5320 for (size_t k = 0; k < numsessionrefs; k++) { 5321 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5322 if (ref->sessionid == sessionid) { 5323 ALOGV(" session %d still exists for %d with %d refs", 5324 sessionid, ref->pid, ref->cnt); 5325 found = true; 5326 break; 5327 } 5328 } 5329 if (!found) { 5330 // remove all effects from the chain 5331 while (ec->mEffects.size()) { 5332 sp<EffectModule> effect = ec->mEffects[0]; 5333 effect->unPin(); 5334 Mutex::Autolock _l (t->mLock); 5335 t->removeEffect_l(effect); 5336 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5337 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5338 if (handle != 0) { 5339 handle->mEffect.clear(); 5340 if (handle->mHasControl && handle->mEnabled) { 5341 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5342 } 5343 } 5344 } 5345 AudioSystem::unregisterEffect(effect->id()); 5346 } 5347 } 5348 } 5349 return; 5350} 5351 5352// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5353AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5354{ 5355 PlaybackThread *thread = NULL; 5356 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5357 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5358 } 5359 return thread; 5360} 5361 5362// checkMixerThread_l() must be called with AudioFlinger::mLock held 5363AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5364{ 5365 PlaybackThread *thread = checkPlaybackThread_l(output); 5366 if (thread != NULL) { 5367 if (thread->type() == ThreadBase::DIRECT) { 5368 thread = NULL; 5369 } 5370 } 5371 return (MixerThread *)thread; 5372} 5373 5374// checkRecordThread_l() must be called with AudioFlinger::mLock held 5375AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5376{ 5377 RecordThread *thread = NULL; 5378 if (mRecordThreads.indexOfKey(input) >= 0) { 5379 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5380 } 5381 return thread; 5382} 5383 5384uint32_t AudioFlinger::nextUniqueId() 5385{ 5386 return android_atomic_inc(&mNextUniqueId); 5387} 5388 5389AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5390{ 5391 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5392 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5393 AudioStreamOut *output = thread->getOutput(); 5394 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5395 return thread; 5396 } 5397 } 5398 return NULL; 5399} 5400 5401uint32_t AudioFlinger::primaryOutputDevice_l() 5402{ 5403 PlaybackThread *thread = primaryPlaybackThread_l(); 5404 5405 if (thread == NULL) { 5406 return 0; 5407 } 5408 5409 return thread->device(); 5410} 5411 5412 5413// ---------------------------------------------------------------------------- 5414// Effect management 5415// ---------------------------------------------------------------------------- 5416 5417 5418status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5419{ 5420 Mutex::Autolock _l(mLock); 5421 return EffectQueryNumberEffects(numEffects); 5422} 5423 5424status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5425{ 5426 Mutex::Autolock _l(mLock); 5427 return EffectQueryEffect(index, descriptor); 5428} 5429 5430status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5431{ 5432 Mutex::Autolock _l(mLock); 5433 return EffectGetDescriptor(pUuid, descriptor); 5434} 5435 5436 5437sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5438 effect_descriptor_t *pDesc, 5439 const sp<IEffectClient>& effectClient, 5440 int32_t priority, 5441 int io, 5442 int sessionId, 5443 status_t *status, 5444 int *id, 5445 int *enabled) 5446{ 5447 status_t lStatus = NO_ERROR; 5448 sp<EffectHandle> handle; 5449 effect_descriptor_t desc; 5450 sp<Client> client; 5451 wp<Client> wclient; 5452 5453 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5454 pid, effectClient.get(), priority, sessionId, io); 5455 5456 if (pDesc == NULL) { 5457 lStatus = BAD_VALUE; 5458 goto Exit; 5459 } 5460 5461 // check audio settings permission for global effects 5462 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5463 lStatus = PERMISSION_DENIED; 5464 goto Exit; 5465 } 5466 5467 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5468 // that can only be created by audio policy manager (running in same process) 5469 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5470 lStatus = PERMISSION_DENIED; 5471 goto Exit; 5472 } 5473 5474 if (io == 0) { 5475 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5476 // output must be specified by AudioPolicyManager when using session 5477 // AUDIO_SESSION_OUTPUT_STAGE 5478 lStatus = BAD_VALUE; 5479 goto Exit; 5480 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5481 // if the output returned by getOutputForEffect() is removed before we lock the 5482 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5483 // and we will exit safely 5484 io = AudioSystem::getOutputForEffect(&desc); 5485 } 5486 } 5487 5488 { 5489 Mutex::Autolock _l(mLock); 5490 5491 5492 if (!EffectIsNullUuid(&pDesc->uuid)) { 5493 // if uuid is specified, request effect descriptor 5494 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5495 if (lStatus < 0) { 5496 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5497 goto Exit; 5498 } 5499 } else { 5500 // if uuid is not specified, look for an available implementation 5501 // of the required type in effect factory 5502 if (EffectIsNullUuid(&pDesc->type)) { 5503 ALOGW("createEffect() no effect type"); 5504 lStatus = BAD_VALUE; 5505 goto Exit; 5506 } 5507 uint32_t numEffects = 0; 5508 effect_descriptor_t d; 5509 d.flags = 0; // prevent compiler warning 5510 bool found = false; 5511 5512 lStatus = EffectQueryNumberEffects(&numEffects); 5513 if (lStatus < 0) { 5514 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5515 goto Exit; 5516 } 5517 for (uint32_t i = 0; i < numEffects; i++) { 5518 lStatus = EffectQueryEffect(i, &desc); 5519 if (lStatus < 0) { 5520 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5521 continue; 5522 } 5523 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5524 // If matching type found save effect descriptor. If the session is 5525 // 0 and the effect is not auxiliary, continue enumeration in case 5526 // an auxiliary version of this effect type is available 5527 found = true; 5528 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5529 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5530 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5531 break; 5532 } 5533 } 5534 } 5535 if (!found) { 5536 lStatus = BAD_VALUE; 5537 ALOGW("createEffect() effect not found"); 5538 goto Exit; 5539 } 5540 // For same effect type, chose auxiliary version over insert version if 5541 // connect to output mix (Compliance to OpenSL ES) 5542 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5543 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5544 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5545 } 5546 } 5547 5548 // Do not allow auxiliary effects on a session different from 0 (output mix) 5549 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5550 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5551 lStatus = INVALID_OPERATION; 5552 goto Exit; 5553 } 5554 5555 // check recording permission for visualizer 5556 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5557 !recordingAllowed()) { 5558 lStatus = PERMISSION_DENIED; 5559 goto Exit; 5560 } 5561 5562 // return effect descriptor 5563 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5564 5565 // If output is not specified try to find a matching audio session ID in one of the 5566 // output threads. 5567 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5568 // because of code checking output when entering the function. 5569 // Note: io is never 0 when creating an effect on an input 5570 if (io == 0) { 5571 // look for the thread where the specified audio session is present 5572 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5573 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5574 io = mPlaybackThreads.keyAt(i); 5575 break; 5576 } 5577 } 5578 if (io == 0) { 5579 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5580 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5581 io = mRecordThreads.keyAt(i); 5582 break; 5583 } 5584 } 5585 } 5586 // If no output thread contains the requested session ID, default to 5587 // first output. The effect chain will be moved to the correct output 5588 // thread when a track with the same session ID is created 5589 if (io == 0 && mPlaybackThreads.size()) { 5590 io = mPlaybackThreads.keyAt(0); 5591 } 5592 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5593 } 5594 ThreadBase *thread = checkRecordThread_l(io); 5595 if (thread == NULL) { 5596 thread = checkPlaybackThread_l(io); 5597 if (thread == NULL) { 5598 ALOGE("createEffect() unknown output thread"); 5599 lStatus = BAD_VALUE; 5600 goto Exit; 5601 } 5602 } 5603 5604 wclient = mClients.valueFor(pid); 5605 5606 if (wclient != NULL) { 5607 client = wclient.promote(); 5608 } else { 5609 client = new Client(this, pid); 5610 mClients.add(pid, client); 5611 } 5612 5613 // create effect on selected output thread 5614 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5615 &desc, enabled, &lStatus); 5616 if (handle != 0 && id != NULL) { 5617 *id = handle->id(); 5618 } 5619 } 5620 5621Exit: 5622 if(status) { 5623 *status = lStatus; 5624 } 5625 return handle; 5626} 5627 5628status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5629{ 5630 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5631 sessionId, srcOutput, dstOutput); 5632 Mutex::Autolock _l(mLock); 5633 if (srcOutput == dstOutput) { 5634 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5635 return NO_ERROR; 5636 } 5637 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5638 if (srcThread == NULL) { 5639 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5640 return BAD_VALUE; 5641 } 5642 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5643 if (dstThread == NULL) { 5644 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5645 return BAD_VALUE; 5646 } 5647 5648 Mutex::Autolock _dl(dstThread->mLock); 5649 Mutex::Autolock _sl(srcThread->mLock); 5650 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5651 5652 return NO_ERROR; 5653} 5654 5655// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5656status_t AudioFlinger::moveEffectChain_l(int sessionId, 5657 AudioFlinger::PlaybackThread *srcThread, 5658 AudioFlinger::PlaybackThread *dstThread, 5659 bool reRegister) 5660{ 5661 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5662 sessionId, srcThread, dstThread); 5663 5664 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5665 if (chain == 0) { 5666 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5667 sessionId, srcThread); 5668 return INVALID_OPERATION; 5669 } 5670 5671 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5672 // so that a new chain is created with correct parameters when first effect is added. This is 5673 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5674 // removed. 5675 srcThread->removeEffectChain_l(chain); 5676 5677 // transfer all effects one by one so that new effect chain is created on new thread with 5678 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5679 int dstOutput = dstThread->id(); 5680 sp<EffectChain> dstChain; 5681 uint32_t strategy = 0; // prevent compiler warning 5682 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5683 while (effect != 0) { 5684 srcThread->removeEffect_l(effect); 5685 dstThread->addEffect_l(effect); 5686 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5687 if (effect->state() == EffectModule::ACTIVE || 5688 effect->state() == EffectModule::STOPPING) { 5689 effect->start(); 5690 } 5691 // if the move request is not received from audio policy manager, the effect must be 5692 // re-registered with the new strategy and output 5693 if (dstChain == 0) { 5694 dstChain = effect->chain().promote(); 5695 if (dstChain == 0) { 5696 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5697 srcThread->addEffect_l(effect); 5698 return NO_INIT; 5699 } 5700 strategy = dstChain->strategy(); 5701 } 5702 if (reRegister) { 5703 AudioSystem::unregisterEffect(effect->id()); 5704 AudioSystem::registerEffect(&effect->desc(), 5705 dstOutput, 5706 strategy, 5707 sessionId, 5708 effect->id()); 5709 } 5710 effect = chain->getEffectFromId_l(0); 5711 } 5712 5713 return NO_ERROR; 5714} 5715 5716 5717// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5718sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5719 const sp<AudioFlinger::Client>& client, 5720 const sp<IEffectClient>& effectClient, 5721 int32_t priority, 5722 int sessionId, 5723 effect_descriptor_t *desc, 5724 int *enabled, 5725 status_t *status 5726 ) 5727{ 5728 sp<EffectModule> effect; 5729 sp<EffectHandle> handle; 5730 status_t lStatus; 5731 sp<EffectChain> chain; 5732 bool chainCreated = false; 5733 bool effectCreated = false; 5734 bool effectRegistered = false; 5735 5736 lStatus = initCheck(); 5737 if (lStatus != NO_ERROR) { 5738 ALOGW("createEffect_l() Audio driver not initialized."); 5739 goto Exit; 5740 } 5741 5742 // Do not allow effects with session ID 0 on direct output or duplicating threads 5743 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5744 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5745 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5746 desc->name, sessionId); 5747 lStatus = BAD_VALUE; 5748 goto Exit; 5749 } 5750 // Only Pre processor effects are allowed on input threads and only on input threads 5751 if ((mType == RECORD && 5752 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5753 (mType != RECORD && 5754 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5755 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5756 desc->name, desc->flags, mType); 5757 lStatus = BAD_VALUE; 5758 goto Exit; 5759 } 5760 5761 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5762 5763 { // scope for mLock 5764 Mutex::Autolock _l(mLock); 5765 5766 // check for existing effect chain with the requested audio session 5767 chain = getEffectChain_l(sessionId); 5768 if (chain == 0) { 5769 // create a new chain for this session 5770 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5771 chain = new EffectChain(this, sessionId); 5772 addEffectChain_l(chain); 5773 chain->setStrategy(getStrategyForSession_l(sessionId)); 5774 chainCreated = true; 5775 } else { 5776 effect = chain->getEffectFromDesc_l(desc); 5777 } 5778 5779 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5780 5781 if (effect == 0) { 5782 int id = mAudioFlinger->nextUniqueId(); 5783 // Check CPU and memory usage 5784 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5785 if (lStatus != NO_ERROR) { 5786 goto Exit; 5787 } 5788 effectRegistered = true; 5789 // create a new effect module if none present in the chain 5790 effect = new EffectModule(this, chain, desc, id, sessionId); 5791 lStatus = effect->status(); 5792 if (lStatus != NO_ERROR) { 5793 goto Exit; 5794 } 5795 lStatus = chain->addEffect_l(effect); 5796 if (lStatus != NO_ERROR) { 5797 goto Exit; 5798 } 5799 effectCreated = true; 5800 5801 effect->setDevice(mDevice); 5802 effect->setMode(mAudioFlinger->getMode()); 5803 } 5804 // create effect handle and connect it to effect module 5805 handle = new EffectHandle(effect, client, effectClient, priority); 5806 lStatus = effect->addHandle(handle); 5807 if (enabled != NULL) { 5808 *enabled = (int)effect->isEnabled(); 5809 } 5810 } 5811 5812Exit: 5813 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5814 Mutex::Autolock _l(mLock); 5815 if (effectCreated) { 5816 chain->removeEffect_l(effect); 5817 } 5818 if (effectRegistered) { 5819 AudioSystem::unregisterEffect(effect->id()); 5820 } 5821 if (chainCreated) { 5822 removeEffectChain_l(chain); 5823 } 5824 handle.clear(); 5825 } 5826 5827 if(status) { 5828 *status = lStatus; 5829 } 5830 return handle; 5831} 5832 5833sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5834{ 5835 sp<EffectModule> effect; 5836 5837 sp<EffectChain> chain = getEffectChain_l(sessionId); 5838 if (chain != 0) { 5839 effect = chain->getEffectFromId_l(effectId); 5840 } 5841 return effect; 5842} 5843 5844// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5845// PlaybackThread::mLock held 5846status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5847{ 5848 // check for existing effect chain with the requested audio session 5849 int sessionId = effect->sessionId(); 5850 sp<EffectChain> chain = getEffectChain_l(sessionId); 5851 bool chainCreated = false; 5852 5853 if (chain == 0) { 5854 // create a new chain for this session 5855 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5856 chain = new EffectChain(this, sessionId); 5857 addEffectChain_l(chain); 5858 chain->setStrategy(getStrategyForSession_l(sessionId)); 5859 chainCreated = true; 5860 } 5861 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5862 5863 if (chain->getEffectFromId_l(effect->id()) != 0) { 5864 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5865 this, effect->desc().name, chain.get()); 5866 return BAD_VALUE; 5867 } 5868 5869 status_t status = chain->addEffect_l(effect); 5870 if (status != NO_ERROR) { 5871 if (chainCreated) { 5872 removeEffectChain_l(chain); 5873 } 5874 return status; 5875 } 5876 5877 effect->setDevice(mDevice); 5878 effect->setMode(mAudioFlinger->getMode()); 5879 return NO_ERROR; 5880} 5881 5882void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5883 5884 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5885 effect_descriptor_t desc = effect->desc(); 5886 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5887 detachAuxEffect_l(effect->id()); 5888 } 5889 5890 sp<EffectChain> chain = effect->chain().promote(); 5891 if (chain != 0) { 5892 // remove effect chain if removing last effect 5893 if (chain->removeEffect_l(effect) == 0) { 5894 removeEffectChain_l(chain); 5895 } 5896 } else { 5897 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5898 } 5899} 5900 5901void AudioFlinger::ThreadBase::lockEffectChains_l( 5902 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5903{ 5904 effectChains = mEffectChains; 5905 for (size_t i = 0; i < mEffectChains.size(); i++) { 5906 mEffectChains[i]->lock(); 5907 } 5908} 5909 5910void AudioFlinger::ThreadBase::unlockEffectChains( 5911 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5912{ 5913 for (size_t i = 0; i < effectChains.size(); i++) { 5914 effectChains[i]->unlock(); 5915 } 5916} 5917 5918sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5919{ 5920 Mutex::Autolock _l(mLock); 5921 return getEffectChain_l(sessionId); 5922} 5923 5924sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5925{ 5926 sp<EffectChain> chain; 5927 5928 size_t size = mEffectChains.size(); 5929 for (size_t i = 0; i < size; i++) { 5930 if (mEffectChains[i]->sessionId() == sessionId) { 5931 chain = mEffectChains[i]; 5932 break; 5933 } 5934 } 5935 return chain; 5936} 5937 5938void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5939{ 5940 Mutex::Autolock _l(mLock); 5941 size_t size = mEffectChains.size(); 5942 for (size_t i = 0; i < size; i++) { 5943 mEffectChains[i]->setMode_l(mode); 5944 } 5945} 5946 5947void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5948 const wp<EffectHandle>& handle, 5949 bool unpiniflast) { 5950 5951 Mutex::Autolock _l(mLock); 5952 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5953 // delete the effect module if removing last handle on it 5954 if (effect->removeHandle(handle) == 0) { 5955 if (!effect->isPinned() || unpiniflast) { 5956 removeEffect_l(effect); 5957 AudioSystem::unregisterEffect(effect->id()); 5958 } 5959 } 5960} 5961 5962status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5963{ 5964 int session = chain->sessionId(); 5965 int16_t *buffer = mMixBuffer; 5966 bool ownsBuffer = false; 5967 5968 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5969 if (session > 0) { 5970 // Only one effect chain can be present in direct output thread and it uses 5971 // the mix buffer as input 5972 if (mType != DIRECT) { 5973 size_t numSamples = mFrameCount * mChannelCount; 5974 buffer = new int16_t[numSamples]; 5975 memset(buffer, 0, numSamples * sizeof(int16_t)); 5976 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5977 ownsBuffer = true; 5978 } 5979 5980 // Attach all tracks with same session ID to this chain. 5981 for (size_t i = 0; i < mTracks.size(); ++i) { 5982 sp<Track> track = mTracks[i]; 5983 if (session == track->sessionId()) { 5984 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5985 track->setMainBuffer(buffer); 5986 chain->incTrackCnt(); 5987 } 5988 } 5989 5990 // indicate all active tracks in the chain 5991 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5992 sp<Track> track = mActiveTracks[i].promote(); 5993 if (track == 0) continue; 5994 if (session == track->sessionId()) { 5995 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5996 chain->incActiveTrackCnt(); 5997 } 5998 } 5999 } 6000 6001 chain->setInBuffer(buffer, ownsBuffer); 6002 chain->setOutBuffer(mMixBuffer); 6003 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6004 // chains list in order to be processed last as it contains output stage effects 6005 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6006 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6007 // after track specific effects and before output stage 6008 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6009 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6010 // Effect chain for other sessions are inserted at beginning of effect 6011 // chains list to be processed before output mix effects. Relative order between other 6012 // sessions is not important 6013 size_t size = mEffectChains.size(); 6014 size_t i = 0; 6015 for (i = 0; i < size; i++) { 6016 if (mEffectChains[i]->sessionId() < session) break; 6017 } 6018 mEffectChains.insertAt(chain, i); 6019 checkSuspendOnAddEffectChain_l(chain); 6020 6021 return NO_ERROR; 6022} 6023 6024size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6025{ 6026 int session = chain->sessionId(); 6027 6028 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6029 6030 for (size_t i = 0; i < mEffectChains.size(); i++) { 6031 if (chain == mEffectChains[i]) { 6032 mEffectChains.removeAt(i); 6033 // detach all active tracks from the chain 6034 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6035 sp<Track> track = mActiveTracks[i].promote(); 6036 if (track == 0) continue; 6037 if (session == track->sessionId()) { 6038 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6039 chain.get(), session); 6040 chain->decActiveTrackCnt(); 6041 } 6042 } 6043 6044 // detach all tracks with same session ID from this chain 6045 for (size_t i = 0; i < mTracks.size(); ++i) { 6046 sp<Track> track = mTracks[i]; 6047 if (session == track->sessionId()) { 6048 track->setMainBuffer(mMixBuffer); 6049 chain->decTrackCnt(); 6050 } 6051 } 6052 break; 6053 } 6054 } 6055 return mEffectChains.size(); 6056} 6057 6058status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6059 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6060{ 6061 Mutex::Autolock _l(mLock); 6062 return attachAuxEffect_l(track, EffectId); 6063} 6064 6065status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6066 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6067{ 6068 status_t status = NO_ERROR; 6069 6070 if (EffectId == 0) { 6071 track->setAuxBuffer(0, NULL); 6072 } else { 6073 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6074 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6075 if (effect != 0) { 6076 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6077 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6078 } else { 6079 status = INVALID_OPERATION; 6080 } 6081 } else { 6082 status = BAD_VALUE; 6083 } 6084 } 6085 return status; 6086} 6087 6088void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6089{ 6090 for (size_t i = 0; i < mTracks.size(); ++i) { 6091 sp<Track> track = mTracks[i]; 6092 if (track->auxEffectId() == effectId) { 6093 attachAuxEffect_l(track, 0); 6094 } 6095 } 6096} 6097 6098status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6099{ 6100 // only one chain per input thread 6101 if (mEffectChains.size() != 0) { 6102 return INVALID_OPERATION; 6103 } 6104 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6105 6106 chain->setInBuffer(NULL); 6107 chain->setOutBuffer(NULL); 6108 6109 checkSuspendOnAddEffectChain_l(chain); 6110 6111 mEffectChains.add(chain); 6112 6113 return NO_ERROR; 6114} 6115 6116size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6117{ 6118 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6119 ALOGW_IF(mEffectChains.size() != 1, 6120 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6121 chain.get(), mEffectChains.size(), this); 6122 if (mEffectChains.size() == 1) { 6123 mEffectChains.removeAt(0); 6124 } 6125 return 0; 6126} 6127 6128// ---------------------------------------------------------------------------- 6129// EffectModule implementation 6130// ---------------------------------------------------------------------------- 6131 6132#undef LOG_TAG 6133#define LOG_TAG "AudioFlinger::EffectModule" 6134 6135AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6136 const wp<AudioFlinger::EffectChain>& chain, 6137 effect_descriptor_t *desc, 6138 int id, 6139 int sessionId) 6140 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6141 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6142{ 6143 ALOGV("Constructor %p", this); 6144 int lStatus; 6145 sp<ThreadBase> thread = mThread.promote(); 6146 if (thread == 0) { 6147 return; 6148 } 6149 6150 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6151 6152 // create effect engine from effect factory 6153 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6154 6155 if (mStatus != NO_ERROR) { 6156 return; 6157 } 6158 lStatus = init(); 6159 if (lStatus < 0) { 6160 mStatus = lStatus; 6161 goto Error; 6162 } 6163 6164 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6165 mPinned = true; 6166 } 6167 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6168 return; 6169Error: 6170 EffectRelease(mEffectInterface); 6171 mEffectInterface = NULL; 6172 ALOGV("Constructor Error %d", mStatus); 6173} 6174 6175AudioFlinger::EffectModule::~EffectModule() 6176{ 6177 ALOGV("Destructor %p", this); 6178 if (mEffectInterface != NULL) { 6179 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6180 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6181 sp<ThreadBase> thread = mThread.promote(); 6182 if (thread != 0) { 6183 audio_stream_t *stream = thread->stream(); 6184 if (stream != NULL) { 6185 stream->remove_audio_effect(stream, mEffectInterface); 6186 } 6187 } 6188 } 6189 // release effect engine 6190 EffectRelease(mEffectInterface); 6191 } 6192} 6193 6194status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6195{ 6196 status_t status; 6197 6198 Mutex::Autolock _l(mLock); 6199 // First handle in mHandles has highest priority and controls the effect module 6200 int priority = handle->priority(); 6201 size_t size = mHandles.size(); 6202 sp<EffectHandle> h; 6203 size_t i; 6204 for (i = 0; i < size; i++) { 6205 h = mHandles[i].promote(); 6206 if (h == 0) continue; 6207 if (h->priority() <= priority) break; 6208 } 6209 // if inserted in first place, move effect control from previous owner to this handle 6210 if (i == 0) { 6211 bool enabled = false; 6212 if (h != 0) { 6213 enabled = h->enabled(); 6214 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6215 } 6216 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6217 status = NO_ERROR; 6218 } else { 6219 status = ALREADY_EXISTS; 6220 } 6221 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6222 mHandles.insertAt(handle, i); 6223 return status; 6224} 6225 6226size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6227{ 6228 Mutex::Autolock _l(mLock); 6229 size_t size = mHandles.size(); 6230 size_t i; 6231 for (i = 0; i < size; i++) { 6232 if (mHandles[i] == handle) break; 6233 } 6234 if (i == size) { 6235 return size; 6236 } 6237 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6238 6239 bool enabled = false; 6240 EffectHandle *hdl = handle.unsafe_get(); 6241 if (hdl != NULL) { 6242 ALOGV("removeHandle() unsafe_get OK"); 6243 enabled = hdl->enabled(); 6244 } 6245 mHandles.removeAt(i); 6246 size = mHandles.size(); 6247 // if removed from first place, move effect control from this handle to next in line 6248 if (i == 0 && size != 0) { 6249 sp<EffectHandle> h = mHandles[0].promote(); 6250 if (h != 0) { 6251 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6252 } 6253 } 6254 6255 // Prevent calls to process() and other functions on effect interface from now on. 6256 // The effect engine will be released by the destructor when the last strong reference on 6257 // this object is released which can happen after next process is called. 6258 if (size == 0 && !mPinned) { 6259 mState = DESTROYED; 6260 } 6261 6262 return size; 6263} 6264 6265sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6266{ 6267 Mutex::Autolock _l(mLock); 6268 sp<EffectHandle> handle; 6269 if (mHandles.size() != 0) { 6270 handle = mHandles[0].promote(); 6271 } 6272 return handle; 6273} 6274 6275void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6276{ 6277 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6278 // keep a strong reference on this EffectModule to avoid calling the 6279 // destructor before we exit 6280 sp<EffectModule> keep(this); 6281 { 6282 sp<ThreadBase> thread = mThread.promote(); 6283 if (thread != 0) { 6284 thread->disconnectEffect(keep, handle, unpiniflast); 6285 } 6286 } 6287} 6288 6289void AudioFlinger::EffectModule::updateState() { 6290 Mutex::Autolock _l(mLock); 6291 6292 switch (mState) { 6293 case RESTART: 6294 reset_l(); 6295 // FALL THROUGH 6296 6297 case STARTING: 6298 // clear auxiliary effect input buffer for next accumulation 6299 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6300 memset(mConfig.inputCfg.buffer.raw, 6301 0, 6302 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6303 } 6304 start_l(); 6305 mState = ACTIVE; 6306 break; 6307 case STOPPING: 6308 stop_l(); 6309 mDisableWaitCnt = mMaxDisableWaitCnt; 6310 mState = STOPPED; 6311 break; 6312 case STOPPED: 6313 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6314 // turn off sequence. 6315 if (--mDisableWaitCnt == 0) { 6316 reset_l(); 6317 mState = IDLE; 6318 } 6319 break; 6320 default: //IDLE , ACTIVE, DESTROYED 6321 break; 6322 } 6323} 6324 6325void AudioFlinger::EffectModule::process() 6326{ 6327 Mutex::Autolock _l(mLock); 6328 6329 if (mState == DESTROYED || mEffectInterface == NULL || 6330 mConfig.inputCfg.buffer.raw == NULL || 6331 mConfig.outputCfg.buffer.raw == NULL) { 6332 return; 6333 } 6334 6335 if (isProcessEnabled()) { 6336 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6337 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6338 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6339 mConfig.inputCfg.buffer.s32, 6340 mConfig.inputCfg.buffer.frameCount/2); 6341 } 6342 6343 // do the actual processing in the effect engine 6344 int ret = (*mEffectInterface)->process(mEffectInterface, 6345 &mConfig.inputCfg.buffer, 6346 &mConfig.outputCfg.buffer); 6347 6348 // force transition to IDLE state when engine is ready 6349 if (mState == STOPPED && ret == -ENODATA) { 6350 mDisableWaitCnt = 1; 6351 } 6352 6353 // clear auxiliary effect input buffer for next accumulation 6354 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6355 memset(mConfig.inputCfg.buffer.raw, 0, 6356 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6357 } 6358 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6359 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6360 // If an insert effect is idle and input buffer is different from output buffer, 6361 // accumulate input onto output 6362 sp<EffectChain> chain = mChain.promote(); 6363 if (chain != 0 && chain->activeTrackCnt() != 0) { 6364 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6365 int16_t *in = mConfig.inputCfg.buffer.s16; 6366 int16_t *out = mConfig.outputCfg.buffer.s16; 6367 for (size_t i = 0; i < frameCnt; i++) { 6368 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6369 } 6370 } 6371 } 6372} 6373 6374void AudioFlinger::EffectModule::reset_l() 6375{ 6376 if (mEffectInterface == NULL) { 6377 return; 6378 } 6379 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6380} 6381 6382status_t AudioFlinger::EffectModule::configure() 6383{ 6384 uint32_t channels; 6385 if (mEffectInterface == NULL) { 6386 return NO_INIT; 6387 } 6388 6389 sp<ThreadBase> thread = mThread.promote(); 6390 if (thread == 0) { 6391 return DEAD_OBJECT; 6392 } 6393 6394 // TODO: handle configuration of effects replacing track process 6395 if (thread->channelCount() == 1) { 6396 channels = AUDIO_CHANNEL_OUT_MONO; 6397 } else { 6398 channels = AUDIO_CHANNEL_OUT_STEREO; 6399 } 6400 6401 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6402 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6403 } else { 6404 mConfig.inputCfg.channels = channels; 6405 } 6406 mConfig.outputCfg.channels = channels; 6407 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6408 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6409 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6410 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6411 mConfig.inputCfg.bufferProvider.cookie = NULL; 6412 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6413 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6414 mConfig.outputCfg.bufferProvider.cookie = NULL; 6415 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6416 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6417 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6418 // Insert effect: 6419 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6420 // always overwrites output buffer: input buffer == output buffer 6421 // - in other sessions: 6422 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6423 // other effect: overwrites output buffer: input buffer == output buffer 6424 // Auxiliary effect: 6425 // accumulates in output buffer: input buffer != output buffer 6426 // Therefore: accumulate <=> input buffer != output buffer 6427 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6428 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6429 } else { 6430 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6431 } 6432 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6433 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6434 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6435 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6436 6437 ALOGV("configure() %p thread %p buffer %p framecount %d", 6438 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6439 6440 status_t cmdStatus; 6441 uint32_t size = sizeof(int); 6442 status_t status = (*mEffectInterface)->command(mEffectInterface, 6443 EFFECT_CMD_SET_CONFIG, 6444 sizeof(effect_config_t), 6445 &mConfig, 6446 &size, 6447 &cmdStatus); 6448 if (status == 0) { 6449 status = cmdStatus; 6450 } 6451 6452 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6453 (1000 * mConfig.outputCfg.buffer.frameCount); 6454 6455 return status; 6456} 6457 6458status_t AudioFlinger::EffectModule::init() 6459{ 6460 Mutex::Autolock _l(mLock); 6461 if (mEffectInterface == NULL) { 6462 return NO_INIT; 6463 } 6464 status_t cmdStatus; 6465 uint32_t size = sizeof(status_t); 6466 status_t status = (*mEffectInterface)->command(mEffectInterface, 6467 EFFECT_CMD_INIT, 6468 0, 6469 NULL, 6470 &size, 6471 &cmdStatus); 6472 if (status == 0) { 6473 status = cmdStatus; 6474 } 6475 return status; 6476} 6477 6478status_t AudioFlinger::EffectModule::start() 6479{ 6480 Mutex::Autolock _l(mLock); 6481 return start_l(); 6482} 6483 6484status_t AudioFlinger::EffectModule::start_l() 6485{ 6486 if (mEffectInterface == NULL) { 6487 return NO_INIT; 6488 } 6489 status_t cmdStatus; 6490 uint32_t size = sizeof(status_t); 6491 status_t status = (*mEffectInterface)->command(mEffectInterface, 6492 EFFECT_CMD_ENABLE, 6493 0, 6494 NULL, 6495 &size, 6496 &cmdStatus); 6497 if (status == 0) { 6498 status = cmdStatus; 6499 } 6500 if (status == 0 && 6501 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6502 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6503 sp<ThreadBase> thread = mThread.promote(); 6504 if (thread != 0) { 6505 audio_stream_t *stream = thread->stream(); 6506 if (stream != NULL) { 6507 stream->add_audio_effect(stream, mEffectInterface); 6508 } 6509 } 6510 } 6511 return status; 6512} 6513 6514status_t AudioFlinger::EffectModule::stop() 6515{ 6516 Mutex::Autolock _l(mLock); 6517 return stop_l(); 6518} 6519 6520status_t AudioFlinger::EffectModule::stop_l() 6521{ 6522 if (mEffectInterface == NULL) { 6523 return NO_INIT; 6524 } 6525 status_t cmdStatus; 6526 uint32_t size = sizeof(status_t); 6527 status_t status = (*mEffectInterface)->command(mEffectInterface, 6528 EFFECT_CMD_DISABLE, 6529 0, 6530 NULL, 6531 &size, 6532 &cmdStatus); 6533 if (status == 0) { 6534 status = cmdStatus; 6535 } 6536 if (status == 0 && 6537 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6538 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6539 sp<ThreadBase> thread = mThread.promote(); 6540 if (thread != 0) { 6541 audio_stream_t *stream = thread->stream(); 6542 if (stream != NULL) { 6543 stream->remove_audio_effect(stream, mEffectInterface); 6544 } 6545 } 6546 } 6547 return status; 6548} 6549 6550status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6551 uint32_t cmdSize, 6552 void *pCmdData, 6553 uint32_t *replySize, 6554 void *pReplyData) 6555{ 6556 Mutex::Autolock _l(mLock); 6557// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6558 6559 if (mState == DESTROYED || mEffectInterface == NULL) { 6560 return NO_INIT; 6561 } 6562 status_t status = (*mEffectInterface)->command(mEffectInterface, 6563 cmdCode, 6564 cmdSize, 6565 pCmdData, 6566 replySize, 6567 pReplyData); 6568 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6569 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6570 for (size_t i = 1; i < mHandles.size(); i++) { 6571 sp<EffectHandle> h = mHandles[i].promote(); 6572 if (h != 0) { 6573 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6574 } 6575 } 6576 } 6577 return status; 6578} 6579 6580status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6581{ 6582 6583 Mutex::Autolock _l(mLock); 6584 ALOGV("setEnabled %p enabled %d", this, enabled); 6585 6586 if (enabled != isEnabled()) { 6587 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6588 if (enabled && status != NO_ERROR) { 6589 return status; 6590 } 6591 6592 switch (mState) { 6593 // going from disabled to enabled 6594 case IDLE: 6595 mState = STARTING; 6596 break; 6597 case STOPPED: 6598 mState = RESTART; 6599 break; 6600 case STOPPING: 6601 mState = ACTIVE; 6602 break; 6603 6604 // going from enabled to disabled 6605 case RESTART: 6606 mState = STOPPED; 6607 break; 6608 case STARTING: 6609 mState = IDLE; 6610 break; 6611 case ACTIVE: 6612 mState = STOPPING; 6613 break; 6614 case DESTROYED: 6615 return NO_ERROR; // simply ignore as we are being destroyed 6616 } 6617 for (size_t i = 1; i < mHandles.size(); i++) { 6618 sp<EffectHandle> h = mHandles[i].promote(); 6619 if (h != 0) { 6620 h->setEnabled(enabled); 6621 } 6622 } 6623 } 6624 return NO_ERROR; 6625} 6626 6627bool AudioFlinger::EffectModule::isEnabled() 6628{ 6629 switch (mState) { 6630 case RESTART: 6631 case STARTING: 6632 case ACTIVE: 6633 return true; 6634 case IDLE: 6635 case STOPPING: 6636 case STOPPED: 6637 case DESTROYED: 6638 default: 6639 return false; 6640 } 6641} 6642 6643bool AudioFlinger::EffectModule::isProcessEnabled() 6644{ 6645 switch (mState) { 6646 case RESTART: 6647 case ACTIVE: 6648 case STOPPING: 6649 case STOPPED: 6650 return true; 6651 case IDLE: 6652 case STARTING: 6653 case DESTROYED: 6654 default: 6655 return false; 6656 } 6657} 6658 6659status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6660{ 6661 Mutex::Autolock _l(mLock); 6662 status_t status = NO_ERROR; 6663 6664 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6665 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6666 if (isProcessEnabled() && 6667 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6668 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6669 status_t cmdStatus; 6670 uint32_t volume[2]; 6671 uint32_t *pVolume = NULL; 6672 uint32_t size = sizeof(volume); 6673 volume[0] = *left; 6674 volume[1] = *right; 6675 if (controller) { 6676 pVolume = volume; 6677 } 6678 status = (*mEffectInterface)->command(mEffectInterface, 6679 EFFECT_CMD_SET_VOLUME, 6680 size, 6681 volume, 6682 &size, 6683 pVolume); 6684 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6685 *left = volume[0]; 6686 *right = volume[1]; 6687 } 6688 } 6689 return status; 6690} 6691 6692status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6693{ 6694 Mutex::Autolock _l(mLock); 6695 status_t status = NO_ERROR; 6696 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6697 // audio pre processing modules on RecordThread can receive both output and 6698 // input device indication in the same call 6699 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6700 if (dev) { 6701 status_t cmdStatus; 6702 uint32_t size = sizeof(status_t); 6703 6704 status = (*mEffectInterface)->command(mEffectInterface, 6705 EFFECT_CMD_SET_DEVICE, 6706 sizeof(uint32_t), 6707 &dev, 6708 &size, 6709 &cmdStatus); 6710 if (status == NO_ERROR) { 6711 status = cmdStatus; 6712 } 6713 } 6714 dev = device & AUDIO_DEVICE_IN_ALL; 6715 if (dev) { 6716 status_t cmdStatus; 6717 uint32_t size = sizeof(status_t); 6718 6719 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6720 EFFECT_CMD_SET_INPUT_DEVICE, 6721 sizeof(uint32_t), 6722 &dev, 6723 &size, 6724 &cmdStatus); 6725 if (status2 == NO_ERROR) { 6726 status2 = cmdStatus; 6727 } 6728 if (status == NO_ERROR) { 6729 status = status2; 6730 } 6731 } 6732 } 6733 return status; 6734} 6735 6736status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6737{ 6738 Mutex::Autolock _l(mLock); 6739 status_t status = NO_ERROR; 6740 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6741 status_t cmdStatus; 6742 uint32_t size = sizeof(status_t); 6743 status = (*mEffectInterface)->command(mEffectInterface, 6744 EFFECT_CMD_SET_AUDIO_MODE, 6745 sizeof(audio_mode_t), 6746 &mode, 6747 &size, 6748 &cmdStatus); 6749 if (status == NO_ERROR) { 6750 status = cmdStatus; 6751 } 6752 } 6753 return status; 6754} 6755 6756void AudioFlinger::EffectModule::setSuspended(bool suspended) 6757{ 6758 Mutex::Autolock _l(mLock); 6759 mSuspended = suspended; 6760} 6761 6762bool AudioFlinger::EffectModule::suspended() const 6763{ 6764 Mutex::Autolock _l(mLock); 6765 return mSuspended; 6766} 6767 6768status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6769{ 6770 const size_t SIZE = 256; 6771 char buffer[SIZE]; 6772 String8 result; 6773 6774 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6775 result.append(buffer); 6776 6777 bool locked = tryLock(mLock); 6778 // failed to lock - AudioFlinger is probably deadlocked 6779 if (!locked) { 6780 result.append("\t\tCould not lock Fx mutex:\n"); 6781 } 6782 6783 result.append("\t\tSession Status State Engine:\n"); 6784 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6785 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6786 result.append(buffer); 6787 6788 result.append("\t\tDescriptor:\n"); 6789 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6790 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6791 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6792 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6793 result.append(buffer); 6794 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6795 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6796 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6797 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6798 result.append(buffer); 6799 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6800 mDescriptor.apiVersion, 6801 mDescriptor.flags); 6802 result.append(buffer); 6803 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6804 mDescriptor.name); 6805 result.append(buffer); 6806 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6807 mDescriptor.implementor); 6808 result.append(buffer); 6809 6810 result.append("\t\t- Input configuration:\n"); 6811 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6812 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6813 (uint32_t)mConfig.inputCfg.buffer.raw, 6814 mConfig.inputCfg.buffer.frameCount, 6815 mConfig.inputCfg.samplingRate, 6816 mConfig.inputCfg.channels, 6817 mConfig.inputCfg.format); 6818 result.append(buffer); 6819 6820 result.append("\t\t- Output configuration:\n"); 6821 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6822 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6823 (uint32_t)mConfig.outputCfg.buffer.raw, 6824 mConfig.outputCfg.buffer.frameCount, 6825 mConfig.outputCfg.samplingRate, 6826 mConfig.outputCfg.channels, 6827 mConfig.outputCfg.format); 6828 result.append(buffer); 6829 6830 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6831 result.append(buffer); 6832 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6833 for (size_t i = 0; i < mHandles.size(); ++i) { 6834 sp<EffectHandle> handle = mHandles[i].promote(); 6835 if (handle != 0) { 6836 handle->dump(buffer, SIZE); 6837 result.append(buffer); 6838 } 6839 } 6840 6841 result.append("\n"); 6842 6843 write(fd, result.string(), result.length()); 6844 6845 if (locked) { 6846 mLock.unlock(); 6847 } 6848 6849 return NO_ERROR; 6850} 6851 6852// ---------------------------------------------------------------------------- 6853// EffectHandle implementation 6854// ---------------------------------------------------------------------------- 6855 6856#undef LOG_TAG 6857#define LOG_TAG "AudioFlinger::EffectHandle" 6858 6859AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6860 const sp<AudioFlinger::Client>& client, 6861 const sp<IEffectClient>& effectClient, 6862 int32_t priority) 6863 : BnEffect(), 6864 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6865 mPriority(priority), mHasControl(false), mEnabled(false) 6866{ 6867 ALOGV("constructor %p", this); 6868 6869 if (client == 0) { 6870 return; 6871 } 6872 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6873 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6874 if (mCblkMemory != 0) { 6875 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6876 6877 if (mCblk != NULL) { 6878 new(mCblk) effect_param_cblk_t(); 6879 mBuffer = (uint8_t *)mCblk + bufOffset; 6880 } 6881 } else { 6882 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6883 return; 6884 } 6885} 6886 6887AudioFlinger::EffectHandle::~EffectHandle() 6888{ 6889 ALOGV("Destructor %p", this); 6890 disconnect(false); 6891 ALOGV("Destructor DONE %p", this); 6892} 6893 6894status_t AudioFlinger::EffectHandle::enable() 6895{ 6896 ALOGV("enable %p", this); 6897 if (!mHasControl) return INVALID_OPERATION; 6898 if (mEffect == 0) return DEAD_OBJECT; 6899 6900 if (mEnabled) { 6901 return NO_ERROR; 6902 } 6903 6904 mEnabled = true; 6905 6906 sp<ThreadBase> thread = mEffect->thread().promote(); 6907 if (thread != 0) { 6908 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6909 } 6910 6911 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6912 if (mEffect->suspended()) { 6913 return NO_ERROR; 6914 } 6915 6916 status_t status = mEffect->setEnabled(true); 6917 if (status != NO_ERROR) { 6918 if (thread != 0) { 6919 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6920 } 6921 mEnabled = false; 6922 } 6923 return status; 6924} 6925 6926status_t AudioFlinger::EffectHandle::disable() 6927{ 6928 ALOGV("disable %p", this); 6929 if (!mHasControl) return INVALID_OPERATION; 6930 if (mEffect == 0) return DEAD_OBJECT; 6931 6932 if (!mEnabled) { 6933 return NO_ERROR; 6934 } 6935 mEnabled = false; 6936 6937 if (mEffect->suspended()) { 6938 return NO_ERROR; 6939 } 6940 6941 status_t status = mEffect->setEnabled(false); 6942 6943 sp<ThreadBase> thread = mEffect->thread().promote(); 6944 if (thread != 0) { 6945 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6946 } 6947 6948 return status; 6949} 6950 6951void AudioFlinger::EffectHandle::disconnect() 6952{ 6953 disconnect(true); 6954} 6955 6956void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6957{ 6958 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6959 if (mEffect == 0) { 6960 return; 6961 } 6962 mEffect->disconnect(this, unpiniflast); 6963 6964 if (mHasControl && mEnabled) { 6965 sp<ThreadBase> thread = mEffect->thread().promote(); 6966 if (thread != 0) { 6967 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6968 } 6969 } 6970 6971 // release sp on module => module destructor can be called now 6972 mEffect.clear(); 6973 if (mClient != 0) { 6974 if (mCblk != NULL) { 6975 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6976 } 6977 mCblkMemory.clear(); // and free the shared memory 6978 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6979 mClient.clear(); 6980 } 6981} 6982 6983status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6984 uint32_t cmdSize, 6985 void *pCmdData, 6986 uint32_t *replySize, 6987 void *pReplyData) 6988{ 6989// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6990// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6991 6992 // only get parameter command is permitted for applications not controlling the effect 6993 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6994 return INVALID_OPERATION; 6995 } 6996 if (mEffect == 0) return DEAD_OBJECT; 6997 if (mClient == 0) return INVALID_OPERATION; 6998 6999 // handle commands that are not forwarded transparently to effect engine 7000 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7001 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7002 // no risk to block the whole media server process or mixer threads is we are stuck here 7003 Mutex::Autolock _l(mCblk->lock); 7004 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7005 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7006 mCblk->serverIndex = 0; 7007 mCblk->clientIndex = 0; 7008 return BAD_VALUE; 7009 } 7010 status_t status = NO_ERROR; 7011 while (mCblk->serverIndex < mCblk->clientIndex) { 7012 int reply; 7013 uint32_t rsize = sizeof(int); 7014 int *p = (int *)(mBuffer + mCblk->serverIndex); 7015 int size = *p++; 7016 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7017 ALOGW("command(): invalid parameter block size"); 7018 break; 7019 } 7020 effect_param_t *param = (effect_param_t *)p; 7021 if (param->psize == 0 || param->vsize == 0) { 7022 ALOGW("command(): null parameter or value size"); 7023 mCblk->serverIndex += size; 7024 continue; 7025 } 7026 uint32_t psize = sizeof(effect_param_t) + 7027 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7028 param->vsize; 7029 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7030 psize, 7031 p, 7032 &rsize, 7033 &reply); 7034 // stop at first error encountered 7035 if (ret != NO_ERROR) { 7036 status = ret; 7037 *(int *)pReplyData = reply; 7038 break; 7039 } else if (reply != NO_ERROR) { 7040 *(int *)pReplyData = reply; 7041 break; 7042 } 7043 mCblk->serverIndex += size; 7044 } 7045 mCblk->serverIndex = 0; 7046 mCblk->clientIndex = 0; 7047 return status; 7048 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7049 *(int *)pReplyData = NO_ERROR; 7050 return enable(); 7051 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7052 *(int *)pReplyData = NO_ERROR; 7053 return disable(); 7054 } 7055 7056 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7057} 7058 7059sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7060 return mCblkMemory; 7061} 7062 7063void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7064{ 7065 ALOGV("setControl %p control %d", this, hasControl); 7066 7067 mHasControl = hasControl; 7068 mEnabled = enabled; 7069 7070 if (signal && mEffectClient != 0) { 7071 mEffectClient->controlStatusChanged(hasControl); 7072 } 7073} 7074 7075void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7076 uint32_t cmdSize, 7077 void *pCmdData, 7078 uint32_t replySize, 7079 void *pReplyData) 7080{ 7081 if (mEffectClient != 0) { 7082 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7083 } 7084} 7085 7086 7087 7088void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7089{ 7090 if (mEffectClient != 0) { 7091 mEffectClient->enableStatusChanged(enabled); 7092 } 7093} 7094 7095status_t AudioFlinger::EffectHandle::onTransact( 7096 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7097{ 7098 return BnEffect::onTransact(code, data, reply, flags); 7099} 7100 7101 7102void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7103{ 7104 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7105 7106 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7107 (mClient == 0) ? getpid() : mClient->pid(), 7108 mPriority, 7109 mHasControl, 7110 !locked, 7111 mCblk ? mCblk->clientIndex : 0, 7112 mCblk ? mCblk->serverIndex : 0 7113 ); 7114 7115 if (locked) { 7116 mCblk->lock.unlock(); 7117 } 7118} 7119 7120#undef LOG_TAG 7121#define LOG_TAG "AudioFlinger::EffectChain" 7122 7123AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7124 int sessionId) 7125 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7126 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7127 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7128{ 7129 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7130 sp<ThreadBase> thread = mThread.promote(); 7131 if (thread == 0) { 7132 return; 7133 } 7134 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7135 thread->frameCount(); 7136} 7137 7138AudioFlinger::EffectChain::~EffectChain() 7139{ 7140 if (mOwnInBuffer) { 7141 delete mInBuffer; 7142 } 7143 7144} 7145 7146// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7147sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7148{ 7149 sp<EffectModule> effect; 7150 size_t size = mEffects.size(); 7151 7152 for (size_t i = 0; i < size; i++) { 7153 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7154 effect = mEffects[i]; 7155 break; 7156 } 7157 } 7158 return effect; 7159} 7160 7161// getEffectFromId_l() must be called with ThreadBase::mLock held 7162sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7163{ 7164 sp<EffectModule> effect; 7165 size_t size = mEffects.size(); 7166 7167 for (size_t i = 0; i < size; i++) { 7168 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7169 if (id == 0 || mEffects[i]->id() == id) { 7170 effect = mEffects[i]; 7171 break; 7172 } 7173 } 7174 return effect; 7175} 7176 7177// getEffectFromType_l() must be called with ThreadBase::mLock held 7178sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7179 const effect_uuid_t *type) 7180{ 7181 sp<EffectModule> effect; 7182 size_t size = mEffects.size(); 7183 7184 for (size_t i = 0; i < size; i++) { 7185 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7186 effect = mEffects[i]; 7187 break; 7188 } 7189 } 7190 return effect; 7191} 7192 7193// Must be called with EffectChain::mLock locked 7194void AudioFlinger::EffectChain::process_l() 7195{ 7196 sp<ThreadBase> thread = mThread.promote(); 7197 if (thread == 0) { 7198 ALOGW("process_l(): cannot promote mixer thread"); 7199 return; 7200 } 7201 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7202 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7203 // always process effects unless no more tracks are on the session and the effect tail 7204 // has been rendered 7205 bool doProcess = true; 7206 if (!isGlobalSession) { 7207 bool tracksOnSession = (trackCnt() != 0); 7208 7209 if (!tracksOnSession && mTailBufferCount == 0) { 7210 doProcess = false; 7211 } 7212 7213 if (activeTrackCnt() == 0) { 7214 // if no track is active and the effect tail has not been rendered, 7215 // the input buffer must be cleared here as the mixer process will not do it 7216 if (tracksOnSession || mTailBufferCount > 0) { 7217 size_t numSamples = thread->frameCount() * thread->channelCount(); 7218 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7219 if (mTailBufferCount > 0) { 7220 mTailBufferCount--; 7221 } 7222 } 7223 } 7224 } 7225 7226 size_t size = mEffects.size(); 7227 if (doProcess) { 7228 for (size_t i = 0; i < size; i++) { 7229 mEffects[i]->process(); 7230 } 7231 } 7232 for (size_t i = 0; i < size; i++) { 7233 mEffects[i]->updateState(); 7234 } 7235} 7236 7237// addEffect_l() must be called with PlaybackThread::mLock held 7238status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7239{ 7240 effect_descriptor_t desc = effect->desc(); 7241 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7242 7243 Mutex::Autolock _l(mLock); 7244 effect->setChain(this); 7245 sp<ThreadBase> thread = mThread.promote(); 7246 if (thread == 0) { 7247 return NO_INIT; 7248 } 7249 effect->setThread(thread); 7250 7251 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7252 // Auxiliary effects are inserted at the beginning of mEffects vector as 7253 // they are processed first and accumulated in chain input buffer 7254 mEffects.insertAt(effect, 0); 7255 7256 // the input buffer for auxiliary effect contains mono samples in 7257 // 32 bit format. This is to avoid saturation in AudoMixer 7258 // accumulation stage. Saturation is done in EffectModule::process() before 7259 // calling the process in effect engine 7260 size_t numSamples = thread->frameCount(); 7261 int32_t *buffer = new int32_t[numSamples]; 7262 memset(buffer, 0, numSamples * sizeof(int32_t)); 7263 effect->setInBuffer((int16_t *)buffer); 7264 // auxiliary effects output samples to chain input buffer for further processing 7265 // by insert effects 7266 effect->setOutBuffer(mInBuffer); 7267 } else { 7268 // Insert effects are inserted at the end of mEffects vector as they are processed 7269 // after track and auxiliary effects. 7270 // Insert effect order as a function of indicated preference: 7271 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7272 // another effect is present 7273 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7274 // last effect claiming first position 7275 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7276 // first effect claiming last position 7277 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7278 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7279 // already present 7280 7281 int size = (int)mEffects.size(); 7282 int idx_insert = size; 7283 int idx_insert_first = -1; 7284 int idx_insert_last = -1; 7285 7286 for (int i = 0; i < size; i++) { 7287 effect_descriptor_t d = mEffects[i]->desc(); 7288 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7289 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7290 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7291 // check invalid effect chaining combinations 7292 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7293 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7294 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7295 return INVALID_OPERATION; 7296 } 7297 // remember position of first insert effect and by default 7298 // select this as insert position for new effect 7299 if (idx_insert == size) { 7300 idx_insert = i; 7301 } 7302 // remember position of last insert effect claiming 7303 // first position 7304 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7305 idx_insert_first = i; 7306 } 7307 // remember position of first insert effect claiming 7308 // last position 7309 if (iPref == EFFECT_FLAG_INSERT_LAST && 7310 idx_insert_last == -1) { 7311 idx_insert_last = i; 7312 } 7313 } 7314 } 7315 7316 // modify idx_insert from first position if needed 7317 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7318 if (idx_insert_last != -1) { 7319 idx_insert = idx_insert_last; 7320 } else { 7321 idx_insert = size; 7322 } 7323 } else { 7324 if (idx_insert_first != -1) { 7325 idx_insert = idx_insert_first + 1; 7326 } 7327 } 7328 7329 // always read samples from chain input buffer 7330 effect->setInBuffer(mInBuffer); 7331 7332 // if last effect in the chain, output samples to chain 7333 // output buffer, otherwise to chain input buffer 7334 if (idx_insert == size) { 7335 if (idx_insert != 0) { 7336 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7337 mEffects[idx_insert-1]->configure(); 7338 } 7339 effect->setOutBuffer(mOutBuffer); 7340 } else { 7341 effect->setOutBuffer(mInBuffer); 7342 } 7343 mEffects.insertAt(effect, idx_insert); 7344 7345 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7346 } 7347 effect->configure(); 7348 return NO_ERROR; 7349} 7350 7351// removeEffect_l() must be called with PlaybackThread::mLock held 7352size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7353{ 7354 Mutex::Autolock _l(mLock); 7355 int size = (int)mEffects.size(); 7356 int i; 7357 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7358 7359 for (i = 0; i < size; i++) { 7360 if (effect == mEffects[i]) { 7361 // calling stop here will remove pre-processing effect from the audio HAL. 7362 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7363 // the middle of a read from audio HAL 7364 if (mEffects[i]->state() == EffectModule::ACTIVE || 7365 mEffects[i]->state() == EffectModule::STOPPING) { 7366 mEffects[i]->stop(); 7367 } 7368 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7369 delete[] effect->inBuffer(); 7370 } else { 7371 if (i == size - 1 && i != 0) { 7372 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7373 mEffects[i - 1]->configure(); 7374 } 7375 } 7376 mEffects.removeAt(i); 7377 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7378 break; 7379 } 7380 } 7381 7382 return mEffects.size(); 7383} 7384 7385// setDevice_l() must be called with PlaybackThread::mLock held 7386void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7387{ 7388 size_t size = mEffects.size(); 7389 for (size_t i = 0; i < size; i++) { 7390 mEffects[i]->setDevice(device); 7391 } 7392} 7393 7394// setMode_l() must be called with PlaybackThread::mLock held 7395void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7396{ 7397 size_t size = mEffects.size(); 7398 for (size_t i = 0; i < size; i++) { 7399 mEffects[i]->setMode(mode); 7400 } 7401} 7402 7403// setVolume_l() must be called with PlaybackThread::mLock held 7404bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7405{ 7406 uint32_t newLeft = *left; 7407 uint32_t newRight = *right; 7408 bool hasControl = false; 7409 int ctrlIdx = -1; 7410 size_t size = mEffects.size(); 7411 7412 // first update volume controller 7413 for (size_t i = size; i > 0; i--) { 7414 if (mEffects[i - 1]->isProcessEnabled() && 7415 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7416 ctrlIdx = i - 1; 7417 hasControl = true; 7418 break; 7419 } 7420 } 7421 7422 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7423 if (hasControl) { 7424 *left = mNewLeftVolume; 7425 *right = mNewRightVolume; 7426 } 7427 return hasControl; 7428 } 7429 7430 mVolumeCtrlIdx = ctrlIdx; 7431 mLeftVolume = newLeft; 7432 mRightVolume = newRight; 7433 7434 // second get volume update from volume controller 7435 if (ctrlIdx >= 0) { 7436 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7437 mNewLeftVolume = newLeft; 7438 mNewRightVolume = newRight; 7439 } 7440 // then indicate volume to all other effects in chain. 7441 // Pass altered volume to effects before volume controller 7442 // and requested volume to effects after controller 7443 uint32_t lVol = newLeft; 7444 uint32_t rVol = newRight; 7445 7446 for (size_t i = 0; i < size; i++) { 7447 if ((int)i == ctrlIdx) continue; 7448 // this also works for ctrlIdx == -1 when there is no volume controller 7449 if ((int)i > ctrlIdx) { 7450 lVol = *left; 7451 rVol = *right; 7452 } 7453 mEffects[i]->setVolume(&lVol, &rVol, false); 7454 } 7455 *left = newLeft; 7456 *right = newRight; 7457 7458 return hasControl; 7459} 7460 7461status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7462{ 7463 const size_t SIZE = 256; 7464 char buffer[SIZE]; 7465 String8 result; 7466 7467 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7468 result.append(buffer); 7469 7470 bool locked = tryLock(mLock); 7471 // failed to lock - AudioFlinger is probably deadlocked 7472 if (!locked) { 7473 result.append("\tCould not lock mutex:\n"); 7474 } 7475 7476 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7477 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7478 mEffects.size(), 7479 (uint32_t)mInBuffer, 7480 (uint32_t)mOutBuffer, 7481 mActiveTrackCnt); 7482 result.append(buffer); 7483 write(fd, result.string(), result.size()); 7484 7485 for (size_t i = 0; i < mEffects.size(); ++i) { 7486 sp<EffectModule> effect = mEffects[i]; 7487 if (effect != 0) { 7488 effect->dump(fd, args); 7489 } 7490 } 7491 7492 if (locked) { 7493 mLock.unlock(); 7494 } 7495 7496 return NO_ERROR; 7497} 7498 7499// must be called with ThreadBase::mLock held 7500void AudioFlinger::EffectChain::setEffectSuspended_l( 7501 const effect_uuid_t *type, bool suspend) 7502{ 7503 sp<SuspendedEffectDesc> desc; 7504 // use effect type UUID timelow as key as there is no real risk of identical 7505 // timeLow fields among effect type UUIDs. 7506 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7507 if (suspend) { 7508 if (index >= 0) { 7509 desc = mSuspendedEffects.valueAt(index); 7510 } else { 7511 desc = new SuspendedEffectDesc(); 7512 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7513 mSuspendedEffects.add(type->timeLow, desc); 7514 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7515 } 7516 if (desc->mRefCount++ == 0) { 7517 sp<EffectModule> effect = getEffectIfEnabled(type); 7518 if (effect != 0) { 7519 desc->mEffect = effect; 7520 effect->setSuspended(true); 7521 effect->setEnabled(false); 7522 } 7523 } 7524 } else { 7525 if (index < 0) { 7526 return; 7527 } 7528 desc = mSuspendedEffects.valueAt(index); 7529 if (desc->mRefCount <= 0) { 7530 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7531 desc->mRefCount = 1; 7532 } 7533 if (--desc->mRefCount == 0) { 7534 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7535 if (desc->mEffect != 0) { 7536 sp<EffectModule> effect = desc->mEffect.promote(); 7537 if (effect != 0) { 7538 effect->setSuspended(false); 7539 sp<EffectHandle> handle = effect->controlHandle(); 7540 if (handle != 0) { 7541 effect->setEnabled(handle->enabled()); 7542 } 7543 } 7544 desc->mEffect.clear(); 7545 } 7546 mSuspendedEffects.removeItemsAt(index); 7547 } 7548 } 7549} 7550 7551// must be called with ThreadBase::mLock held 7552void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7553{ 7554 sp<SuspendedEffectDesc> desc; 7555 7556 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7557 if (suspend) { 7558 if (index >= 0) { 7559 desc = mSuspendedEffects.valueAt(index); 7560 } else { 7561 desc = new SuspendedEffectDesc(); 7562 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7563 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7564 } 7565 if (desc->mRefCount++ == 0) { 7566 Vector< sp<EffectModule> > effects; 7567 getSuspendEligibleEffects(effects); 7568 for (size_t i = 0; i < effects.size(); i++) { 7569 setEffectSuspended_l(&effects[i]->desc().type, true); 7570 } 7571 } 7572 } else { 7573 if (index < 0) { 7574 return; 7575 } 7576 desc = mSuspendedEffects.valueAt(index); 7577 if (desc->mRefCount <= 0) { 7578 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7579 desc->mRefCount = 1; 7580 } 7581 if (--desc->mRefCount == 0) { 7582 Vector<const effect_uuid_t *> types; 7583 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7584 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7585 continue; 7586 } 7587 types.add(&mSuspendedEffects.valueAt(i)->mType); 7588 } 7589 for (size_t i = 0; i < types.size(); i++) { 7590 setEffectSuspended_l(types[i], false); 7591 } 7592 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7593 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7594 } 7595 } 7596} 7597 7598 7599// The volume effect is used for automated tests only 7600#ifndef OPENSL_ES_H_ 7601static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7602 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7603const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7604#endif //OPENSL_ES_H_ 7605 7606bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7607{ 7608 // auxiliary effects and visualizer are never suspended on output mix 7609 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7610 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7611 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7612 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7613 return false; 7614 } 7615 return true; 7616} 7617 7618void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 7619{ 7620 effects.clear(); 7621 for (size_t i = 0; i < mEffects.size(); i++) { 7622 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 7623 effects.add(mEffects[i]); 7624 } 7625 } 7626} 7627 7628sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7629 const effect_uuid_t *type) 7630{ 7631 sp<EffectModule> effect; 7632 effect = getEffectFromType_l(type); 7633 if (effect != 0 && !effect->isEnabled()) { 7634 effect.clear(); 7635 } 7636 return effect; 7637} 7638 7639void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7640 bool enabled) 7641{ 7642 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7643 if (enabled) { 7644 if (index < 0) { 7645 // if the effect is not suspend check if all effects are suspended 7646 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7647 if (index < 0) { 7648 return; 7649 } 7650 if (!isEffectEligibleForSuspend(effect->desc())) { 7651 return; 7652 } 7653 setEffectSuspended_l(&effect->desc().type, enabled); 7654 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7655 if (index < 0) { 7656 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7657 return; 7658 } 7659 } 7660 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7661 effect->desc().type.timeLow); 7662 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7663 // if effect is requested to suspended but was not yet enabled, supend it now. 7664 if (desc->mEffect == 0) { 7665 desc->mEffect = effect; 7666 effect->setEnabled(false); 7667 effect->setSuspended(true); 7668 } 7669 } else { 7670 if (index < 0) { 7671 return; 7672 } 7673 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7674 effect->desc().type.timeLow); 7675 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7676 desc->mEffect.clear(); 7677 effect->setSuspended(false); 7678 } 7679} 7680 7681#undef LOG_TAG 7682#define LOG_TAG "AudioFlinger" 7683 7684// ---------------------------------------------------------------------------- 7685 7686status_t AudioFlinger::onTransact( 7687 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7688{ 7689 return BnAudioFlinger::onTransact(code, data, reply, flags); 7690} 7691 7692}; // namespace android 7693