AudioFlinger.cpp revision 1a0ae5be3d1273cba12584b33830d859510fbf82
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 != NULL) { 813 result = thread->setParameters(keyValuePairs); 814 return result; 815 } 816 return BAD_VALUE; 817} 818 819String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 820{ 821// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 822// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 823 824 if (ioHandle == 0) { 825 String8 out_s8; 826 827 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 828 audio_hw_device_t *dev = mAudioHwDevs[i]; 829 char *s = dev->get_parameters(dev, keys.string()); 830 out_s8 += String8(s); 831 free(s); 832 } 833 return out_s8; 834 } 835 836 Mutex::Autolock _l(mLock); 837 838 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 839 if (playbackThread != NULL) { 840 return playbackThread->getParameters(keys); 841 } 842 RecordThread *recordThread = checkRecordThread_l(ioHandle); 843 if (recordThread != NULL) { 844 return recordThread->getParameters(keys); 845 } 846 return String8(""); 847} 848 849size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) 850{ 851 status_t ret = initCheck(); 852 if (ret != NO_ERROR) { 853 return 0; 854 } 855 856 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 857} 858 859unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 860{ 861 if (ioHandle == 0) { 862 return 0; 863 } 864 865 Mutex::Autolock _l(mLock); 866 867 RecordThread *recordThread = checkRecordThread_l(ioHandle); 868 if (recordThread != NULL) { 869 return recordThread->getInputFramesLost(); 870 } 871 return 0; 872} 873 874status_t AudioFlinger::setVoiceVolume(float value) 875{ 876 status_t ret = initCheck(); 877 if (ret != NO_ERROR) { 878 return ret; 879 } 880 881 // check calling permissions 882 if (!settingsAllowed()) { 883 return PERMISSION_DENIED; 884 } 885 886 AutoMutex lock(mHardwareLock); 887 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 888 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 889 mHardwareStatus = AUDIO_HW_IDLE; 890 891 return ret; 892} 893 894status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 895{ 896 status_t status; 897 898 Mutex::Autolock _l(mLock); 899 900 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 901 if (playbackThread != NULL) { 902 return playbackThread->getRenderPosition(halFrames, dspFrames); 903 } 904 905 return BAD_VALUE; 906} 907 908void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 909{ 910 911 Mutex::Autolock _l(mLock); 912 913 int pid = IPCThreadState::self()->getCallingPid(); 914 if (mNotificationClients.indexOfKey(pid) < 0) { 915 sp<NotificationClient> notificationClient = new NotificationClient(this, 916 client, 917 pid); 918 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 919 920 mNotificationClients.add(pid, notificationClient); 921 922 sp<IBinder> binder = client->asBinder(); 923 binder->linkToDeath(notificationClient); 924 925 // the config change is always sent from playback or record threads to avoid deadlock 926 // with AudioSystem::gLock 927 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 928 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 929 } 930 931 for (size_t i = 0; i < mRecordThreads.size(); i++) { 932 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 933 } 934 } 935} 936 937void AudioFlinger::removeNotificationClient(pid_t pid) 938{ 939 Mutex::Autolock _l(mLock); 940 941 int index = mNotificationClients.indexOfKey(pid); 942 if (index >= 0) { 943 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 944 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 945 mNotificationClients.removeItem(pid); 946 } 947 948 ALOGV("%d died, releasing its sessions", pid); 949 int num = mAudioSessionRefs.size(); 950 bool removed = false; 951 for (int i = 0; i< num; i++) { 952 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 953 ALOGV(" pid %d @ %d", ref->pid, i); 954 if (ref->pid == pid) { 955 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 956 mAudioSessionRefs.removeAt(i); 957 delete ref; 958 removed = true; 959 i--; 960 num--; 961 } 962 } 963 if (removed) { 964 purgeStaleEffects_l(); 965 } 966} 967 968// audioConfigChanged_l() must be called with AudioFlinger::mLock held 969void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 970{ 971 size_t size = mNotificationClients.size(); 972 for (size_t i = 0; i < size; i++) { 973 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 974 param2); 975 } 976} 977 978// removeClient_l() must be called with AudioFlinger::mLock held 979void AudioFlinger::removeClient_l(pid_t pid) 980{ 981 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 982 mClients.removeItem(pid); 983} 984 985 986// ---------------------------------------------------------------------------- 987 988AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device, 989 type_t type) 990 : Thread(false), 991 mType(type), 992 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 993 // mChannelMask 994 mChannelCount(0), 995 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 996 mParamStatus(NO_ERROR), 997 mStandby(false), mId(id), mExiting(false), 998 mDevice(device), 999 mDeathRecipient(new PMDeathRecipient(this)) 1000{ 1001} 1002 1003AudioFlinger::ThreadBase::~ThreadBase() 1004{ 1005 mParamCond.broadcast(); 1006 // do not lock the mutex in destructor 1007 releaseWakeLock_l(); 1008 if (mPowerManager != 0) { 1009 sp<IBinder> binder = mPowerManager->asBinder(); 1010 binder->unlinkToDeath(mDeathRecipient); 1011 } 1012} 1013 1014void AudioFlinger::ThreadBase::exit() 1015{ 1016 // keep a strong ref on ourself so that we won't get 1017 // destroyed in the middle of requestExitAndWait() 1018 sp <ThreadBase> strongMe = this; 1019 1020 ALOGV("ThreadBase::exit"); 1021 { 1022 AutoMutex lock(mLock); 1023 mExiting = true; 1024 requestExit(); 1025 mWaitWorkCV.signal(); 1026 } 1027 requestExitAndWait(); 1028} 1029 1030uint32_t AudioFlinger::ThreadBase::sampleRate() const 1031{ 1032 return mSampleRate; 1033} 1034 1035int AudioFlinger::ThreadBase::channelCount() const 1036{ 1037 return (int)mChannelCount; 1038} 1039 1040audio_format_t AudioFlinger::ThreadBase::format() const 1041{ 1042 return mFormat; 1043} 1044 1045size_t AudioFlinger::ThreadBase::frameCount() const 1046{ 1047 return mFrameCount; 1048} 1049 1050status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1051{ 1052 status_t status; 1053 1054 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1055 Mutex::Autolock _l(mLock); 1056 1057 mNewParameters.add(keyValuePairs); 1058 mWaitWorkCV.signal(); 1059 // wait condition with timeout in case the thread loop has exited 1060 // before the request could be processed 1061 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1062 status = mParamStatus; 1063 mWaitWorkCV.signal(); 1064 } else { 1065 status = TIMED_OUT; 1066 } 1067 return status; 1068} 1069 1070void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1071{ 1072 Mutex::Autolock _l(mLock); 1073 sendConfigEvent_l(event, param); 1074} 1075 1076// sendConfigEvent_l() must be called with ThreadBase::mLock held 1077void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1078{ 1079 ConfigEvent configEvent; 1080 configEvent.mEvent = event; 1081 configEvent.mParam = param; 1082 mConfigEvents.add(configEvent); 1083 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1084 mWaitWorkCV.signal(); 1085} 1086 1087void AudioFlinger::ThreadBase::processConfigEvents() 1088{ 1089 mLock.lock(); 1090 while(!mConfigEvents.isEmpty()) { 1091 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1092 ConfigEvent configEvent = mConfigEvents[0]; 1093 mConfigEvents.removeAt(0); 1094 // release mLock before locking AudioFlinger mLock: lock order is always 1095 // AudioFlinger then ThreadBase to avoid cross deadlock 1096 mLock.unlock(); 1097 mAudioFlinger->mLock.lock(); 1098 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1099 mAudioFlinger->mLock.unlock(); 1100 mLock.lock(); 1101 } 1102 mLock.unlock(); 1103} 1104 1105status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1106{ 1107 const size_t SIZE = 256; 1108 char buffer[SIZE]; 1109 String8 result; 1110 1111 bool locked = tryLock(mLock); 1112 if (!locked) { 1113 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1114 write(fd, buffer, strlen(buffer)); 1115 } 1116 1117 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1122 result.append(buffer); 1123 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1124 result.append(buffer); 1125 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1126 result.append(buffer); 1127 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1128 result.append(buffer); 1129 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1130 result.append(buffer); 1131 1132 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1133 result.append(buffer); 1134 result.append(" Index Command"); 1135 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1136 snprintf(buffer, SIZE, "\n %02d ", i); 1137 result.append(buffer); 1138 result.append(mNewParameters[i]); 1139 } 1140 1141 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1142 result.append(buffer); 1143 snprintf(buffer, SIZE, " Index event param\n"); 1144 result.append(buffer); 1145 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1146 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1147 result.append(buffer); 1148 } 1149 result.append("\n"); 1150 1151 write(fd, result.string(), result.size()); 1152 1153 if (locked) { 1154 mLock.unlock(); 1155 } 1156 return NO_ERROR; 1157} 1158 1159status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1160{ 1161 const size_t SIZE = 256; 1162 char buffer[SIZE]; 1163 String8 result; 1164 1165 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1166 write(fd, buffer, strlen(buffer)); 1167 1168 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1169 sp<EffectChain> chain = mEffectChains[i]; 1170 if (chain != 0) { 1171 chain->dump(fd, args); 1172 } 1173 } 1174 return NO_ERROR; 1175} 1176 1177void AudioFlinger::ThreadBase::acquireWakeLock() 1178{ 1179 Mutex::Autolock _l(mLock); 1180 acquireWakeLock_l(); 1181} 1182 1183void AudioFlinger::ThreadBase::acquireWakeLock_l() 1184{ 1185 if (mPowerManager == 0) { 1186 // use checkService() to avoid blocking if power service is not up yet 1187 sp<IBinder> binder = 1188 defaultServiceManager()->checkService(String16("power")); 1189 if (binder == 0) { 1190 ALOGW("Thread %s cannot connect to the power manager service", mName); 1191 } else { 1192 mPowerManager = interface_cast<IPowerManager>(binder); 1193 binder->linkToDeath(mDeathRecipient); 1194 } 1195 } 1196 if (mPowerManager != 0) { 1197 sp<IBinder> binder = new BBinder(); 1198 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1199 binder, 1200 String16(mName)); 1201 if (status == NO_ERROR) { 1202 mWakeLockToken = binder; 1203 } 1204 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1205 } 1206} 1207 1208void AudioFlinger::ThreadBase::releaseWakeLock() 1209{ 1210 Mutex::Autolock _l(mLock); 1211 releaseWakeLock_l(); 1212} 1213 1214void AudioFlinger::ThreadBase::releaseWakeLock_l() 1215{ 1216 if (mWakeLockToken != 0) { 1217 ALOGV("releaseWakeLock_l() %s", mName); 1218 if (mPowerManager != 0) { 1219 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1220 } 1221 mWakeLockToken.clear(); 1222 } 1223} 1224 1225void AudioFlinger::ThreadBase::clearPowerManager() 1226{ 1227 Mutex::Autolock _l(mLock); 1228 releaseWakeLock_l(); 1229 mPowerManager.clear(); 1230} 1231 1232void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1233{ 1234 sp<ThreadBase> thread = mThread.promote(); 1235 if (thread != 0) { 1236 thread->clearPowerManager(); 1237 } 1238 ALOGW("power manager service died !!!"); 1239} 1240 1241void AudioFlinger::ThreadBase::setEffectSuspended( 1242 const effect_uuid_t *type, bool suspend, int sessionId) 1243{ 1244 Mutex::Autolock _l(mLock); 1245 setEffectSuspended_l(type, suspend, sessionId); 1246} 1247 1248void AudioFlinger::ThreadBase::setEffectSuspended_l( 1249 const effect_uuid_t *type, bool suspend, int sessionId) 1250{ 1251 sp<EffectChain> chain; 1252 chain = getEffectChain_l(sessionId); 1253 if (chain != 0) { 1254 if (type != NULL) { 1255 chain->setEffectSuspended_l(type, suspend); 1256 } else { 1257 chain->setEffectSuspendedAll_l(suspend); 1258 } 1259 } 1260 1261 updateSuspendedSessions_l(type, suspend, sessionId); 1262} 1263 1264void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1265{ 1266 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1267 if (index < 0) { 1268 return; 1269 } 1270 1271 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1272 mSuspendedSessions.editValueAt(index); 1273 1274 for (size_t i = 0; i < sessionEffects.size(); i++) { 1275 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1276 for (int j = 0; j < desc->mRefCount; j++) { 1277 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1278 chain->setEffectSuspendedAll_l(true); 1279 } else { 1280 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1281 desc->mType.timeLow); 1282 chain->setEffectSuspended_l(&desc->mType, true); 1283 } 1284 } 1285 } 1286} 1287 1288void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1289 bool suspend, 1290 int sessionId) 1291{ 1292 int index = mSuspendedSessions.indexOfKey(sessionId); 1293 1294 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1295 1296 if (suspend) { 1297 if (index >= 0) { 1298 sessionEffects = mSuspendedSessions.editValueAt(index); 1299 } else { 1300 mSuspendedSessions.add(sessionId, sessionEffects); 1301 } 1302 } else { 1303 if (index < 0) { 1304 return; 1305 } 1306 sessionEffects = mSuspendedSessions.editValueAt(index); 1307 } 1308 1309 1310 int key = EffectChain::kKeyForSuspendAll; 1311 if (type != NULL) { 1312 key = type->timeLow; 1313 } 1314 index = sessionEffects.indexOfKey(key); 1315 1316 sp <SuspendedSessionDesc> desc; 1317 if (suspend) { 1318 if (index >= 0) { 1319 desc = sessionEffects.valueAt(index); 1320 } else { 1321 desc = new SuspendedSessionDesc(); 1322 if (type != NULL) { 1323 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1324 } 1325 sessionEffects.add(key, desc); 1326 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1327 } 1328 desc->mRefCount++; 1329 } else { 1330 if (index < 0) { 1331 return; 1332 } 1333 desc = sessionEffects.valueAt(index); 1334 if (--desc->mRefCount == 0) { 1335 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1336 sessionEffects.removeItemsAt(index); 1337 if (sessionEffects.isEmpty()) { 1338 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1339 sessionId); 1340 mSuspendedSessions.removeItem(sessionId); 1341 } 1342 } 1343 } 1344 if (!sessionEffects.isEmpty()) { 1345 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1346 } 1347} 1348 1349void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1350 bool enabled, 1351 int sessionId) 1352{ 1353 Mutex::Autolock _l(mLock); 1354 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1355} 1356 1357void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1358 bool enabled, 1359 int sessionId) 1360{ 1361 if (mType != RECORD) { 1362 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1363 // another session. This gives the priority to well behaved effect control panels 1364 // and applications not using global effects. 1365 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1366 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1367 } 1368 } 1369 1370 sp<EffectChain> chain = getEffectChain_l(sessionId); 1371 if (chain != 0) { 1372 chain->checkSuspendOnEffectEnabled(effect, enabled); 1373 } 1374} 1375 1376// ---------------------------------------------------------------------------- 1377 1378AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1379 AudioStreamOut* output, 1380 int id, 1381 uint32_t device, 1382 type_t type) 1383 : ThreadBase(audioFlinger, id, device, type), 1384 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1385 // Assumes constructor is called by AudioFlinger with it's mLock held, 1386 // but it would be safer to explicitly pass initial masterMute as parameter 1387 mMasterMute(audioFlinger->masterMute_l()), 1388 // mStreamTypes[] initialized in constructor body 1389 mOutput(output), 1390 // Assumes constructor is called by AudioFlinger with it's mLock held, 1391 // but it would be safer to explicitly pass initial masterVolume as parameter 1392 mMasterVolume(audioFlinger->masterVolume_l()), 1393 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1394{ 1395 snprintf(mName, kNameLength, "AudioOut_%d", id); 1396 1397 readOutputParameters(); 1398 1399 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1400 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1401 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1402 stream = (audio_stream_type_t) (stream + 1)) { 1403 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1404 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1405 // initialized by stream_type_t default constructor 1406 // mStreamTypes[stream].valid = true; 1407 } 1408} 1409 1410AudioFlinger::PlaybackThread::~PlaybackThread() 1411{ 1412 delete [] mMixBuffer; 1413} 1414 1415status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1416{ 1417 dumpInternals(fd, args); 1418 dumpTracks(fd, args); 1419 dumpEffectChains(fd, args); 1420 return NO_ERROR; 1421} 1422 1423status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1424{ 1425 const size_t SIZE = 256; 1426 char buffer[SIZE]; 1427 String8 result; 1428 1429 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1430 result.append(buffer); 1431 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1432 for (size_t i = 0; i < mTracks.size(); ++i) { 1433 sp<Track> track = mTracks[i]; 1434 if (track != 0) { 1435 track->dump(buffer, SIZE); 1436 result.append(buffer); 1437 } 1438 } 1439 1440 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1441 result.append(buffer); 1442 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1443 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1444 sp<Track> track = mActiveTracks[i].promote(); 1445 if (track != 0) { 1446 track->dump(buffer, SIZE); 1447 result.append(buffer); 1448 } 1449 } 1450 write(fd, result.string(), result.size()); 1451 return NO_ERROR; 1452} 1453 1454status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1455{ 1456 const size_t SIZE = 256; 1457 char buffer[SIZE]; 1458 String8 result; 1459 1460 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1461 result.append(buffer); 1462 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1463 result.append(buffer); 1464 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1465 result.append(buffer); 1466 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1467 result.append(buffer); 1468 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1469 result.append(buffer); 1470 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1471 result.append(buffer); 1472 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1473 result.append(buffer); 1474 write(fd, result.string(), result.size()); 1475 1476 dumpBase(fd, args); 1477 1478 return NO_ERROR; 1479} 1480 1481// Thread virtuals 1482status_t AudioFlinger::PlaybackThread::readyToRun() 1483{ 1484 status_t status = initCheck(); 1485 if (status == NO_ERROR) { 1486 ALOGI("AudioFlinger's thread %p ready to run", this); 1487 } else { 1488 ALOGE("No working audio driver found."); 1489 } 1490 return status; 1491} 1492 1493void AudioFlinger::PlaybackThread::onFirstRef() 1494{ 1495 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1496} 1497 1498// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1499sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1500 const sp<AudioFlinger::Client>& client, 1501 audio_stream_type_t streamType, 1502 uint32_t sampleRate, 1503 audio_format_t format, 1504 uint32_t channelMask, 1505 int frameCount, 1506 const sp<IMemory>& sharedBuffer, 1507 int sessionId, 1508 status_t *status) 1509{ 1510 sp<Track> track; 1511 status_t lStatus; 1512 1513 if (mType == DIRECT) { 1514 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1515 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1516 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1517 "for output %p with format %d", 1518 sampleRate, format, channelMask, mOutput, mFormat); 1519 lStatus = BAD_VALUE; 1520 goto Exit; 1521 } 1522 } 1523 } else { 1524 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1525 if (sampleRate > mSampleRate*2) { 1526 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1527 lStatus = BAD_VALUE; 1528 goto Exit; 1529 } 1530 } 1531 1532 lStatus = initCheck(); 1533 if (lStatus != NO_ERROR) { 1534 ALOGE("Audio driver not initialized."); 1535 goto Exit; 1536 } 1537 1538 { // scope for mLock 1539 Mutex::Autolock _l(mLock); 1540 1541 // all tracks in same audio session must share the same routing strategy otherwise 1542 // conflicts will happen when tracks are moved from one output to another by audio policy 1543 // manager 1544 uint32_t strategy = 1545 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1546 for (size_t i = 0; i < mTracks.size(); ++i) { 1547 sp<Track> t = mTracks[i]; 1548 if (t != 0) { 1549 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1550 if (sessionId == t->sessionId() && strategy != actual) { 1551 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1552 strategy, actual); 1553 lStatus = BAD_VALUE; 1554 goto Exit; 1555 } 1556 } 1557 } 1558 1559 track = new Track(this, client, streamType, sampleRate, format, 1560 channelMask, frameCount, sharedBuffer, sessionId); 1561 if (track->getCblk() == NULL || track->name() < 0) { 1562 lStatus = NO_MEMORY; 1563 goto Exit; 1564 } 1565 mTracks.add(track); 1566 1567 sp<EffectChain> chain = getEffectChain_l(sessionId); 1568 if (chain != 0) { 1569 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1570 track->setMainBuffer(chain->inBuffer()); 1571 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1572 chain->incTrackCnt(); 1573 } 1574 1575 // invalidate track immediately if the stream type was moved to another thread since 1576 // createTrack() was called by the client process. 1577 if (!mStreamTypes[streamType].valid) { 1578 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1579 this, streamType); 1580 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1581 } 1582 } 1583 lStatus = NO_ERROR; 1584 1585Exit: 1586 if(status) { 1587 *status = lStatus; 1588 } 1589 return track; 1590} 1591 1592uint32_t AudioFlinger::PlaybackThread::latency() const 1593{ 1594 Mutex::Autolock _l(mLock); 1595 if (initCheck() == NO_ERROR) { 1596 return mOutput->stream->get_latency(mOutput->stream); 1597 } else { 1598 return 0; 1599 } 1600} 1601 1602status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1603{ 1604 mMasterVolume = value; 1605 return NO_ERROR; 1606} 1607 1608status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1609{ 1610 mMasterMute = muted; 1611 return NO_ERROR; 1612} 1613 1614float AudioFlinger::PlaybackThread::masterVolume() const 1615{ 1616 return mMasterVolume; 1617} 1618 1619bool AudioFlinger::PlaybackThread::masterMute() const 1620{ 1621 return mMasterMute; 1622} 1623 1624status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1625{ 1626 mStreamTypes[stream].volume = value; 1627 return NO_ERROR; 1628} 1629 1630status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1631{ 1632 mStreamTypes[stream].mute = muted; 1633 return NO_ERROR; 1634} 1635 1636float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1637{ 1638 return mStreamTypes[stream].volume; 1639} 1640 1641bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1642{ 1643 return mStreamTypes[stream].mute; 1644} 1645 1646// addTrack_l() must be called with ThreadBase::mLock held 1647status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1648{ 1649 status_t status = ALREADY_EXISTS; 1650 1651 // set retry count for buffer fill 1652 track->mRetryCount = kMaxTrackStartupRetries; 1653 if (mActiveTracks.indexOf(track) < 0) { 1654 // the track is newly added, make sure it fills up all its 1655 // buffers before playing. This is to ensure the client will 1656 // effectively get the latency it requested. 1657 track->mFillingUpStatus = Track::FS_FILLING; 1658 track->mResetDone = false; 1659 mActiveTracks.add(track); 1660 if (track->mainBuffer() != mMixBuffer) { 1661 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1662 if (chain != 0) { 1663 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1664 chain->incActiveTrackCnt(); 1665 } 1666 } 1667 1668 status = NO_ERROR; 1669 } 1670 1671 ALOGV("mWaitWorkCV.broadcast"); 1672 mWaitWorkCV.broadcast(); 1673 1674 return status; 1675} 1676 1677// destroyTrack_l() must be called with ThreadBase::mLock held 1678void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1679{ 1680 track->mState = TrackBase::TERMINATED; 1681 if (mActiveTracks.indexOf(track) < 0) { 1682 removeTrack_l(track); 1683 } 1684} 1685 1686void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1687{ 1688 mTracks.remove(track); 1689 deleteTrackName_l(track->name()); 1690 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1691 if (chain != 0) { 1692 chain->decTrackCnt(); 1693 } 1694} 1695 1696String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1697{ 1698 String8 out_s8 = String8(""); 1699 char *s; 1700 1701 Mutex::Autolock _l(mLock); 1702 if (initCheck() != NO_ERROR) { 1703 return out_s8; 1704 } 1705 1706 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1707 out_s8 = String8(s); 1708 free(s); 1709 return out_s8; 1710} 1711 1712// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1713void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1714 AudioSystem::OutputDescriptor desc; 1715 void *param2 = NULL; 1716 1717 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1718 1719 switch (event) { 1720 case AudioSystem::OUTPUT_OPENED: 1721 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1722 desc.channels = mChannelMask; 1723 desc.samplingRate = mSampleRate; 1724 desc.format = mFormat; 1725 desc.frameCount = mFrameCount; 1726 desc.latency = latency(); 1727 param2 = &desc; 1728 break; 1729 1730 case AudioSystem::STREAM_CONFIG_CHANGED: 1731 param2 = ¶m; 1732 case AudioSystem::OUTPUT_CLOSED: 1733 default: 1734 break; 1735 } 1736 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1737} 1738 1739void AudioFlinger::PlaybackThread::readOutputParameters() 1740{ 1741 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1742 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1743 mChannelCount = (uint16_t)popcount(mChannelMask); 1744 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1745 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1746 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1747 1748 // FIXME - Current mixer implementation only supports stereo output: Always 1749 // Allocate a stereo buffer even if HW output is mono. 1750 delete[] mMixBuffer; 1751 mMixBuffer = new int16_t[mFrameCount * 2]; 1752 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1753 1754 // force reconfiguration of effect chains and engines to take new buffer size and audio 1755 // parameters into account 1756 // Note that mLock is not held when readOutputParameters() is called from the constructor 1757 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1758 // matter. 1759 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1760 Vector< sp<EffectChain> > effectChains = mEffectChains; 1761 for (size_t i = 0; i < effectChains.size(); i ++) { 1762 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1763 } 1764} 1765 1766status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1767{ 1768 if (halFrames == NULL || dspFrames == NULL) { 1769 return BAD_VALUE; 1770 } 1771 Mutex::Autolock _l(mLock); 1772 if (initCheck() != NO_ERROR) { 1773 return INVALID_OPERATION; 1774 } 1775 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1776 1777 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1778} 1779 1780uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1781{ 1782 Mutex::Autolock _l(mLock); 1783 uint32_t result = 0; 1784 if (getEffectChain_l(sessionId) != 0) { 1785 result = EFFECT_SESSION; 1786 } 1787 1788 for (size_t i = 0; i < mTracks.size(); ++i) { 1789 sp<Track> track = mTracks[i]; 1790 if (sessionId == track->sessionId() && 1791 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1792 result |= TRACK_SESSION; 1793 break; 1794 } 1795 } 1796 1797 return result; 1798} 1799 1800uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1801{ 1802 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1803 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1804 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1805 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1806 } 1807 for (size_t i = 0; i < mTracks.size(); i++) { 1808 sp<Track> track = mTracks[i]; 1809 if (sessionId == track->sessionId() && 1810 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1811 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1812 } 1813 } 1814 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1815} 1816 1817 1818AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1819{ 1820 Mutex::Autolock _l(mLock); 1821 return mOutput; 1822} 1823 1824AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1825{ 1826 Mutex::Autolock _l(mLock); 1827 AudioStreamOut *output = mOutput; 1828 mOutput = NULL; 1829 return output; 1830} 1831 1832// this method must always be called either with ThreadBase mLock held or inside the thread loop 1833audio_stream_t* AudioFlinger::PlaybackThread::stream() 1834{ 1835 if (mOutput == NULL) { 1836 return NULL; 1837 } 1838 return &mOutput->stream->common; 1839} 1840 1841uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1842{ 1843 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1844 // decoding and transfer time. So sleeping for half of the latency would likely cause 1845 // underruns 1846 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1847 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1848 } else { 1849 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1850 } 1851} 1852 1853// ---------------------------------------------------------------------------- 1854 1855AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1856 int id, uint32_t device, type_t type) 1857 : PlaybackThread(audioFlinger, output, id, device, type), 1858 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1859 mPrevMixerStatus(MIXER_IDLE) 1860{ 1861 // FIXME - Current mixer implementation only supports stereo output 1862 if (mChannelCount == 1) { 1863 ALOGE("Invalid audio hardware channel count"); 1864 } 1865} 1866 1867AudioFlinger::MixerThread::~MixerThread() 1868{ 1869 delete mAudioMixer; 1870} 1871 1872bool AudioFlinger::MixerThread::threadLoop() 1873{ 1874 Vector< sp<Track> > tracksToRemove; 1875 mixer_state mixerStatus = MIXER_IDLE; 1876 nsecs_t standbyTime = systemTime(); 1877 size_t mixBufferSize = mFrameCount * mFrameSize; 1878 // FIXME: Relaxed timing because of a certain device that can't meet latency 1879 // Should be reduced to 2x after the vendor fixes the driver issue 1880 // increase threshold again due to low power audio mode. The way this warning threshold is 1881 // calculated and its usefulness should be reconsidered anyway. 1882 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1883 nsecs_t lastWarning = 0; 1884 bool longStandbyExit = false; 1885 uint32_t activeSleepTime = activeSleepTimeUs(); 1886 uint32_t idleSleepTime = idleSleepTimeUs(); 1887 uint32_t sleepTime = idleSleepTime; 1888 uint32_t sleepTimeShift = 0; 1889 Vector< sp<EffectChain> > effectChains; 1890#ifdef DEBUG_CPU_USAGE 1891 ThreadCpuUsage cpu; 1892 const CentralTendencyStatistics& stats = cpu.statistics(); 1893#endif 1894 1895 acquireWakeLock(); 1896 1897 while (!exitPending()) 1898 { 1899#ifdef DEBUG_CPU_USAGE 1900 cpu.sampleAndEnable(); 1901 unsigned n = stats.n(); 1902 // cpu.elapsed() is expensive, so don't call it every loop 1903 if ((n & 127) == 1) { 1904 long long elapsed = cpu.elapsed(); 1905 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1906 double perLoop = elapsed / (double) n; 1907 double perLoop100 = perLoop * 0.01; 1908 double mean = stats.mean(); 1909 double stddev = stats.stddev(); 1910 double minimum = stats.minimum(); 1911 double maximum = stats.maximum(); 1912 cpu.resetStatistics(); 1913 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", 1914 elapsed * .000000001, n, perLoop * .000001, 1915 mean * .001, 1916 stddev * .001, 1917 minimum * .001, 1918 maximum * .001, 1919 mean / perLoop100, 1920 stddev / perLoop100, 1921 minimum / perLoop100, 1922 maximum / perLoop100); 1923 } 1924 } 1925#endif 1926 processConfigEvents(); 1927 1928 mixerStatus = MIXER_IDLE; 1929 { // scope for mLock 1930 1931 Mutex::Autolock _l(mLock); 1932 1933 if (checkForNewParameters_l()) { 1934 mixBufferSize = mFrameCount * mFrameSize; 1935 // FIXME: Relaxed timing because of a certain device that can't meet latency 1936 // Should be reduced to 2x after the vendor fixes the driver issue 1937 // increase threshold again due to low power audio mode. The way this warning 1938 // threshold is calculated and its usefulness should be reconsidered anyway. 1939 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1940 activeSleepTime = activeSleepTimeUs(); 1941 idleSleepTime = idleSleepTimeUs(); 1942 } 1943 1944 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1945 1946 // put audio hardware into standby after short delay 1947 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1948 mSuspended)) { 1949 if (!mStandby) { 1950 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1951 mOutput->stream->common.standby(&mOutput->stream->common); 1952 mStandby = true; 1953 mBytesWritten = 0; 1954 } 1955 1956 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1957 // we're about to wait, flush the binder command buffer 1958 IPCThreadState::self()->flushCommands(); 1959 1960 if (exitPending()) break; 1961 1962 releaseWakeLock_l(); 1963 // wait until we have something to do... 1964 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1965 mWaitWorkCV.wait(mLock); 1966 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1967 acquireWakeLock_l(); 1968 1969 mPrevMixerStatus = MIXER_IDLE; 1970 if (!mMasterMute) { 1971 char value[PROPERTY_VALUE_MAX]; 1972 property_get("ro.audio.silent", value, "0"); 1973 if (atoi(value)) { 1974 ALOGD("Silence is golden"); 1975 setMasterMute(true); 1976 } 1977 } 1978 1979 standbyTime = systemTime() + kStandbyTimeInNsecs; 1980 sleepTime = idleSleepTime; 1981 sleepTimeShift = 0; 1982 continue; 1983 } 1984 } 1985 1986 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1987 1988 // prevent any changes in effect chain list and in each effect chain 1989 // during mixing and effect process as the audio buffers could be deleted 1990 // or modified if an effect is created or deleted 1991 lockEffectChains_l(effectChains); 1992 } 1993 1994 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1995 // mix buffers... 1996 mAudioMixer->process(); 1997 // increase sleep time progressively when application underrun condition clears. 1998 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1999 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2000 // such that we would underrun the audio HAL. 2001 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2002 sleepTimeShift--; 2003 } 2004 sleepTime = 0; 2005 standbyTime = systemTime() + kStandbyTimeInNsecs; 2006 //TODO: delay standby when effects have a tail 2007 } else { 2008 // If no tracks are ready, sleep once for the duration of an output 2009 // buffer size, then write 0s to the output 2010 if (sleepTime == 0) { 2011 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2012 sleepTime = activeSleepTime >> sleepTimeShift; 2013 if (sleepTime < kMinThreadSleepTimeUs) { 2014 sleepTime = kMinThreadSleepTimeUs; 2015 } 2016 // reduce sleep time in case of consecutive application underruns to avoid 2017 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2018 // duration we would end up writing less data than needed by the audio HAL if 2019 // the condition persists. 2020 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2021 sleepTimeShift++; 2022 } 2023 } else { 2024 sleepTime = idleSleepTime; 2025 } 2026 } else if (mBytesWritten != 0 || 2027 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2028 memset (mMixBuffer, 0, mixBufferSize); 2029 sleepTime = 0; 2030 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2031 } 2032 // TODO add standby time extension fct of effect tail 2033 } 2034 2035 if (mSuspended) { 2036 sleepTime = suspendSleepTimeUs(); 2037 } 2038 // sleepTime == 0 means we must write to audio hardware 2039 if (sleepTime == 0) { 2040 for (size_t i = 0; i < effectChains.size(); i ++) { 2041 effectChains[i]->process_l(); 2042 } 2043 // enable changes in effect chain 2044 unlockEffectChains(effectChains); 2045 mLastWriteTime = systemTime(); 2046 mInWrite = true; 2047 mBytesWritten += mixBufferSize; 2048 2049 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2050 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2051 mNumWrites++; 2052 mInWrite = false; 2053 nsecs_t now = systemTime(); 2054 nsecs_t delta = now - mLastWriteTime; 2055 if (!mStandby && delta > maxPeriod) { 2056 mNumDelayedWrites++; 2057 if ((now - lastWarning) > kWarningThrottleNs) { 2058 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2059 ns2ms(delta), mNumDelayedWrites, this); 2060 lastWarning = now; 2061 } 2062 if (mStandby) { 2063 longStandbyExit = true; 2064 } 2065 } 2066 mStandby = false; 2067 } else { 2068 // enable changes in effect chain 2069 unlockEffectChains(effectChains); 2070 usleep(sleepTime); 2071 } 2072 2073 // finally let go of all our tracks, without the lock held 2074 // since we can't guarantee the destructors won't acquire that 2075 // same lock. 2076 tracksToRemove.clear(); 2077 2078 // Effect chains will be actually deleted here if they were removed from 2079 // mEffectChains list during mixing or effects processing 2080 effectChains.clear(); 2081 } 2082 2083 if (!mStandby) { 2084 mOutput->stream->common.standby(&mOutput->stream->common); 2085 } 2086 2087 releaseWakeLock(); 2088 2089 ALOGV("MixerThread %p exiting", this); 2090 return false; 2091} 2092 2093// prepareTracks_l() must be called with ThreadBase::mLock held 2094AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2095 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2096{ 2097 2098 mixer_state mixerStatus = MIXER_IDLE; 2099 // find out which tracks need to be processed 2100 size_t count = activeTracks.size(); 2101 size_t mixedTracks = 0; 2102 size_t tracksWithEffect = 0; 2103 2104 float masterVolume = mMasterVolume; 2105 bool masterMute = mMasterMute; 2106 2107 if (masterMute) { 2108 masterVolume = 0; 2109 } 2110 // Delegate master volume control to effect in output mix effect chain if needed 2111 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2112 if (chain != 0) { 2113 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2114 chain->setVolume_l(&v, &v); 2115 masterVolume = (float)((v + (1 << 23)) >> 24); 2116 chain.clear(); 2117 } 2118 2119 for (size_t i=0 ; i<count ; i++) { 2120 sp<Track> t = activeTracks[i].promote(); 2121 if (t == 0) continue; 2122 2123 // this const just means the local variable doesn't change 2124 Track* const track = t.get(); 2125 audio_track_cblk_t* cblk = track->cblk(); 2126 2127 // The first time a track is added we wait 2128 // for all its buffers to be filled before processing it 2129 int name = track->name(); 2130 // make sure that we have enough frames to mix one full buffer. 2131 // enforce this condition only once to enable draining the buffer in case the client 2132 // app does not call stop() and relies on underrun to stop: 2133 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2134 // during last round 2135 uint32_t minFrames = 1; 2136 if (!track->isStopped() && !track->isPausing() && 2137 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2138 if (t->sampleRate() == (int)mSampleRate) { 2139 minFrames = mFrameCount; 2140 } else { 2141 // +1 for rounding and +1 for additional sample needed for interpolation 2142 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2143 // add frames already consumed but not yet released by the resampler 2144 // because cblk->framesReady() will include these frames 2145 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2146 // the minimum track buffer size is normally twice the number of frames necessary 2147 // to fill one buffer and the resampler should not leave more than one buffer worth 2148 // of unreleased frames after each pass, but just in case... 2149 ALOG_ASSERT(minFrames <= cblk->frameCount); 2150 } 2151 } 2152 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2153 !track->isPaused() && !track->isTerminated()) 2154 { 2155 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2156 2157 mixedTracks++; 2158 2159 // track->mainBuffer() != mMixBuffer means there is an effect chain 2160 // connected to the track 2161 chain.clear(); 2162 if (track->mainBuffer() != mMixBuffer) { 2163 chain = getEffectChain_l(track->sessionId()); 2164 // Delegate volume control to effect in track effect chain if needed 2165 if (chain != 0) { 2166 tracksWithEffect++; 2167 } else { 2168 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2169 name, track->sessionId()); 2170 } 2171 } 2172 2173 2174 int param = AudioMixer::VOLUME; 2175 if (track->mFillingUpStatus == Track::FS_FILLED) { 2176 // no ramp for the first volume setting 2177 track->mFillingUpStatus = Track::FS_ACTIVE; 2178 if (track->mState == TrackBase::RESUMING) { 2179 track->mState = TrackBase::ACTIVE; 2180 param = AudioMixer::RAMP_VOLUME; 2181 } 2182 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2183 } else if (cblk->server != 0) { 2184 // If the track is stopped before the first frame was mixed, 2185 // do not apply ramp 2186 param = AudioMixer::RAMP_VOLUME; 2187 } 2188 2189 // compute volume for this track 2190 uint32_t vl, vr, va; 2191 if (track->isMuted() || track->isPausing() || 2192 mStreamTypes[track->type()].mute) { 2193 vl = vr = va = 0; 2194 if (track->isPausing()) { 2195 track->setPaused(); 2196 } 2197 } else { 2198 2199 // read original volumes with volume control 2200 float typeVolume = mStreamTypes[track->type()].volume; 2201 float v = masterVolume * typeVolume; 2202 uint32_t vlr = cblk->getVolumeLR(); 2203 vl = vlr & 0xFFFF; 2204 vr = vlr >> 16; 2205 // track volumes come from shared memory, so can't be trusted and must be clamped 2206 if (vl > MAX_GAIN_INT) { 2207 ALOGV("Track left volume out of range: %04X", vl); 2208 vl = MAX_GAIN_INT; 2209 } 2210 if (vr > MAX_GAIN_INT) { 2211 ALOGV("Track right volume out of range: %04X", vr); 2212 vr = MAX_GAIN_INT; 2213 } 2214 // now apply the master volume and stream type volume 2215 vl = (uint32_t)(v * vl) << 12; 2216 vr = (uint32_t)(v * vr) << 12; 2217 // assuming master volume and stream type volume each go up to 1.0, 2218 // vl and vr are now in 8.24 format 2219 2220 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2221 // send level comes from shared memory and so may be corrupt 2222 if (sendLevel >= MAX_GAIN_INT) { 2223 ALOGV("Track send level out of range: %04X", sendLevel); 2224 sendLevel = MAX_GAIN_INT; 2225 } 2226 va = (uint32_t)(v * sendLevel); 2227 } 2228 // Delegate volume control to effect in track effect chain if needed 2229 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2230 // Do not ramp volume if volume is controlled by effect 2231 param = AudioMixer::VOLUME; 2232 track->mHasVolumeController = true; 2233 } else { 2234 // force no volume ramp when volume controller was just disabled or removed 2235 // from effect chain to avoid volume spike 2236 if (track->mHasVolumeController) { 2237 param = AudioMixer::VOLUME; 2238 } 2239 track->mHasVolumeController = false; 2240 } 2241 2242 // Convert volumes from 8.24 to 4.12 format 2243 int16_t left, right, aux; 2244 // This additional clamping is needed in case chain->setVolume_l() overshot 2245 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2246 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2247 left = int16_t(v_clamped); 2248 v_clamped = (vr + (1 << 11)) >> 12; 2249 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2250 right = int16_t(v_clamped); 2251 2252 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2253 aux = int16_t(va); 2254 2255 // XXX: these things DON'T need to be done each time 2256 mAudioMixer->setBufferProvider(name, track); 2257 mAudioMixer->enable(name); 2258 2259 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2260 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2261 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2262 mAudioMixer->setParameter( 2263 name, 2264 AudioMixer::TRACK, 2265 AudioMixer::FORMAT, (void *)track->format()); 2266 mAudioMixer->setParameter( 2267 name, 2268 AudioMixer::TRACK, 2269 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2270 mAudioMixer->setParameter( 2271 name, 2272 AudioMixer::RESAMPLE, 2273 AudioMixer::SAMPLE_RATE, 2274 (void *)(cblk->sampleRate)); 2275 mAudioMixer->setParameter( 2276 name, 2277 AudioMixer::TRACK, 2278 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2279 mAudioMixer->setParameter( 2280 name, 2281 AudioMixer::TRACK, 2282 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2283 2284 // reset retry count 2285 track->mRetryCount = kMaxTrackRetries; 2286 // If one track is ready, set the mixer ready if: 2287 // - the mixer was not ready during previous round OR 2288 // - no other track is not ready 2289 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2290 mixerStatus != MIXER_TRACKS_ENABLED) { 2291 mixerStatus = MIXER_TRACKS_READY; 2292 } 2293 } else { 2294 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2295 if (track->isStopped()) { 2296 track->reset(); 2297 } 2298 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2299 // We have consumed all the buffers of this track. 2300 // Remove it from the list of active tracks. 2301 tracksToRemove->add(track); 2302 } else { 2303 // No buffers for this track. Give it a few chances to 2304 // fill a buffer, then remove it from active list. 2305 if (--(track->mRetryCount) <= 0) { 2306 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2307 tracksToRemove->add(track); 2308 // indicate to client process that the track was disabled because of underrun 2309 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2310 // If one track is not ready, mark the mixer also not ready if: 2311 // - the mixer was ready during previous round OR 2312 // - no other track is ready 2313 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2314 mixerStatus != MIXER_TRACKS_READY) { 2315 mixerStatus = MIXER_TRACKS_ENABLED; 2316 } 2317 } 2318 mAudioMixer->disable(name); 2319 } 2320 } 2321 2322 // remove all the tracks that need to be... 2323 count = tracksToRemove->size(); 2324 if (CC_UNLIKELY(count)) { 2325 for (size_t i=0 ; i<count ; i++) { 2326 const sp<Track>& track = tracksToRemove->itemAt(i); 2327 mActiveTracks.remove(track); 2328 if (track->mainBuffer() != mMixBuffer) { 2329 chain = getEffectChain_l(track->sessionId()); 2330 if (chain != 0) { 2331 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2332 chain->decActiveTrackCnt(); 2333 } 2334 } 2335 if (track->isTerminated()) { 2336 removeTrack_l(track); 2337 } 2338 } 2339 } 2340 2341 // mix buffer must be cleared if all tracks are connected to an 2342 // effect chain as in this case the mixer will not write to 2343 // mix buffer and track effects will accumulate into it 2344 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2345 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2346 } 2347 2348 mPrevMixerStatus = mixerStatus; 2349 return mixerStatus; 2350} 2351 2352void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2353{ 2354 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2355 this, streamType, mTracks.size()); 2356 Mutex::Autolock _l(mLock); 2357 2358 size_t size = mTracks.size(); 2359 for (size_t i = 0; i < size; i++) { 2360 sp<Track> t = mTracks[i]; 2361 if (t->type() == streamType) { 2362 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2363 t->mCblk->cv.signal(); 2364 } 2365 } 2366} 2367 2368void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2369{ 2370 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2371 this, streamType, valid); 2372 Mutex::Autolock _l(mLock); 2373 2374 mStreamTypes[streamType].valid = valid; 2375} 2376 2377// getTrackName_l() must be called with ThreadBase::mLock held 2378int AudioFlinger::MixerThread::getTrackName_l() 2379{ 2380 return mAudioMixer->getTrackName(); 2381} 2382 2383// deleteTrackName_l() must be called with ThreadBase::mLock held 2384void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2385{ 2386 ALOGV("remove track (%d) and delete from mixer", name); 2387 mAudioMixer->deleteTrackName(name); 2388} 2389 2390// checkForNewParameters_l() must be called with ThreadBase::mLock held 2391bool AudioFlinger::MixerThread::checkForNewParameters_l() 2392{ 2393 bool reconfig = false; 2394 2395 while (!mNewParameters.isEmpty()) { 2396 status_t status = NO_ERROR; 2397 String8 keyValuePair = mNewParameters[0]; 2398 AudioParameter param = AudioParameter(keyValuePair); 2399 int value; 2400 2401 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2402 reconfig = true; 2403 } 2404 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2405 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2406 status = BAD_VALUE; 2407 } else { 2408 reconfig = true; 2409 } 2410 } 2411 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2412 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2413 status = BAD_VALUE; 2414 } else { 2415 reconfig = true; 2416 } 2417 } 2418 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2419 // do not accept frame count changes if tracks are open as the track buffer 2420 // size depends on frame count and correct behavior would not be guaranteed 2421 // if frame count is changed after track creation 2422 if (!mTracks.isEmpty()) { 2423 status = INVALID_OPERATION; 2424 } else { 2425 reconfig = true; 2426 } 2427 } 2428 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2429 // when changing the audio output device, call addBatteryData to notify 2430 // the change 2431 if ((int)mDevice != value) { 2432 uint32_t params = 0; 2433 // check whether speaker is on 2434 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2435 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2436 } 2437 2438 int deviceWithoutSpeaker 2439 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2440 // check if any other device (except speaker) is on 2441 if (value & deviceWithoutSpeaker ) { 2442 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2443 } 2444 2445 if (params != 0) { 2446 addBatteryData(params); 2447 } 2448 } 2449 2450 // forward device change to effects that have requested to be 2451 // aware of attached audio device. 2452 mDevice = (uint32_t)value; 2453 for (size_t i = 0; i < mEffectChains.size(); i++) { 2454 mEffectChains[i]->setDevice_l(mDevice); 2455 } 2456 } 2457 2458 if (status == NO_ERROR) { 2459 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2460 keyValuePair.string()); 2461 if (!mStandby && status == INVALID_OPERATION) { 2462 mOutput->stream->common.standby(&mOutput->stream->common); 2463 mStandby = true; 2464 mBytesWritten = 0; 2465 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2466 keyValuePair.string()); 2467 } 2468 if (status == NO_ERROR && reconfig) { 2469 delete mAudioMixer; 2470 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2471 mAudioMixer = NULL; 2472 readOutputParameters(); 2473 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2474 for (size_t i = 0; i < mTracks.size() ; i++) { 2475 int name = getTrackName_l(); 2476 if (name < 0) break; 2477 mTracks[i]->mName = name; 2478 // limit track sample rate to 2 x new output sample rate 2479 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2480 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2481 } 2482 } 2483 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2484 } 2485 } 2486 2487 mNewParameters.removeAt(0); 2488 2489 mParamStatus = status; 2490 mParamCond.signal(); 2491 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2492 // already timed out waiting for the status and will never signal the condition. 2493 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2494 } 2495 return reconfig; 2496} 2497 2498status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2499{ 2500 const size_t SIZE = 256; 2501 char buffer[SIZE]; 2502 String8 result; 2503 2504 PlaybackThread::dumpInternals(fd, args); 2505 2506 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2507 result.append(buffer); 2508 write(fd, result.string(), result.size()); 2509 return NO_ERROR; 2510} 2511 2512uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2513{ 2514 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2515} 2516 2517uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2518{ 2519 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2520} 2521 2522// ---------------------------------------------------------------------------- 2523AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2524 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2525 // mLeftVolFloat, mRightVolFloat 2526 // mLeftVolShort, mRightVolShort 2527{ 2528} 2529 2530AudioFlinger::DirectOutputThread::~DirectOutputThread() 2531{ 2532} 2533 2534static inline 2535int32_t mul(int16_t in, int16_t v) 2536{ 2537#if defined(__arm__) && !defined(__thumb__) 2538 int32_t out; 2539 asm( "smulbb %[out], %[in], %[v] \n" 2540 : [out]"=r"(out) 2541 : [in]"%r"(in), [v]"r"(v) 2542 : ); 2543 return out; 2544#else 2545 return in * int32_t(v); 2546#endif 2547} 2548 2549void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2550{ 2551 // Do not apply volume on compressed audio 2552 if (!audio_is_linear_pcm(mFormat)) { 2553 return; 2554 } 2555 2556 // convert to signed 16 bit before volume calculation 2557 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2558 size_t count = mFrameCount * mChannelCount; 2559 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2560 int16_t *dst = mMixBuffer + count-1; 2561 while(count--) { 2562 *dst-- = (int16_t)(*src--^0x80) << 8; 2563 } 2564 } 2565 2566 size_t frameCount = mFrameCount; 2567 int16_t *out = mMixBuffer; 2568 if (ramp) { 2569 if (mChannelCount == 1) { 2570 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2571 int32_t vlInc = d / (int32_t)frameCount; 2572 int32_t vl = ((int32_t)mLeftVolShort << 16); 2573 do { 2574 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2575 out++; 2576 vl += vlInc; 2577 } while (--frameCount); 2578 2579 } else { 2580 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2581 int32_t vlInc = d / (int32_t)frameCount; 2582 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2583 int32_t vrInc = d / (int32_t)frameCount; 2584 int32_t vl = ((int32_t)mLeftVolShort << 16); 2585 int32_t vr = ((int32_t)mRightVolShort << 16); 2586 do { 2587 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2588 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2589 out += 2; 2590 vl += vlInc; 2591 vr += vrInc; 2592 } while (--frameCount); 2593 } 2594 } else { 2595 if (mChannelCount == 1) { 2596 do { 2597 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2598 out++; 2599 } while (--frameCount); 2600 } else { 2601 do { 2602 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2603 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2604 out += 2; 2605 } while (--frameCount); 2606 } 2607 } 2608 2609 // convert back to unsigned 8 bit after volume calculation 2610 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2611 size_t count = mFrameCount * mChannelCount; 2612 int16_t *src = mMixBuffer; 2613 uint8_t *dst = (uint8_t *)mMixBuffer; 2614 while(count--) { 2615 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2616 } 2617 } 2618 2619 mLeftVolShort = leftVol; 2620 mRightVolShort = rightVol; 2621} 2622 2623bool AudioFlinger::DirectOutputThread::threadLoop() 2624{ 2625 mixer_state mixerStatus = MIXER_IDLE; 2626 sp<Track> trackToRemove; 2627 sp<Track> activeTrack; 2628 nsecs_t standbyTime = systemTime(); 2629 int8_t *curBuf; 2630 size_t mixBufferSize = mFrameCount*mFrameSize; 2631 uint32_t activeSleepTime = activeSleepTimeUs(); 2632 uint32_t idleSleepTime = idleSleepTimeUs(); 2633 uint32_t sleepTime = idleSleepTime; 2634 // use shorter standby delay as on normal output to release 2635 // hardware resources as soon as possible 2636 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2637 2638 acquireWakeLock(); 2639 2640 while (!exitPending()) 2641 { 2642 bool rampVolume; 2643 uint16_t leftVol; 2644 uint16_t rightVol; 2645 Vector< sp<EffectChain> > effectChains; 2646 2647 processConfigEvents(); 2648 2649 mixerStatus = MIXER_IDLE; 2650 2651 { // scope for the mLock 2652 2653 Mutex::Autolock _l(mLock); 2654 2655 if (checkForNewParameters_l()) { 2656 mixBufferSize = mFrameCount*mFrameSize; 2657 activeSleepTime = activeSleepTimeUs(); 2658 idleSleepTime = idleSleepTimeUs(); 2659 standbyDelay = microseconds(activeSleepTime*2); 2660 } 2661 2662 // put audio hardware into standby after short delay 2663 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2664 mSuspended)) { 2665 // wait until we have something to do... 2666 if (!mStandby) { 2667 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2668 mOutput->stream->common.standby(&mOutput->stream->common); 2669 mStandby = true; 2670 mBytesWritten = 0; 2671 } 2672 2673 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2674 // we're about to wait, flush the binder command buffer 2675 IPCThreadState::self()->flushCommands(); 2676 2677 if (exitPending()) break; 2678 2679 releaseWakeLock_l(); 2680 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2681 mWaitWorkCV.wait(mLock); 2682 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2683 acquireWakeLock_l(); 2684 2685 if (!mMasterMute) { 2686 char value[PROPERTY_VALUE_MAX]; 2687 property_get("ro.audio.silent", value, "0"); 2688 if (atoi(value)) { 2689 ALOGD("Silence is golden"); 2690 setMasterMute(true); 2691 } 2692 } 2693 2694 standbyTime = systemTime() + standbyDelay; 2695 sleepTime = idleSleepTime; 2696 continue; 2697 } 2698 } 2699 2700 effectChains = mEffectChains; 2701 2702 // find out which tracks need to be processed 2703 if (mActiveTracks.size() != 0) { 2704 sp<Track> t = mActiveTracks[0].promote(); 2705 if (t == 0) continue; 2706 2707 Track* const track = t.get(); 2708 audio_track_cblk_t* cblk = track->cblk(); 2709 2710 // The first time a track is added we wait 2711 // for all its buffers to be filled before processing it 2712 if (cblk->framesReady() && track->isReady() && 2713 !track->isPaused() && !track->isTerminated()) 2714 { 2715 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2716 2717 if (track->mFillingUpStatus == Track::FS_FILLED) { 2718 track->mFillingUpStatus = Track::FS_ACTIVE; 2719 mLeftVolFloat = mRightVolFloat = 0; 2720 mLeftVolShort = mRightVolShort = 0; 2721 if (track->mState == TrackBase::RESUMING) { 2722 track->mState = TrackBase::ACTIVE; 2723 rampVolume = true; 2724 } 2725 } else if (cblk->server != 0) { 2726 // If the track is stopped before the first frame was mixed, 2727 // do not apply ramp 2728 rampVolume = true; 2729 } 2730 // compute volume for this track 2731 float left, right; 2732 if (track->isMuted() || mMasterMute || track->isPausing() || 2733 mStreamTypes[track->type()].mute) { 2734 left = right = 0; 2735 if (track->isPausing()) { 2736 track->setPaused(); 2737 } 2738 } else { 2739 float typeVolume = mStreamTypes[track->type()].volume; 2740 float v = mMasterVolume * typeVolume; 2741 uint32_t vlr = cblk->getVolumeLR(); 2742 float v_clamped = v * (vlr & 0xFFFF); 2743 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2744 left = v_clamped/MAX_GAIN; 2745 v_clamped = v * (vlr >> 16); 2746 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2747 right = v_clamped/MAX_GAIN; 2748 } 2749 2750 if (left != mLeftVolFloat || right != mRightVolFloat) { 2751 mLeftVolFloat = left; 2752 mRightVolFloat = right; 2753 2754 // If audio HAL implements volume control, 2755 // force software volume to nominal value 2756 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2757 left = 1.0f; 2758 right = 1.0f; 2759 } 2760 2761 // Convert volumes from float to 8.24 2762 uint32_t vl = (uint32_t)(left * (1 << 24)); 2763 uint32_t vr = (uint32_t)(right * (1 << 24)); 2764 2765 // Delegate volume control to effect in track effect chain if needed 2766 // only one effect chain can be present on DirectOutputThread, so if 2767 // there is one, the track is connected to it 2768 if (!effectChains.isEmpty()) { 2769 // Do not ramp volume if volume is controlled by effect 2770 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2771 rampVolume = false; 2772 } 2773 } 2774 2775 // Convert volumes from 8.24 to 4.12 format 2776 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2777 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2778 leftVol = (uint16_t)v_clamped; 2779 v_clamped = (vr + (1 << 11)) >> 12; 2780 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2781 rightVol = (uint16_t)v_clamped; 2782 } else { 2783 leftVol = mLeftVolShort; 2784 rightVol = mRightVolShort; 2785 rampVolume = false; 2786 } 2787 2788 // reset retry count 2789 track->mRetryCount = kMaxTrackRetriesDirect; 2790 activeTrack = t; 2791 mixerStatus = MIXER_TRACKS_READY; 2792 } else { 2793 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2794 if (track->isStopped()) { 2795 track->reset(); 2796 } 2797 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2798 // We have consumed all the buffers of this track. 2799 // Remove it from the list of active tracks. 2800 trackToRemove = track; 2801 } else { 2802 // No buffers for this track. Give it a few chances to 2803 // fill a buffer, then remove it from active list. 2804 if (--(track->mRetryCount) <= 0) { 2805 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2806 trackToRemove = track; 2807 } else { 2808 mixerStatus = MIXER_TRACKS_ENABLED; 2809 } 2810 } 2811 } 2812 } 2813 2814 // remove all the tracks that need to be... 2815 if (CC_UNLIKELY(trackToRemove != 0)) { 2816 mActiveTracks.remove(trackToRemove); 2817 if (!effectChains.isEmpty()) { 2818 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2819 trackToRemove->sessionId()); 2820 effectChains[0]->decActiveTrackCnt(); 2821 } 2822 if (trackToRemove->isTerminated()) { 2823 removeTrack_l(trackToRemove); 2824 } 2825 } 2826 2827 lockEffectChains_l(effectChains); 2828 } 2829 2830 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2831 AudioBufferProvider::Buffer buffer; 2832 size_t frameCount = mFrameCount; 2833 curBuf = (int8_t *)mMixBuffer; 2834 // output audio to hardware 2835 while (frameCount) { 2836 buffer.frameCount = frameCount; 2837 activeTrack->getNextBuffer(&buffer); 2838 if (CC_UNLIKELY(buffer.raw == NULL)) { 2839 memset(curBuf, 0, frameCount * mFrameSize); 2840 break; 2841 } 2842 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2843 frameCount -= buffer.frameCount; 2844 curBuf += buffer.frameCount * mFrameSize; 2845 activeTrack->releaseBuffer(&buffer); 2846 } 2847 sleepTime = 0; 2848 standbyTime = systemTime() + standbyDelay; 2849 } else { 2850 if (sleepTime == 0) { 2851 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2852 sleepTime = activeSleepTime; 2853 } else { 2854 sleepTime = idleSleepTime; 2855 } 2856 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2857 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2858 sleepTime = 0; 2859 } 2860 } 2861 2862 if (mSuspended) { 2863 sleepTime = suspendSleepTimeUs(); 2864 } 2865 // sleepTime == 0 means we must write to audio hardware 2866 if (sleepTime == 0) { 2867 if (mixerStatus == MIXER_TRACKS_READY) { 2868 applyVolume(leftVol, rightVol, rampVolume); 2869 } 2870 for (size_t i = 0; i < effectChains.size(); i ++) { 2871 effectChains[i]->process_l(); 2872 } 2873 unlockEffectChains(effectChains); 2874 2875 mLastWriteTime = systemTime(); 2876 mInWrite = true; 2877 mBytesWritten += mixBufferSize; 2878 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2879 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2880 mNumWrites++; 2881 mInWrite = false; 2882 mStandby = false; 2883 } else { 2884 unlockEffectChains(effectChains); 2885 usleep(sleepTime); 2886 } 2887 2888 // finally let go of removed track, without the lock held 2889 // since we can't guarantee the destructors won't acquire that 2890 // same lock. 2891 trackToRemove.clear(); 2892 activeTrack.clear(); 2893 2894 // Effect chains will be actually deleted here if they were removed from 2895 // mEffectChains list during mixing or effects processing 2896 effectChains.clear(); 2897 } 2898 2899 if (!mStandby) { 2900 mOutput->stream->common.standby(&mOutput->stream->common); 2901 } 2902 2903 releaseWakeLock(); 2904 2905 ALOGV("DirectOutputThread %p exiting", this); 2906 return false; 2907} 2908 2909// getTrackName_l() must be called with ThreadBase::mLock held 2910int AudioFlinger::DirectOutputThread::getTrackName_l() 2911{ 2912 return 0; 2913} 2914 2915// deleteTrackName_l() must be called with ThreadBase::mLock held 2916void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2917{ 2918} 2919 2920// checkForNewParameters_l() must be called with ThreadBase::mLock held 2921bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2922{ 2923 bool reconfig = false; 2924 2925 while (!mNewParameters.isEmpty()) { 2926 status_t status = NO_ERROR; 2927 String8 keyValuePair = mNewParameters[0]; 2928 AudioParameter param = AudioParameter(keyValuePair); 2929 int value; 2930 2931 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2932 // do not accept frame count changes if tracks are open as the track buffer 2933 // size depends on frame count and correct behavior would not be garantied 2934 // if frame count is changed after track creation 2935 if (!mTracks.isEmpty()) { 2936 status = INVALID_OPERATION; 2937 } else { 2938 reconfig = true; 2939 } 2940 } 2941 if (status == NO_ERROR) { 2942 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2943 keyValuePair.string()); 2944 if (!mStandby && status == INVALID_OPERATION) { 2945 mOutput->stream->common.standby(&mOutput->stream->common); 2946 mStandby = true; 2947 mBytesWritten = 0; 2948 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2949 keyValuePair.string()); 2950 } 2951 if (status == NO_ERROR && reconfig) { 2952 readOutputParameters(); 2953 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2954 } 2955 } 2956 2957 mNewParameters.removeAt(0); 2958 2959 mParamStatus = status; 2960 mParamCond.signal(); 2961 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2962 // already timed out waiting for the status and will never signal the condition. 2963 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2964 } 2965 return reconfig; 2966} 2967 2968uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2969{ 2970 uint32_t time; 2971 if (audio_is_linear_pcm(mFormat)) { 2972 time = PlaybackThread::activeSleepTimeUs(); 2973 } else { 2974 time = 10000; 2975 } 2976 return time; 2977} 2978 2979uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2980{ 2981 uint32_t time; 2982 if (audio_is_linear_pcm(mFormat)) { 2983 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2984 } else { 2985 time = 10000; 2986 } 2987 return time; 2988} 2989 2990uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2991{ 2992 uint32_t time; 2993 if (audio_is_linear_pcm(mFormat)) { 2994 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2995 } else { 2996 time = 10000; 2997 } 2998 return time; 2999} 3000 3001 3002// ---------------------------------------------------------------------------- 3003 3004AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3005 AudioFlinger::MixerThread* mainThread, int id) 3006 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3007 mWaitTimeMs(UINT_MAX) 3008{ 3009 addOutputTrack(mainThread); 3010} 3011 3012AudioFlinger::DuplicatingThread::~DuplicatingThread() 3013{ 3014 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3015 mOutputTracks[i]->destroy(); 3016 } 3017 mOutputTracks.clear(); 3018} 3019 3020bool AudioFlinger::DuplicatingThread::threadLoop() 3021{ 3022 Vector< sp<Track> > tracksToRemove; 3023 mixer_state mixerStatus = MIXER_IDLE; 3024 nsecs_t standbyTime = systemTime(); 3025 size_t mixBufferSize = mFrameCount*mFrameSize; 3026 SortedVector< sp<OutputTrack> > outputTracks; 3027 uint32_t writeFrames = 0; 3028 uint32_t activeSleepTime = activeSleepTimeUs(); 3029 uint32_t idleSleepTime = idleSleepTimeUs(); 3030 uint32_t sleepTime = idleSleepTime; 3031 Vector< sp<EffectChain> > effectChains; 3032 3033 acquireWakeLock(); 3034 3035 while (!exitPending()) 3036 { 3037 processConfigEvents(); 3038 3039 mixerStatus = MIXER_IDLE; 3040 { // scope for the mLock 3041 3042 Mutex::Autolock _l(mLock); 3043 3044 if (checkForNewParameters_l()) { 3045 mixBufferSize = mFrameCount*mFrameSize; 3046 updateWaitTime(); 3047 activeSleepTime = activeSleepTimeUs(); 3048 idleSleepTime = idleSleepTimeUs(); 3049 } 3050 3051 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3052 3053 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3054 outputTracks.add(mOutputTracks[i]); 3055 } 3056 3057 // put audio hardware into standby after short delay 3058 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3059 mSuspended)) { 3060 if (!mStandby) { 3061 for (size_t i = 0; i < outputTracks.size(); i++) { 3062 outputTracks[i]->stop(); 3063 } 3064 mStandby = true; 3065 mBytesWritten = 0; 3066 } 3067 3068 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3069 // we're about to wait, flush the binder command buffer 3070 IPCThreadState::self()->flushCommands(); 3071 outputTracks.clear(); 3072 3073 if (exitPending()) break; 3074 3075 releaseWakeLock_l(); 3076 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3077 mWaitWorkCV.wait(mLock); 3078 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3079 acquireWakeLock_l(); 3080 3081 mPrevMixerStatus = MIXER_IDLE; 3082 if (!mMasterMute) { 3083 char value[PROPERTY_VALUE_MAX]; 3084 property_get("ro.audio.silent", value, "0"); 3085 if (atoi(value)) { 3086 ALOGD("Silence is golden"); 3087 setMasterMute(true); 3088 } 3089 } 3090 3091 standbyTime = systemTime() + kStandbyTimeInNsecs; 3092 sleepTime = idleSleepTime; 3093 continue; 3094 } 3095 } 3096 3097 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3098 3099 // prevent any changes in effect chain list and in each effect chain 3100 // during mixing and effect process as the audio buffers could be deleted 3101 // or modified if an effect is created or deleted 3102 lockEffectChains_l(effectChains); 3103 } 3104 3105 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3106 // mix buffers... 3107 if (outputsReady(outputTracks)) { 3108 mAudioMixer->process(); 3109 } else { 3110 memset(mMixBuffer, 0, mixBufferSize); 3111 } 3112 sleepTime = 0; 3113 writeFrames = mFrameCount; 3114 } else { 3115 if (sleepTime == 0) { 3116 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3117 sleepTime = activeSleepTime; 3118 } else { 3119 sleepTime = idleSleepTime; 3120 } 3121 } else if (mBytesWritten != 0) { 3122 // flush remaining overflow buffers in output tracks 3123 for (size_t i = 0; i < outputTracks.size(); i++) { 3124 if (outputTracks[i]->isActive()) { 3125 sleepTime = 0; 3126 writeFrames = 0; 3127 memset(mMixBuffer, 0, mixBufferSize); 3128 break; 3129 } 3130 } 3131 } 3132 } 3133 3134 if (mSuspended) { 3135 sleepTime = suspendSleepTimeUs(); 3136 } 3137 // sleepTime == 0 means we must write to audio hardware 3138 if (sleepTime == 0) { 3139 for (size_t i = 0; i < effectChains.size(); i ++) { 3140 effectChains[i]->process_l(); 3141 } 3142 // enable changes in effect chain 3143 unlockEffectChains(effectChains); 3144 3145 standbyTime = systemTime() + kStandbyTimeInNsecs; 3146 for (size_t i = 0; i < outputTracks.size(); i++) { 3147 outputTracks[i]->write(mMixBuffer, writeFrames); 3148 } 3149 mStandby = false; 3150 mBytesWritten += mixBufferSize; 3151 } else { 3152 // enable changes in effect chain 3153 unlockEffectChains(effectChains); 3154 usleep(sleepTime); 3155 } 3156 3157 // finally let go of all our tracks, without the lock held 3158 // since we can't guarantee the destructors won't acquire that 3159 // same lock. 3160 tracksToRemove.clear(); 3161 outputTracks.clear(); 3162 3163 // Effect chains will be actually deleted here if they were removed from 3164 // mEffectChains list during mixing or effects processing 3165 effectChains.clear(); 3166 } 3167 3168 releaseWakeLock(); 3169 3170 return false; 3171} 3172 3173void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3174{ 3175 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3176 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3177 this, 3178 mSampleRate, 3179 mFormat, 3180 mChannelMask, 3181 frameCount); 3182 if (outputTrack->cblk() != NULL) { 3183 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3184 mOutputTracks.add(outputTrack); 3185 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3186 updateWaitTime(); 3187 } 3188} 3189 3190void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3191{ 3192 Mutex::Autolock _l(mLock); 3193 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3194 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3195 mOutputTracks[i]->destroy(); 3196 mOutputTracks.removeAt(i); 3197 updateWaitTime(); 3198 return; 3199 } 3200 } 3201 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3202} 3203 3204void AudioFlinger::DuplicatingThread::updateWaitTime() 3205{ 3206 mWaitTimeMs = UINT_MAX; 3207 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3208 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3209 if (strong != NULL) { 3210 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3211 if (waitTimeMs < mWaitTimeMs) { 3212 mWaitTimeMs = waitTimeMs; 3213 } 3214 } 3215 } 3216} 3217 3218 3219bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3220{ 3221 for (size_t i = 0; i < outputTracks.size(); i++) { 3222 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3223 if (thread == 0) { 3224 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3225 return false; 3226 } 3227 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3228 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3229 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3230 return false; 3231 } 3232 } 3233 return true; 3234} 3235 3236uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3237{ 3238 return (mWaitTimeMs * 1000) / 2; 3239} 3240 3241// ---------------------------------------------------------------------------- 3242 3243// TrackBase constructor must be called with AudioFlinger::mLock held 3244AudioFlinger::ThreadBase::TrackBase::TrackBase( 3245 const wp<ThreadBase>& thread, 3246 const sp<Client>& client, 3247 uint32_t sampleRate, 3248 audio_format_t format, 3249 uint32_t channelMask, 3250 int frameCount, 3251 uint32_t flags, 3252 const sp<IMemory>& sharedBuffer, 3253 int sessionId) 3254 : RefBase(), 3255 mThread(thread), 3256 mClient(client), 3257 mCblk(NULL), 3258 // mBuffer 3259 // mBufferEnd 3260 mFrameCount(0), 3261 mState(IDLE), 3262 mClientTid(-1), 3263 mFormat(format), 3264 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3265 mSessionId(sessionId) 3266 // mChannelCount 3267 // mChannelMask 3268{ 3269 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3270 3271 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3272 size_t size = sizeof(audio_track_cblk_t); 3273 uint8_t channelCount = popcount(channelMask); 3274 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3275 if (sharedBuffer == 0) { 3276 size += bufferSize; 3277 } 3278 3279 if (client != NULL) { 3280 mCblkMemory = client->heap()->allocate(size); 3281 if (mCblkMemory != 0) { 3282 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3283 if (mCblk != NULL) { // construct the shared structure in-place. 3284 new(mCblk) audio_track_cblk_t(); 3285 // clear all buffers 3286 mCblk->frameCount = frameCount; 3287 mCblk->sampleRate = sampleRate; 3288 mChannelCount = channelCount; 3289 mChannelMask = channelMask; 3290 if (sharedBuffer == 0) { 3291 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3292 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3293 // Force underrun condition to avoid false underrun callback until first data is 3294 // written to buffer (other flags are cleared) 3295 mCblk->flags = CBLK_UNDERRUN_ON; 3296 } else { 3297 mBuffer = sharedBuffer->pointer(); 3298 } 3299 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3300 } 3301 } else { 3302 ALOGE("not enough memory for AudioTrack size=%u", size); 3303 client->heap()->dump("AudioTrack"); 3304 return; 3305 } 3306 } else { 3307 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3308 // construct the shared structure in-place. 3309 new(mCblk) audio_track_cblk_t(); 3310 // clear all buffers 3311 mCblk->frameCount = frameCount; 3312 mCblk->sampleRate = sampleRate; 3313 mChannelCount = channelCount; 3314 mChannelMask = channelMask; 3315 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3316 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3317 // Force underrun condition to avoid false underrun callback until first data is 3318 // written to buffer (other flags are cleared) 3319 mCblk->flags = CBLK_UNDERRUN_ON; 3320 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3321 } 3322} 3323 3324AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3325{ 3326 if (mCblk != NULL) { 3327 if (mClient == 0) { 3328 delete mCblk; 3329 } else { 3330 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3331 } 3332 } 3333 mCblkMemory.clear(); // and free the shared memory 3334 if (mClient != NULL) { 3335 // Client destructor must run with AudioFlinger mutex locked 3336 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3337 mClient.clear(); 3338 } 3339} 3340 3341void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3342{ 3343 buffer->raw = NULL; 3344 mFrameCount = buffer->frameCount; 3345 step(); 3346 buffer->frameCount = 0; 3347} 3348 3349bool AudioFlinger::ThreadBase::TrackBase::step() { 3350 bool result; 3351 audio_track_cblk_t* cblk = this->cblk(); 3352 3353 result = cblk->stepServer(mFrameCount); 3354 if (!result) { 3355 ALOGV("stepServer failed acquiring cblk mutex"); 3356 mFlags |= STEPSERVER_FAILED; 3357 } 3358 return result; 3359} 3360 3361void AudioFlinger::ThreadBase::TrackBase::reset() { 3362 audio_track_cblk_t* cblk = this->cblk(); 3363 3364 cblk->user = 0; 3365 cblk->server = 0; 3366 cblk->userBase = 0; 3367 cblk->serverBase = 0; 3368 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3369 ALOGV("TrackBase::reset"); 3370} 3371 3372sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3373{ 3374 return mCblkMemory; 3375} 3376 3377int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3378 return (int)mCblk->sampleRate; 3379} 3380 3381int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3382 return (const int)mChannelCount; 3383} 3384 3385uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3386 return mChannelMask; 3387} 3388 3389void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3390 audio_track_cblk_t* cblk = this->cblk(); 3391 size_t frameSize = cblk->frameSize; 3392 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3393 int8_t *bufferEnd = bufferStart + frames * frameSize; 3394 3395 // Check validity of returned pointer in case the track control block would have been corrupted. 3396 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3397 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3398 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3399 server %d, serverBase %d, user %d, userBase %d", 3400 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3401 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3402 return NULL; 3403 } 3404 3405 return bufferStart; 3406} 3407 3408// ---------------------------------------------------------------------------- 3409 3410// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3411AudioFlinger::PlaybackThread::Track::Track( 3412 const wp<ThreadBase>& thread, 3413 const sp<Client>& client, 3414 audio_stream_type_t streamType, 3415 uint32_t sampleRate, 3416 audio_format_t format, 3417 uint32_t channelMask, 3418 int frameCount, 3419 const sp<IMemory>& sharedBuffer, 3420 int sessionId) 3421 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3422 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3423 mAuxEffectId(0), mHasVolumeController(false) 3424{ 3425 if (mCblk != NULL) { 3426 sp<ThreadBase> baseThread = thread.promote(); 3427 if (baseThread != 0) { 3428 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3429 mName = playbackThread->getTrackName_l(); 3430 mMainBuffer = playbackThread->mixBuffer(); 3431 } 3432 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3433 if (mName < 0) { 3434 ALOGE("no more track names available"); 3435 } 3436 mStreamType = streamType; 3437 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3438 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3439 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3440 } 3441} 3442 3443AudioFlinger::PlaybackThread::Track::~Track() 3444{ 3445 ALOGV("PlaybackThread::Track destructor"); 3446 sp<ThreadBase> thread = mThread.promote(); 3447 if (thread != 0) { 3448 Mutex::Autolock _l(thread->mLock); 3449 mState = TERMINATED; 3450 } 3451} 3452 3453void AudioFlinger::PlaybackThread::Track::destroy() 3454{ 3455 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3456 // by removing it from mTracks vector, so there is a risk that this Tracks's 3457 // desctructor is called. As the destructor needs to lock mLock, 3458 // we must acquire a strong reference on this Track before locking mLock 3459 // here so that the destructor is called only when exiting this function. 3460 // On the other hand, as long as Track::destroy() is only called by 3461 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3462 // this Track with its member mTrack. 3463 sp<Track> keep(this); 3464 { // scope for mLock 3465 sp<ThreadBase> thread = mThread.promote(); 3466 if (thread != 0) { 3467 if (!isOutputTrack()) { 3468 if (mState == ACTIVE || mState == RESUMING) { 3469 AudioSystem::stopOutput(thread->id(), 3470 (audio_stream_type_t)mStreamType, 3471 mSessionId); 3472 3473 // to track the speaker usage 3474 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3475 } 3476 AudioSystem::releaseOutput(thread->id()); 3477 } 3478 Mutex::Autolock _l(thread->mLock); 3479 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3480 playbackThread->destroyTrack_l(this); 3481 } 3482 } 3483} 3484 3485void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3486{ 3487 uint32_t vlr = mCblk->getVolumeLR(); 3488 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", 3489 mName - AudioMixer::TRACK0, 3490 (mClient == NULL) ? getpid() : mClient->pid(), 3491 mStreamType, 3492 mFormat, 3493 mChannelMask, 3494 mSessionId, 3495 mFrameCount, 3496 mState, 3497 mMute, 3498 mFillingUpStatus, 3499 mCblk->sampleRate, 3500 vlr & 0xFFFF, 3501 vlr >> 16, 3502 mCblk->server, 3503 mCblk->user, 3504 (int)mMainBuffer, 3505 (int)mAuxBuffer); 3506} 3507 3508status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3509{ 3510 audio_track_cblk_t* cblk = this->cblk(); 3511 uint32_t framesReady; 3512 uint32_t framesReq = buffer->frameCount; 3513 3514 // Check if last stepServer failed, try to step now 3515 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3516 if (!step()) goto getNextBuffer_exit; 3517 ALOGV("stepServer recovered"); 3518 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3519 } 3520 3521 framesReady = cblk->framesReady(); 3522 3523 if (CC_LIKELY(framesReady)) { 3524 uint32_t s = cblk->server; 3525 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3526 3527 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3528 if (framesReq > framesReady) { 3529 framesReq = framesReady; 3530 } 3531 if (s + framesReq > bufferEnd) { 3532 framesReq = bufferEnd - s; 3533 } 3534 3535 buffer->raw = getBuffer(s, framesReq); 3536 if (buffer->raw == NULL) goto getNextBuffer_exit; 3537 3538 buffer->frameCount = framesReq; 3539 return NO_ERROR; 3540 } 3541 3542getNextBuffer_exit: 3543 buffer->raw = NULL; 3544 buffer->frameCount = 0; 3545 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3546 return NOT_ENOUGH_DATA; 3547} 3548 3549bool AudioFlinger::PlaybackThread::Track::isReady() const { 3550 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3551 3552 if (mCblk->framesReady() >= mCblk->frameCount || 3553 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3554 mFillingUpStatus = FS_FILLED; 3555 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3556 return true; 3557 } 3558 return false; 3559} 3560 3561status_t AudioFlinger::PlaybackThread::Track::start() 3562{ 3563 status_t status = NO_ERROR; 3564 ALOGV("start(%d), calling thread %d session %d", 3565 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3566 sp<ThreadBase> thread = mThread.promote(); 3567 if (thread != 0) { 3568 Mutex::Autolock _l(thread->mLock); 3569 track_state state = mState; 3570 // here the track could be either new, or restarted 3571 // in both cases "unstop" the track 3572 if (mState == PAUSED) { 3573 mState = TrackBase::RESUMING; 3574 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3575 } else { 3576 mState = TrackBase::ACTIVE; 3577 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3578 } 3579 3580 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3581 thread->mLock.unlock(); 3582 status = AudioSystem::startOutput(thread->id(), 3583 (audio_stream_type_t)mStreamType, 3584 mSessionId); 3585 thread->mLock.lock(); 3586 3587 // to track the speaker usage 3588 if (status == NO_ERROR) { 3589 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3590 } 3591 } 3592 if (status == NO_ERROR) { 3593 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3594 playbackThread->addTrack_l(this); 3595 } else { 3596 mState = state; 3597 } 3598 } else { 3599 status = BAD_VALUE; 3600 } 3601 return status; 3602} 3603 3604void AudioFlinger::PlaybackThread::Track::stop() 3605{ 3606 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3607 sp<ThreadBase> thread = mThread.promote(); 3608 if (thread != 0) { 3609 Mutex::Autolock _l(thread->mLock); 3610 track_state state = mState; 3611 if (mState > STOPPED) { 3612 mState = STOPPED; 3613 // If the track is not active (PAUSED and buffers full), flush buffers 3614 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3615 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3616 reset(); 3617 } 3618 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3619 } 3620 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3621 thread->mLock.unlock(); 3622 AudioSystem::stopOutput(thread->id(), 3623 (audio_stream_type_t)mStreamType, 3624 mSessionId); 3625 thread->mLock.lock(); 3626 3627 // to track the speaker usage 3628 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3629 } 3630 } 3631} 3632 3633void AudioFlinger::PlaybackThread::Track::pause() 3634{ 3635 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3636 sp<ThreadBase> thread = mThread.promote(); 3637 if (thread != 0) { 3638 Mutex::Autolock _l(thread->mLock); 3639 if (mState == ACTIVE || mState == RESUMING) { 3640 mState = PAUSING; 3641 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3642 if (!isOutputTrack()) { 3643 thread->mLock.unlock(); 3644 AudioSystem::stopOutput(thread->id(), 3645 (audio_stream_type_t)mStreamType, 3646 mSessionId); 3647 thread->mLock.lock(); 3648 3649 // to track the speaker usage 3650 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3651 } 3652 } 3653 } 3654} 3655 3656void AudioFlinger::PlaybackThread::Track::flush() 3657{ 3658 ALOGV("flush(%d)", mName); 3659 sp<ThreadBase> thread = mThread.promote(); 3660 if (thread != 0) { 3661 Mutex::Autolock _l(thread->mLock); 3662 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3663 return; 3664 } 3665 // No point remaining in PAUSED state after a flush => go to 3666 // STOPPED state 3667 mState = STOPPED; 3668 3669 // do not reset the track if it is still in the process of being stopped or paused. 3670 // this will be done by prepareTracks_l() when the track is stopped. 3671 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3672 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3673 reset(); 3674 } 3675 } 3676} 3677 3678void AudioFlinger::PlaybackThread::Track::reset() 3679{ 3680 // Do not reset twice to avoid discarding data written just after a flush and before 3681 // the audioflinger thread detects the track is stopped. 3682 if (!mResetDone) { 3683 TrackBase::reset(); 3684 // Force underrun condition to avoid false underrun callback until first data is 3685 // written to buffer 3686 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3687 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3688 mFillingUpStatus = FS_FILLING; 3689 mResetDone = true; 3690 } 3691} 3692 3693void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3694{ 3695 mMute = muted; 3696} 3697 3698status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3699{ 3700 status_t status = DEAD_OBJECT; 3701 sp<ThreadBase> thread = mThread.promote(); 3702 if (thread != 0) { 3703 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3704 status = playbackThread->attachAuxEffect(this, EffectId); 3705 } 3706 return status; 3707} 3708 3709void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3710{ 3711 mAuxEffectId = EffectId; 3712 mAuxBuffer = buffer; 3713} 3714 3715// ---------------------------------------------------------------------------- 3716 3717// RecordTrack constructor must be called with AudioFlinger::mLock held 3718AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3719 const wp<ThreadBase>& thread, 3720 const sp<Client>& client, 3721 uint32_t sampleRate, 3722 audio_format_t format, 3723 uint32_t channelMask, 3724 int frameCount, 3725 uint32_t flags, 3726 int sessionId) 3727 : TrackBase(thread, client, sampleRate, format, 3728 channelMask, frameCount, flags, 0, sessionId), 3729 mOverflow(false) 3730{ 3731 if (mCblk != NULL) { 3732 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3733 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3734 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3735 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3736 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3737 } else { 3738 mCblk->frameSize = sizeof(int8_t); 3739 } 3740 } 3741} 3742 3743AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3744{ 3745 sp<ThreadBase> thread = mThread.promote(); 3746 if (thread != 0) { 3747 AudioSystem::releaseInput(thread->id()); 3748 } 3749} 3750 3751status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3752{ 3753 audio_track_cblk_t* cblk = this->cblk(); 3754 uint32_t framesAvail; 3755 uint32_t framesReq = buffer->frameCount; 3756 3757 // Check if last stepServer failed, try to step now 3758 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3759 if (!step()) goto getNextBuffer_exit; 3760 ALOGV("stepServer recovered"); 3761 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3762 } 3763 3764 framesAvail = cblk->framesAvailable_l(); 3765 3766 if (CC_LIKELY(framesAvail)) { 3767 uint32_t s = cblk->server; 3768 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3769 3770 if (framesReq > framesAvail) { 3771 framesReq = framesAvail; 3772 } 3773 if (s + framesReq > bufferEnd) { 3774 framesReq = bufferEnd - s; 3775 } 3776 3777 buffer->raw = getBuffer(s, framesReq); 3778 if (buffer->raw == NULL) goto getNextBuffer_exit; 3779 3780 buffer->frameCount = framesReq; 3781 return NO_ERROR; 3782 } 3783 3784getNextBuffer_exit: 3785 buffer->raw = NULL; 3786 buffer->frameCount = 0; 3787 return NOT_ENOUGH_DATA; 3788} 3789 3790status_t AudioFlinger::RecordThread::RecordTrack::start() 3791{ 3792 sp<ThreadBase> thread = mThread.promote(); 3793 if (thread != 0) { 3794 RecordThread *recordThread = (RecordThread *)thread.get(); 3795 return recordThread->start(this); 3796 } else { 3797 return BAD_VALUE; 3798 } 3799} 3800 3801void AudioFlinger::RecordThread::RecordTrack::stop() 3802{ 3803 sp<ThreadBase> thread = mThread.promote(); 3804 if (thread != 0) { 3805 RecordThread *recordThread = (RecordThread *)thread.get(); 3806 recordThread->stop(this); 3807 TrackBase::reset(); 3808 // Force overerrun condition to avoid false overrun callback until first data is 3809 // read from buffer 3810 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3811 } 3812} 3813 3814void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3815{ 3816 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3817 (mClient == NULL) ? getpid() : mClient->pid(), 3818 mFormat, 3819 mChannelMask, 3820 mSessionId, 3821 mFrameCount, 3822 mState, 3823 mCblk->sampleRate, 3824 mCblk->server, 3825 mCblk->user); 3826} 3827 3828 3829// ---------------------------------------------------------------------------- 3830 3831AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3832 const wp<ThreadBase>& thread, 3833 DuplicatingThread *sourceThread, 3834 uint32_t sampleRate, 3835 audio_format_t format, 3836 uint32_t channelMask, 3837 int frameCount) 3838 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3839 mActive(false), mSourceThread(sourceThread) 3840{ 3841 3842 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3843 if (mCblk != NULL) { 3844 mCblk->flags |= CBLK_DIRECTION_OUT; 3845 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3846 mOutBuffer.frameCount = 0; 3847 playbackThread->mTracks.add(this); 3848 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3849 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3850 mCblk, mBuffer, mCblk->buffers, 3851 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3852 } else { 3853 ALOGW("Error creating output track on thread %p", playbackThread); 3854 } 3855} 3856 3857AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3858{ 3859 clearBufferQueue(); 3860} 3861 3862status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3863{ 3864 status_t status = Track::start(); 3865 if (status != NO_ERROR) { 3866 return status; 3867 } 3868 3869 mActive = true; 3870 mRetryCount = 127; 3871 return status; 3872} 3873 3874void AudioFlinger::PlaybackThread::OutputTrack::stop() 3875{ 3876 Track::stop(); 3877 clearBufferQueue(); 3878 mOutBuffer.frameCount = 0; 3879 mActive = false; 3880} 3881 3882bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3883{ 3884 Buffer *pInBuffer; 3885 Buffer inBuffer; 3886 uint32_t channelCount = mChannelCount; 3887 bool outputBufferFull = false; 3888 inBuffer.frameCount = frames; 3889 inBuffer.i16 = data; 3890 3891 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3892 3893 if (!mActive && frames != 0) { 3894 start(); 3895 sp<ThreadBase> thread = mThread.promote(); 3896 if (thread != 0) { 3897 MixerThread *mixerThread = (MixerThread *)thread.get(); 3898 if (mCblk->frameCount > frames){ 3899 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3900 uint32_t startFrames = (mCblk->frameCount - frames); 3901 pInBuffer = new Buffer; 3902 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3903 pInBuffer->frameCount = startFrames; 3904 pInBuffer->i16 = pInBuffer->mBuffer; 3905 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3906 mBufferQueue.add(pInBuffer); 3907 } else { 3908 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3909 } 3910 } 3911 } 3912 } 3913 3914 while (waitTimeLeftMs) { 3915 // First write pending buffers, then new data 3916 if (mBufferQueue.size()) { 3917 pInBuffer = mBufferQueue.itemAt(0); 3918 } else { 3919 pInBuffer = &inBuffer; 3920 } 3921 3922 if (pInBuffer->frameCount == 0) { 3923 break; 3924 } 3925 3926 if (mOutBuffer.frameCount == 0) { 3927 mOutBuffer.frameCount = pInBuffer->frameCount; 3928 nsecs_t startTime = systemTime(); 3929 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3930 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3931 outputBufferFull = true; 3932 break; 3933 } 3934 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3935 if (waitTimeLeftMs >= waitTimeMs) { 3936 waitTimeLeftMs -= waitTimeMs; 3937 } else { 3938 waitTimeLeftMs = 0; 3939 } 3940 } 3941 3942 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3943 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3944 mCblk->stepUser(outFrames); 3945 pInBuffer->frameCount -= outFrames; 3946 pInBuffer->i16 += outFrames * channelCount; 3947 mOutBuffer.frameCount -= outFrames; 3948 mOutBuffer.i16 += outFrames * channelCount; 3949 3950 if (pInBuffer->frameCount == 0) { 3951 if (mBufferQueue.size()) { 3952 mBufferQueue.removeAt(0); 3953 delete [] pInBuffer->mBuffer; 3954 delete pInBuffer; 3955 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3956 } else { 3957 break; 3958 } 3959 } 3960 } 3961 3962 // If we could not write all frames, allocate a buffer and queue it for next time. 3963 if (inBuffer.frameCount) { 3964 sp<ThreadBase> thread = mThread.promote(); 3965 if (thread != 0 && !thread->standby()) { 3966 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3967 pInBuffer = new Buffer; 3968 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3969 pInBuffer->frameCount = inBuffer.frameCount; 3970 pInBuffer->i16 = pInBuffer->mBuffer; 3971 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3972 mBufferQueue.add(pInBuffer); 3973 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3974 } else { 3975 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3976 } 3977 } 3978 } 3979 3980 // Calling write() with a 0 length buffer, means that no more data will be written: 3981 // If no more buffers are pending, fill output track buffer to make sure it is started 3982 // by output mixer. 3983 if (frames == 0 && mBufferQueue.size() == 0) { 3984 if (mCblk->user < mCblk->frameCount) { 3985 frames = mCblk->frameCount - mCblk->user; 3986 pInBuffer = new Buffer; 3987 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3988 pInBuffer->frameCount = frames; 3989 pInBuffer->i16 = pInBuffer->mBuffer; 3990 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3991 mBufferQueue.add(pInBuffer); 3992 } else if (mActive) { 3993 stop(); 3994 } 3995 } 3996 3997 return outputBufferFull; 3998} 3999 4000status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4001{ 4002 int active; 4003 status_t result; 4004 audio_track_cblk_t* cblk = mCblk; 4005 uint32_t framesReq = buffer->frameCount; 4006 4007// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4008 buffer->frameCount = 0; 4009 4010 uint32_t framesAvail = cblk->framesAvailable(); 4011 4012 4013 if (framesAvail == 0) { 4014 Mutex::Autolock _l(cblk->lock); 4015 goto start_loop_here; 4016 while (framesAvail == 0) { 4017 active = mActive; 4018 if (CC_UNLIKELY(!active)) { 4019 ALOGV("Not active and NO_MORE_BUFFERS"); 4020 return NO_MORE_BUFFERS; 4021 } 4022 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4023 if (result != NO_ERROR) { 4024 return NO_MORE_BUFFERS; 4025 } 4026 // read the server count again 4027 start_loop_here: 4028 framesAvail = cblk->framesAvailable_l(); 4029 } 4030 } 4031 4032// if (framesAvail < framesReq) { 4033// return NO_MORE_BUFFERS; 4034// } 4035 4036 if (framesReq > framesAvail) { 4037 framesReq = framesAvail; 4038 } 4039 4040 uint32_t u = cblk->user; 4041 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4042 4043 if (u + framesReq > bufferEnd) { 4044 framesReq = bufferEnd - u; 4045 } 4046 4047 buffer->frameCount = framesReq; 4048 buffer->raw = (void *)cblk->buffer(u); 4049 return NO_ERROR; 4050} 4051 4052 4053void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4054{ 4055 size_t size = mBufferQueue.size(); 4056 Buffer *pBuffer; 4057 4058 for (size_t i = 0; i < size; i++) { 4059 pBuffer = mBufferQueue.itemAt(i); 4060 delete [] pBuffer->mBuffer; 4061 delete pBuffer; 4062 } 4063 mBufferQueue.clear(); 4064} 4065 4066// ---------------------------------------------------------------------------- 4067 4068AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4069 : RefBase(), 4070 mAudioFlinger(audioFlinger), 4071 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4072 mPid(pid) 4073{ 4074 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4075} 4076 4077// Client destructor must be called with AudioFlinger::mLock held 4078AudioFlinger::Client::~Client() 4079{ 4080 mAudioFlinger->removeClient_l(mPid); 4081} 4082 4083sp<MemoryDealer> AudioFlinger::Client::heap() const 4084{ 4085 return mMemoryDealer; 4086} 4087 4088// ---------------------------------------------------------------------------- 4089 4090AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4091 const sp<IAudioFlingerClient>& client, 4092 pid_t pid) 4093 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4094{ 4095} 4096 4097AudioFlinger::NotificationClient::~NotificationClient() 4098{ 4099} 4100 4101void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4102{ 4103 sp<NotificationClient> keep(this); 4104 { 4105 mAudioFlinger->removeNotificationClient(mPid); 4106 } 4107} 4108 4109// ---------------------------------------------------------------------------- 4110 4111AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4112 : BnAudioTrack(), 4113 mTrack(track) 4114{ 4115} 4116 4117AudioFlinger::TrackHandle::~TrackHandle() { 4118 // just stop the track on deletion, associated resources 4119 // will be freed from the main thread once all pending buffers have 4120 // been played. Unless it's not in the active track list, in which 4121 // case we free everything now... 4122 mTrack->destroy(); 4123} 4124 4125sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4126 return mTrack->getCblk(); 4127} 4128 4129status_t AudioFlinger::TrackHandle::start() { 4130 return mTrack->start(); 4131} 4132 4133void AudioFlinger::TrackHandle::stop() { 4134 mTrack->stop(); 4135} 4136 4137void AudioFlinger::TrackHandle::flush() { 4138 mTrack->flush(); 4139} 4140 4141void AudioFlinger::TrackHandle::mute(bool e) { 4142 mTrack->mute(e); 4143} 4144 4145void AudioFlinger::TrackHandle::pause() { 4146 mTrack->pause(); 4147} 4148 4149status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4150{ 4151 return mTrack->attachAuxEffect(EffectId); 4152} 4153 4154status_t AudioFlinger::TrackHandle::onTransact( 4155 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4156{ 4157 return BnAudioTrack::onTransact(code, data, reply, flags); 4158} 4159 4160// ---------------------------------------------------------------------------- 4161 4162sp<IAudioRecord> AudioFlinger::openRecord( 4163 pid_t pid, 4164 int input, 4165 uint32_t sampleRate, 4166 audio_format_t format, 4167 uint32_t channelMask, 4168 int frameCount, 4169 uint32_t flags, 4170 int *sessionId, 4171 status_t *status) 4172{ 4173 sp<RecordThread::RecordTrack> recordTrack; 4174 sp<RecordHandle> recordHandle; 4175 sp<Client> client; 4176 wp<Client> wclient; 4177 status_t lStatus; 4178 RecordThread *thread; 4179 size_t inFrameCount; 4180 int lSessionId; 4181 4182 // check calling permissions 4183 if (!recordingAllowed()) { 4184 lStatus = PERMISSION_DENIED; 4185 goto Exit; 4186 } 4187 4188 // add client to list 4189 { // scope for mLock 4190 Mutex::Autolock _l(mLock); 4191 thread = checkRecordThread_l(input); 4192 if (thread == NULL) { 4193 lStatus = BAD_VALUE; 4194 goto Exit; 4195 } 4196 4197 wclient = mClients.valueFor(pid); 4198 if (wclient != NULL) { 4199 client = wclient.promote(); 4200 } else { 4201 client = new Client(this, pid); 4202 mClients.add(pid, client); 4203 } 4204 4205 // If no audio session id is provided, create one here 4206 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4207 lSessionId = *sessionId; 4208 } else { 4209 lSessionId = nextUniqueId(); 4210 if (sessionId != NULL) { 4211 *sessionId = lSessionId; 4212 } 4213 } 4214 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4215 recordTrack = thread->createRecordTrack_l(client, 4216 sampleRate, 4217 format, 4218 channelMask, 4219 frameCount, 4220 flags, 4221 lSessionId, 4222 &lStatus); 4223 } 4224 if (lStatus != NO_ERROR) { 4225 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4226 // destructor is called by the TrackBase destructor with mLock held 4227 client.clear(); 4228 recordTrack.clear(); 4229 goto Exit; 4230 } 4231 4232 // return to handle to client 4233 recordHandle = new RecordHandle(recordTrack); 4234 lStatus = NO_ERROR; 4235 4236Exit: 4237 if (status) { 4238 *status = lStatus; 4239 } 4240 return recordHandle; 4241} 4242 4243// ---------------------------------------------------------------------------- 4244 4245AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4246 : BnAudioRecord(), 4247 mRecordTrack(recordTrack) 4248{ 4249} 4250 4251AudioFlinger::RecordHandle::~RecordHandle() { 4252 stop(); 4253} 4254 4255sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4256 return mRecordTrack->getCblk(); 4257} 4258 4259status_t AudioFlinger::RecordHandle::start() { 4260 ALOGV("RecordHandle::start()"); 4261 return mRecordTrack->start(); 4262} 4263 4264void AudioFlinger::RecordHandle::stop() { 4265 ALOGV("RecordHandle::stop()"); 4266 mRecordTrack->stop(); 4267} 4268 4269status_t AudioFlinger::RecordHandle::onTransact( 4270 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4271{ 4272 return BnAudioRecord::onTransact(code, data, reply, flags); 4273} 4274 4275// ---------------------------------------------------------------------------- 4276 4277AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4278 AudioStreamIn *input, 4279 uint32_t sampleRate, 4280 uint32_t channels, 4281 int id, 4282 uint32_t device) : 4283 ThreadBase(audioFlinger, id, device, RECORD), 4284 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4285 // mRsmpInIndex and mInputBytes set by readInputParameters() 4286 mReqChannelCount(popcount(channels)), 4287 mReqSampleRate(sampleRate) 4288 // mBytesRead is only meaningful while active, and so is cleared in start() 4289 // (but might be better to also clear here for dump?) 4290{ 4291 snprintf(mName, kNameLength, "AudioIn_%d", id); 4292 4293 readInputParameters(); 4294} 4295 4296 4297AudioFlinger::RecordThread::~RecordThread() 4298{ 4299 delete[] mRsmpInBuffer; 4300 delete mResampler; 4301 delete[] mRsmpOutBuffer; 4302} 4303 4304void AudioFlinger::RecordThread::onFirstRef() 4305{ 4306 run(mName, PRIORITY_URGENT_AUDIO); 4307} 4308 4309status_t AudioFlinger::RecordThread::readyToRun() 4310{ 4311 status_t status = initCheck(); 4312 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4313 return status; 4314} 4315 4316bool AudioFlinger::RecordThread::threadLoop() 4317{ 4318 AudioBufferProvider::Buffer buffer; 4319 sp<RecordTrack> activeTrack; 4320 Vector< sp<EffectChain> > effectChains; 4321 4322 nsecs_t lastWarning = 0; 4323 4324 acquireWakeLock(); 4325 4326 // start recording 4327 while (!exitPending()) { 4328 4329 processConfigEvents(); 4330 4331 { // scope for mLock 4332 Mutex::Autolock _l(mLock); 4333 checkForNewParameters_l(); 4334 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4335 if (!mStandby) { 4336 mInput->stream->common.standby(&mInput->stream->common); 4337 mStandby = true; 4338 } 4339 4340 if (exitPending()) break; 4341 4342 releaseWakeLock_l(); 4343 ALOGV("RecordThread: loop stopping"); 4344 // go to sleep 4345 mWaitWorkCV.wait(mLock); 4346 ALOGV("RecordThread: loop starting"); 4347 acquireWakeLock_l(); 4348 continue; 4349 } 4350 if (mActiveTrack != 0) { 4351 if (mActiveTrack->mState == TrackBase::PAUSING) { 4352 if (!mStandby) { 4353 mInput->stream->common.standby(&mInput->stream->common); 4354 mStandby = true; 4355 } 4356 mActiveTrack.clear(); 4357 mStartStopCond.broadcast(); 4358 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4359 if (mReqChannelCount != mActiveTrack->channelCount()) { 4360 mActiveTrack.clear(); 4361 mStartStopCond.broadcast(); 4362 } else if (mBytesRead != 0) { 4363 // record start succeeds only if first read from audio input 4364 // succeeds 4365 if (mBytesRead > 0) { 4366 mActiveTrack->mState = TrackBase::ACTIVE; 4367 } else { 4368 mActiveTrack.clear(); 4369 } 4370 mStartStopCond.broadcast(); 4371 } 4372 mStandby = false; 4373 } 4374 } 4375 lockEffectChains_l(effectChains); 4376 } 4377 4378 if (mActiveTrack != 0) { 4379 if (mActiveTrack->mState != TrackBase::ACTIVE && 4380 mActiveTrack->mState != TrackBase::RESUMING) { 4381 unlockEffectChains(effectChains); 4382 usleep(kRecordThreadSleepUs); 4383 continue; 4384 } 4385 for (size_t i = 0; i < effectChains.size(); i ++) { 4386 effectChains[i]->process_l(); 4387 } 4388 4389 buffer.frameCount = mFrameCount; 4390 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4391 size_t framesOut = buffer.frameCount; 4392 if (mResampler == NULL) { 4393 // no resampling 4394 while (framesOut) { 4395 size_t framesIn = mFrameCount - mRsmpInIndex; 4396 if (framesIn) { 4397 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4398 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4399 if (framesIn > framesOut) 4400 framesIn = framesOut; 4401 mRsmpInIndex += framesIn; 4402 framesOut -= framesIn; 4403 if ((int)mChannelCount == mReqChannelCount || 4404 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4405 memcpy(dst, src, framesIn * mFrameSize); 4406 } else { 4407 int16_t *src16 = (int16_t *)src; 4408 int16_t *dst16 = (int16_t *)dst; 4409 if (mChannelCount == 1) { 4410 while (framesIn--) { 4411 *dst16++ = *src16; 4412 *dst16++ = *src16++; 4413 } 4414 } else { 4415 while (framesIn--) { 4416 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4417 src16 += 2; 4418 } 4419 } 4420 } 4421 } 4422 if (framesOut && mFrameCount == mRsmpInIndex) { 4423 if (framesOut == mFrameCount && 4424 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4425 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4426 framesOut = 0; 4427 } else { 4428 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4429 mRsmpInIndex = 0; 4430 } 4431 if (mBytesRead < 0) { 4432 ALOGE("Error reading audio input"); 4433 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4434 // Force input into standby so that it tries to 4435 // recover at next read attempt 4436 mInput->stream->common.standby(&mInput->stream->common); 4437 usleep(kRecordThreadSleepUs); 4438 } 4439 mRsmpInIndex = mFrameCount; 4440 framesOut = 0; 4441 buffer.frameCount = 0; 4442 } 4443 } 4444 } 4445 } else { 4446 // resampling 4447 4448 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4449 // alter output frame count as if we were expecting stereo samples 4450 if (mChannelCount == 1 && mReqChannelCount == 1) { 4451 framesOut >>= 1; 4452 } 4453 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4454 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4455 // are 32 bit aligned which should be always true. 4456 if (mChannelCount == 2 && mReqChannelCount == 1) { 4457 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4458 // the resampler always outputs stereo samples: do post stereo to mono conversion 4459 int16_t *src = (int16_t *)mRsmpOutBuffer; 4460 int16_t *dst = buffer.i16; 4461 while (framesOut--) { 4462 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4463 src += 2; 4464 } 4465 } else { 4466 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4467 } 4468 4469 } 4470 mActiveTrack->releaseBuffer(&buffer); 4471 mActiveTrack->overflow(); 4472 } 4473 // client isn't retrieving buffers fast enough 4474 else { 4475 if (!mActiveTrack->setOverflow()) { 4476 nsecs_t now = systemTime(); 4477 if ((now - lastWarning) > kWarningThrottleNs) { 4478 ALOGW("RecordThread: buffer overflow"); 4479 lastWarning = now; 4480 } 4481 } 4482 // Release the processor for a while before asking for a new buffer. 4483 // This will give the application more chance to read from the buffer and 4484 // clear the overflow. 4485 usleep(kRecordThreadSleepUs); 4486 } 4487 } 4488 // enable changes in effect chain 4489 unlockEffectChains(effectChains); 4490 effectChains.clear(); 4491 } 4492 4493 if (!mStandby) { 4494 mInput->stream->common.standby(&mInput->stream->common); 4495 } 4496 mActiveTrack.clear(); 4497 4498 mStartStopCond.broadcast(); 4499 4500 releaseWakeLock(); 4501 4502 ALOGV("RecordThread %p exiting", this); 4503 return false; 4504} 4505 4506 4507sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4508 const sp<AudioFlinger::Client>& client, 4509 uint32_t sampleRate, 4510 audio_format_t format, 4511 int channelMask, 4512 int frameCount, 4513 uint32_t flags, 4514 int sessionId, 4515 status_t *status) 4516{ 4517 sp<RecordTrack> track; 4518 status_t lStatus; 4519 4520 lStatus = initCheck(); 4521 if (lStatus != NO_ERROR) { 4522 ALOGE("Audio driver not initialized."); 4523 goto Exit; 4524 } 4525 4526 { // scope for mLock 4527 Mutex::Autolock _l(mLock); 4528 4529 track = new RecordTrack(this, client, sampleRate, 4530 format, channelMask, frameCount, flags, sessionId); 4531 4532 if (track->getCblk() == NULL) { 4533 lStatus = NO_MEMORY; 4534 goto Exit; 4535 } 4536 4537 mTrack = track.get(); 4538 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4539 bool suspend = audio_is_bluetooth_sco_device( 4540 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4541 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4542 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4543 } 4544 lStatus = NO_ERROR; 4545 4546Exit: 4547 if (status) { 4548 *status = lStatus; 4549 } 4550 return track; 4551} 4552 4553status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4554{ 4555 ALOGV("RecordThread::start"); 4556 sp <ThreadBase> strongMe = this; 4557 status_t status = NO_ERROR; 4558 { 4559 AutoMutex lock(mLock); 4560 if (mActiveTrack != 0) { 4561 if (recordTrack != mActiveTrack.get()) { 4562 status = -EBUSY; 4563 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4564 mActiveTrack->mState = TrackBase::ACTIVE; 4565 } 4566 return status; 4567 } 4568 4569 recordTrack->mState = TrackBase::IDLE; 4570 mActiveTrack = recordTrack; 4571 mLock.unlock(); 4572 status_t status = AudioSystem::startInput(mId); 4573 mLock.lock(); 4574 if (status != NO_ERROR) { 4575 mActiveTrack.clear(); 4576 return status; 4577 } 4578 mRsmpInIndex = mFrameCount; 4579 mBytesRead = 0; 4580 if (mResampler != NULL) { 4581 mResampler->reset(); 4582 } 4583 mActiveTrack->mState = TrackBase::RESUMING; 4584 // signal thread to start 4585 ALOGV("Signal record thread"); 4586 mWaitWorkCV.signal(); 4587 // do not wait for mStartStopCond if exiting 4588 if (mExiting) { 4589 mActiveTrack.clear(); 4590 status = INVALID_OPERATION; 4591 goto startError; 4592 } 4593 mStartStopCond.wait(mLock); 4594 if (mActiveTrack == 0) { 4595 ALOGV("Record failed to start"); 4596 status = BAD_VALUE; 4597 goto startError; 4598 } 4599 ALOGV("Record started OK"); 4600 return status; 4601 } 4602startError: 4603 AudioSystem::stopInput(mId); 4604 return status; 4605} 4606 4607void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4608 ALOGV("RecordThread::stop"); 4609 sp <ThreadBase> strongMe = this; 4610 { 4611 AutoMutex lock(mLock); 4612 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4613 mActiveTrack->mState = TrackBase::PAUSING; 4614 // do not wait for mStartStopCond if exiting 4615 if (mExiting) { 4616 return; 4617 } 4618 mStartStopCond.wait(mLock); 4619 // if we have been restarted, recordTrack == mActiveTrack.get() here 4620 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4621 mLock.unlock(); 4622 AudioSystem::stopInput(mId); 4623 mLock.lock(); 4624 ALOGV("Record stopped OK"); 4625 } 4626 } 4627 } 4628} 4629 4630status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4631{ 4632 const size_t SIZE = 256; 4633 char buffer[SIZE]; 4634 String8 result; 4635 pid_t pid = 0; 4636 4637 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4638 result.append(buffer); 4639 4640 if (mActiveTrack != 0) { 4641 result.append("Active Track:\n"); 4642 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4643 mActiveTrack->dump(buffer, SIZE); 4644 result.append(buffer); 4645 4646 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4647 result.append(buffer); 4648 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4649 result.append(buffer); 4650 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4651 result.append(buffer); 4652 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4653 result.append(buffer); 4654 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4655 result.append(buffer); 4656 4657 4658 } else { 4659 result.append("No record client\n"); 4660 } 4661 write(fd, result.string(), result.size()); 4662 4663 dumpBase(fd, args); 4664 dumpEffectChains(fd, args); 4665 4666 return NO_ERROR; 4667} 4668 4669status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4670{ 4671 size_t framesReq = buffer->frameCount; 4672 size_t framesReady = mFrameCount - mRsmpInIndex; 4673 int channelCount; 4674 4675 if (framesReady == 0) { 4676 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4677 if (mBytesRead < 0) { 4678 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4679 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4680 // Force input into standby so that it tries to 4681 // recover at next read attempt 4682 mInput->stream->common.standby(&mInput->stream->common); 4683 usleep(kRecordThreadSleepUs); 4684 } 4685 buffer->raw = NULL; 4686 buffer->frameCount = 0; 4687 return NOT_ENOUGH_DATA; 4688 } 4689 mRsmpInIndex = 0; 4690 framesReady = mFrameCount; 4691 } 4692 4693 if (framesReq > framesReady) { 4694 framesReq = framesReady; 4695 } 4696 4697 if (mChannelCount == 1 && mReqChannelCount == 2) { 4698 channelCount = 1; 4699 } else { 4700 channelCount = 2; 4701 } 4702 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4703 buffer->frameCount = framesReq; 4704 return NO_ERROR; 4705} 4706 4707void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4708{ 4709 mRsmpInIndex += buffer->frameCount; 4710 buffer->frameCount = 0; 4711} 4712 4713bool AudioFlinger::RecordThread::checkForNewParameters_l() 4714{ 4715 bool reconfig = false; 4716 4717 while (!mNewParameters.isEmpty()) { 4718 status_t status = NO_ERROR; 4719 String8 keyValuePair = mNewParameters[0]; 4720 AudioParameter param = AudioParameter(keyValuePair); 4721 int value; 4722 audio_format_t reqFormat = mFormat; 4723 int reqSamplingRate = mReqSampleRate; 4724 int reqChannelCount = mReqChannelCount; 4725 4726 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4727 reqSamplingRate = value; 4728 reconfig = true; 4729 } 4730 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4731 reqFormat = (audio_format_t) value; 4732 reconfig = true; 4733 } 4734 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4735 reqChannelCount = popcount(value); 4736 reconfig = true; 4737 } 4738 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4739 // do not accept frame count changes if tracks are open as the track buffer 4740 // size depends on frame count and correct behavior would not be garantied 4741 // if frame count is changed after track creation 4742 if (mActiveTrack != 0) { 4743 status = INVALID_OPERATION; 4744 } else { 4745 reconfig = true; 4746 } 4747 } 4748 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4749 // forward device change to effects that have requested to be 4750 // aware of attached audio device. 4751 for (size_t i = 0; i < mEffectChains.size(); i++) { 4752 mEffectChains[i]->setDevice_l(value); 4753 } 4754 // store input device and output device but do not forward output device to audio HAL. 4755 // Note that status is ignored by the caller for output device 4756 // (see AudioFlinger::setParameters() 4757 if (value & AUDIO_DEVICE_OUT_ALL) { 4758 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4759 status = BAD_VALUE; 4760 } else { 4761 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4762 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4763 if (mTrack != NULL) { 4764 bool suspend = audio_is_bluetooth_sco_device( 4765 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4766 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4767 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4768 } 4769 } 4770 mDevice |= (uint32_t)value; 4771 } 4772 if (status == NO_ERROR) { 4773 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4774 if (status == INVALID_OPERATION) { 4775 mInput->stream->common.standby(&mInput->stream->common); 4776 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4777 } 4778 if (reconfig) { 4779 if (status == BAD_VALUE && 4780 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4781 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4782 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4783 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4784 (reqChannelCount < 3)) { 4785 status = NO_ERROR; 4786 } 4787 if (status == NO_ERROR) { 4788 readInputParameters(); 4789 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4790 } 4791 } 4792 } 4793 4794 mNewParameters.removeAt(0); 4795 4796 mParamStatus = status; 4797 mParamCond.signal(); 4798 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4799 // already timed out waiting for the status and will never signal the condition. 4800 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4801 } 4802 return reconfig; 4803} 4804 4805String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4806{ 4807 char *s; 4808 String8 out_s8 = String8(); 4809 4810 Mutex::Autolock _l(mLock); 4811 if (initCheck() != NO_ERROR) { 4812 return out_s8; 4813 } 4814 4815 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4816 out_s8 = String8(s); 4817 free(s); 4818 return out_s8; 4819} 4820 4821void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4822 AudioSystem::OutputDescriptor desc; 4823 void *param2 = NULL; 4824 4825 switch (event) { 4826 case AudioSystem::INPUT_OPENED: 4827 case AudioSystem::INPUT_CONFIG_CHANGED: 4828 desc.channels = mChannelMask; 4829 desc.samplingRate = mSampleRate; 4830 desc.format = mFormat; 4831 desc.frameCount = mFrameCount; 4832 desc.latency = 0; 4833 param2 = &desc; 4834 break; 4835 4836 case AudioSystem::INPUT_CLOSED: 4837 default: 4838 break; 4839 } 4840 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4841} 4842 4843void AudioFlinger::RecordThread::readInputParameters() 4844{ 4845 delete mRsmpInBuffer; 4846 // mRsmpInBuffer is always assigned a new[] below 4847 delete mRsmpOutBuffer; 4848 mRsmpOutBuffer = NULL; 4849 delete mResampler; 4850 mResampler = NULL; 4851 4852 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4853 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4854 mChannelCount = (uint16_t)popcount(mChannelMask); 4855 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4856 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4857 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4858 mFrameCount = mInputBytes / mFrameSize; 4859 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4860 4861 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4862 { 4863 int channelCount; 4864 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4865 // stereo to mono post process as the resampler always outputs stereo. 4866 if (mChannelCount == 1 && mReqChannelCount == 2) { 4867 channelCount = 1; 4868 } else { 4869 channelCount = 2; 4870 } 4871 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4872 mResampler->setSampleRate(mSampleRate); 4873 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4874 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4875 4876 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4877 if (mChannelCount == 1 && mReqChannelCount == 1) { 4878 mFrameCount >>= 1; 4879 } 4880 4881 } 4882 mRsmpInIndex = mFrameCount; 4883} 4884 4885unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4886{ 4887 Mutex::Autolock _l(mLock); 4888 if (initCheck() != NO_ERROR) { 4889 return 0; 4890 } 4891 4892 return mInput->stream->get_input_frames_lost(mInput->stream); 4893} 4894 4895uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4896{ 4897 Mutex::Autolock _l(mLock); 4898 uint32_t result = 0; 4899 if (getEffectChain_l(sessionId) != 0) { 4900 result = EFFECT_SESSION; 4901 } 4902 4903 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4904 result |= TRACK_SESSION; 4905 } 4906 4907 return result; 4908} 4909 4910AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4911{ 4912 Mutex::Autolock _l(mLock); 4913 return mTrack; 4914} 4915 4916AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4917{ 4918 Mutex::Autolock _l(mLock); 4919 return mInput; 4920} 4921 4922AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4923{ 4924 Mutex::Autolock _l(mLock); 4925 AudioStreamIn *input = mInput; 4926 mInput = NULL; 4927 return input; 4928} 4929 4930// this method must always be called either with ThreadBase mLock held or inside the thread loop 4931audio_stream_t* AudioFlinger::RecordThread::stream() 4932{ 4933 if (mInput == NULL) { 4934 return NULL; 4935 } 4936 return &mInput->stream->common; 4937} 4938 4939 4940// ---------------------------------------------------------------------------- 4941 4942int AudioFlinger::openOutput(uint32_t *pDevices, 4943 uint32_t *pSamplingRate, 4944 audio_format_t *pFormat, 4945 uint32_t *pChannels, 4946 uint32_t *pLatencyMs, 4947 uint32_t flags) 4948{ 4949 status_t status; 4950 PlaybackThread *thread = NULL; 4951 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4952 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4953 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4954 uint32_t channels = pChannels ? *pChannels : 0; 4955 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4956 audio_stream_out_t *outStream; 4957 audio_hw_device_t *outHwDev; 4958 4959 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4960 pDevices ? *pDevices : 0, 4961 samplingRate, 4962 format, 4963 channels, 4964 flags); 4965 4966 if (pDevices == NULL || *pDevices == 0) { 4967 return 0; 4968 } 4969 4970 Mutex::Autolock _l(mLock); 4971 4972 outHwDev = findSuitableHwDev_l(*pDevices); 4973 if (outHwDev == NULL) 4974 return 0; 4975 4976 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4977 &channels, &samplingRate, &outStream); 4978 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4979 outStream, 4980 samplingRate, 4981 format, 4982 channels, 4983 status); 4984 4985 mHardwareStatus = AUDIO_HW_IDLE; 4986 if (outStream != NULL) { 4987 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4988 int id = nextUniqueId(); 4989 4990 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4991 (format != AUDIO_FORMAT_PCM_16_BIT) || 4992 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4993 thread = new DirectOutputThread(this, output, id, *pDevices); 4994 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4995 } else { 4996 thread = new MixerThread(this, output, id, *pDevices); 4997 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4998 } 4999 mPlaybackThreads.add(id, thread); 5000 5001 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5002 if (pFormat != NULL) *pFormat = format; 5003 if (pChannels != NULL) *pChannels = channels; 5004 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5005 5006 // notify client processes of the new output creation 5007 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5008 return id; 5009 } 5010 5011 return 0; 5012} 5013 5014int AudioFlinger::openDuplicateOutput(int output1, int output2) 5015{ 5016 Mutex::Autolock _l(mLock); 5017 MixerThread *thread1 = checkMixerThread_l(output1); 5018 MixerThread *thread2 = checkMixerThread_l(output2); 5019 5020 if (thread1 == NULL || thread2 == NULL) { 5021 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5022 return 0; 5023 } 5024 5025 int id = nextUniqueId(); 5026 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5027 thread->addOutputTrack(thread2); 5028 mPlaybackThreads.add(id, thread); 5029 // notify client processes of the new output creation 5030 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5031 return id; 5032} 5033 5034status_t AudioFlinger::closeOutput(int output) 5035{ 5036 // keep strong reference on the playback thread so that 5037 // it is not destroyed while exit() is executed 5038 sp <PlaybackThread> thread; 5039 { 5040 Mutex::Autolock _l(mLock); 5041 thread = checkPlaybackThread_l(output); 5042 if (thread == NULL) { 5043 return BAD_VALUE; 5044 } 5045 5046 ALOGV("closeOutput() %d", output); 5047 5048 if (thread->type() == ThreadBase::MIXER) { 5049 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5050 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5051 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5052 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5053 } 5054 } 5055 } 5056 void *param2 = NULL; 5057 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5058 mPlaybackThreads.removeItem(output); 5059 } 5060 thread->exit(); 5061 5062 if (thread->type() != ThreadBase::DUPLICATING) { 5063 AudioStreamOut *out = thread->clearOutput(); 5064 assert(out != NULL); 5065 // from now on thread->mOutput is NULL 5066 out->hwDev->close_output_stream(out->hwDev, out->stream); 5067 delete out; 5068 } 5069 return NO_ERROR; 5070} 5071 5072status_t AudioFlinger::suspendOutput(int output) 5073{ 5074 Mutex::Autolock _l(mLock); 5075 PlaybackThread *thread = checkPlaybackThread_l(output); 5076 5077 if (thread == NULL) { 5078 return BAD_VALUE; 5079 } 5080 5081 ALOGV("suspendOutput() %d", output); 5082 thread->suspend(); 5083 5084 return NO_ERROR; 5085} 5086 5087status_t AudioFlinger::restoreOutput(int output) 5088{ 5089 Mutex::Autolock _l(mLock); 5090 PlaybackThread *thread = checkPlaybackThread_l(output); 5091 5092 if (thread == NULL) { 5093 return BAD_VALUE; 5094 } 5095 5096 ALOGV("restoreOutput() %d", output); 5097 5098 thread->restore(); 5099 5100 return NO_ERROR; 5101} 5102 5103int AudioFlinger::openInput(uint32_t *pDevices, 5104 uint32_t *pSamplingRate, 5105 audio_format_t *pFormat, 5106 uint32_t *pChannels, 5107 audio_in_acoustics_t acoustics) 5108{ 5109 status_t status; 5110 RecordThread *thread = NULL; 5111 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5112 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5113 uint32_t channels = pChannels ? *pChannels : 0; 5114 uint32_t reqSamplingRate = samplingRate; 5115 audio_format_t reqFormat = format; 5116 uint32_t reqChannels = channels; 5117 audio_stream_in_t *inStream; 5118 audio_hw_device_t *inHwDev; 5119 5120 if (pDevices == NULL || *pDevices == 0) { 5121 return 0; 5122 } 5123 5124 Mutex::Autolock _l(mLock); 5125 5126 inHwDev = findSuitableHwDev_l(*pDevices); 5127 if (inHwDev == NULL) 5128 return 0; 5129 5130 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5131 &channels, &samplingRate, 5132 acoustics, 5133 &inStream); 5134 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5135 inStream, 5136 samplingRate, 5137 format, 5138 channels, 5139 acoustics, 5140 status); 5141 5142 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5143 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5144 // or stereo to mono conversions on 16 bit PCM inputs. 5145 if (inStream == NULL && status == BAD_VALUE && 5146 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5147 (samplingRate <= 2 * reqSamplingRate) && 5148 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5149 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5150 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5151 &channels, &samplingRate, 5152 acoustics, 5153 &inStream); 5154 } 5155 5156 if (inStream != NULL) { 5157 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5158 5159 int id = nextUniqueId(); 5160 // Start record thread 5161 // RecorThread require both input and output device indication to forward to audio 5162 // pre processing modules 5163 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5164 thread = new RecordThread(this, 5165 input, 5166 reqSamplingRate, 5167 reqChannels, 5168 id, 5169 device); 5170 mRecordThreads.add(id, thread); 5171 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5172 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5173 if (pFormat != NULL) *pFormat = format; 5174 if (pChannels != NULL) *pChannels = reqChannels; 5175 5176 input->stream->common.standby(&input->stream->common); 5177 5178 // notify client processes of the new input creation 5179 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5180 return id; 5181 } 5182 5183 return 0; 5184} 5185 5186status_t AudioFlinger::closeInput(int input) 5187{ 5188 // keep strong reference on the record thread so that 5189 // it is not destroyed while exit() is executed 5190 sp <RecordThread> thread; 5191 { 5192 Mutex::Autolock _l(mLock); 5193 thread = checkRecordThread_l(input); 5194 if (thread == NULL) { 5195 return BAD_VALUE; 5196 } 5197 5198 ALOGV("closeInput() %d", input); 5199 void *param2 = NULL; 5200 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5201 mRecordThreads.removeItem(input); 5202 } 5203 thread->exit(); 5204 5205 AudioStreamIn *in = thread->clearInput(); 5206 assert(in != NULL); 5207 // from now on thread->mInput is NULL 5208 in->hwDev->close_input_stream(in->hwDev, in->stream); 5209 delete in; 5210 5211 return NO_ERROR; 5212} 5213 5214status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5215{ 5216 Mutex::Autolock _l(mLock); 5217 MixerThread *dstThread = checkMixerThread_l(output); 5218 if (dstThread == NULL) { 5219 ALOGW("setStreamOutput() bad output id %d", output); 5220 return BAD_VALUE; 5221 } 5222 5223 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5224 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5225 5226 dstThread->setStreamValid(stream, true); 5227 5228 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5229 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5230 if (thread != dstThread && 5231 thread->type() != ThreadBase::DIRECT) { 5232 MixerThread *srcThread = (MixerThread *)thread; 5233 srcThread->setStreamValid(stream, false); 5234 srcThread->invalidateTracks(stream); 5235 } 5236 } 5237 5238 return NO_ERROR; 5239} 5240 5241 5242int AudioFlinger::newAudioSessionId() 5243{ 5244 return nextUniqueId(); 5245} 5246 5247void AudioFlinger::acquireAudioSessionId(int audioSession) 5248{ 5249 Mutex::Autolock _l(mLock); 5250 int caller = IPCThreadState::self()->getCallingPid(); 5251 ALOGV("acquiring %d from %d", audioSession, caller); 5252 int num = mAudioSessionRefs.size(); 5253 for (int i = 0; i< num; i++) { 5254 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5255 if (ref->sessionid == audioSession && ref->pid == caller) { 5256 ref->cnt++; 5257 ALOGV(" incremented refcount to %d", ref->cnt); 5258 return; 5259 } 5260 } 5261 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5262 ALOGV(" added new entry for %d", audioSession); 5263} 5264 5265void AudioFlinger::releaseAudioSessionId(int audioSession) 5266{ 5267 Mutex::Autolock _l(mLock); 5268 int caller = IPCThreadState::self()->getCallingPid(); 5269 ALOGV("releasing %d from %d", audioSession, caller); 5270 int num = mAudioSessionRefs.size(); 5271 for (int i = 0; i< num; i++) { 5272 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5273 if (ref->sessionid == audioSession && ref->pid == caller) { 5274 ref->cnt--; 5275 ALOGV(" decremented refcount to %d", ref->cnt); 5276 if (ref->cnt == 0) { 5277 mAudioSessionRefs.removeAt(i); 5278 delete ref; 5279 purgeStaleEffects_l(); 5280 } 5281 return; 5282 } 5283 } 5284 ALOGW("session id %d not found for pid %d", audioSession, caller); 5285} 5286 5287void AudioFlinger::purgeStaleEffects_l() { 5288 5289 ALOGV("purging stale effects"); 5290 5291 Vector< sp<EffectChain> > chains; 5292 5293 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5294 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5295 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5296 sp<EffectChain> ec = t->mEffectChains[j]; 5297 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5298 chains.push(ec); 5299 } 5300 } 5301 } 5302 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5303 sp<RecordThread> t = mRecordThreads.valueAt(i); 5304 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5305 sp<EffectChain> ec = t->mEffectChains[j]; 5306 chains.push(ec); 5307 } 5308 } 5309 5310 for (size_t i = 0; i < chains.size(); i++) { 5311 sp<EffectChain> ec = chains[i]; 5312 int sessionid = ec->sessionId(); 5313 sp<ThreadBase> t = ec->mThread.promote(); 5314 if (t == 0) { 5315 continue; 5316 } 5317 size_t numsessionrefs = mAudioSessionRefs.size(); 5318 bool found = false; 5319 for (size_t k = 0; k < numsessionrefs; k++) { 5320 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5321 if (ref->sessionid == sessionid) { 5322 ALOGV(" session %d still exists for %d with %d refs", 5323 sessionid, ref->pid, ref->cnt); 5324 found = true; 5325 break; 5326 } 5327 } 5328 if (!found) { 5329 // remove all effects from the chain 5330 while (ec->mEffects.size()) { 5331 sp<EffectModule> effect = ec->mEffects[0]; 5332 effect->unPin(); 5333 Mutex::Autolock _l (t->mLock); 5334 t->removeEffect_l(effect); 5335 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5336 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5337 if (handle != 0) { 5338 handle->mEffect.clear(); 5339 if (handle->mHasControl && handle->mEnabled) { 5340 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5341 } 5342 } 5343 } 5344 AudioSystem::unregisterEffect(effect->id()); 5345 } 5346 } 5347 } 5348 return; 5349} 5350 5351// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5352AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5353{ 5354 PlaybackThread *thread = NULL; 5355 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5356 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5357 } 5358 return thread; 5359} 5360 5361// checkMixerThread_l() must be called with AudioFlinger::mLock held 5362AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5363{ 5364 PlaybackThread *thread = checkPlaybackThread_l(output); 5365 if (thread != NULL) { 5366 if (thread->type() == ThreadBase::DIRECT) { 5367 thread = NULL; 5368 } 5369 } 5370 return (MixerThread *)thread; 5371} 5372 5373// checkRecordThread_l() must be called with AudioFlinger::mLock held 5374AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5375{ 5376 RecordThread *thread = NULL; 5377 if (mRecordThreads.indexOfKey(input) >= 0) { 5378 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5379 } 5380 return thread; 5381} 5382 5383uint32_t AudioFlinger::nextUniqueId() 5384{ 5385 return android_atomic_inc(&mNextUniqueId); 5386} 5387 5388AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5389{ 5390 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5391 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5392 AudioStreamOut *output = thread->getOutput(); 5393 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5394 return thread; 5395 } 5396 } 5397 return NULL; 5398} 5399 5400uint32_t AudioFlinger::primaryOutputDevice_l() 5401{ 5402 PlaybackThread *thread = primaryPlaybackThread_l(); 5403 5404 if (thread == NULL) { 5405 return 0; 5406 } 5407 5408 return thread->device(); 5409} 5410 5411 5412// ---------------------------------------------------------------------------- 5413// Effect management 5414// ---------------------------------------------------------------------------- 5415 5416 5417status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5418{ 5419 Mutex::Autolock _l(mLock); 5420 return EffectQueryNumberEffects(numEffects); 5421} 5422 5423status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5424{ 5425 Mutex::Autolock _l(mLock); 5426 return EffectQueryEffect(index, descriptor); 5427} 5428 5429status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5430{ 5431 Mutex::Autolock _l(mLock); 5432 return EffectGetDescriptor(pUuid, descriptor); 5433} 5434 5435 5436sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5437 effect_descriptor_t *pDesc, 5438 const sp<IEffectClient>& effectClient, 5439 int32_t priority, 5440 int io, 5441 int sessionId, 5442 status_t *status, 5443 int *id, 5444 int *enabled) 5445{ 5446 status_t lStatus = NO_ERROR; 5447 sp<EffectHandle> handle; 5448 effect_descriptor_t desc; 5449 sp<Client> client; 5450 wp<Client> wclient; 5451 5452 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5453 pid, effectClient.get(), priority, sessionId, io); 5454 5455 if (pDesc == NULL) { 5456 lStatus = BAD_VALUE; 5457 goto Exit; 5458 } 5459 5460 // check audio settings permission for global effects 5461 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5462 lStatus = PERMISSION_DENIED; 5463 goto Exit; 5464 } 5465 5466 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5467 // that can only be created by audio policy manager (running in same process) 5468 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5469 lStatus = PERMISSION_DENIED; 5470 goto Exit; 5471 } 5472 5473 if (io == 0) { 5474 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5475 // output must be specified by AudioPolicyManager when using session 5476 // AUDIO_SESSION_OUTPUT_STAGE 5477 lStatus = BAD_VALUE; 5478 goto Exit; 5479 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5480 // if the output returned by getOutputForEffect() is removed before we lock the 5481 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5482 // and we will exit safely 5483 io = AudioSystem::getOutputForEffect(&desc); 5484 } 5485 } 5486 5487 { 5488 Mutex::Autolock _l(mLock); 5489 5490 5491 if (!EffectIsNullUuid(&pDesc->uuid)) { 5492 // if uuid is specified, request effect descriptor 5493 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5494 if (lStatus < 0) { 5495 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5496 goto Exit; 5497 } 5498 } else { 5499 // if uuid is not specified, look for an available implementation 5500 // of the required type in effect factory 5501 if (EffectIsNullUuid(&pDesc->type)) { 5502 ALOGW("createEffect() no effect type"); 5503 lStatus = BAD_VALUE; 5504 goto Exit; 5505 } 5506 uint32_t numEffects = 0; 5507 effect_descriptor_t d; 5508 d.flags = 0; // prevent compiler warning 5509 bool found = false; 5510 5511 lStatus = EffectQueryNumberEffects(&numEffects); 5512 if (lStatus < 0) { 5513 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5514 goto Exit; 5515 } 5516 for (uint32_t i = 0; i < numEffects; i++) { 5517 lStatus = EffectQueryEffect(i, &desc); 5518 if (lStatus < 0) { 5519 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5520 continue; 5521 } 5522 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5523 // If matching type found save effect descriptor. If the session is 5524 // 0 and the effect is not auxiliary, continue enumeration in case 5525 // an auxiliary version of this effect type is available 5526 found = true; 5527 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5528 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5529 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5530 break; 5531 } 5532 } 5533 } 5534 if (!found) { 5535 lStatus = BAD_VALUE; 5536 ALOGW("createEffect() effect not found"); 5537 goto Exit; 5538 } 5539 // For same effect type, chose auxiliary version over insert version if 5540 // connect to output mix (Compliance to OpenSL ES) 5541 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5542 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5543 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5544 } 5545 } 5546 5547 // Do not allow auxiliary effects on a session different from 0 (output mix) 5548 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5549 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5550 lStatus = INVALID_OPERATION; 5551 goto Exit; 5552 } 5553 5554 // check recording permission for visualizer 5555 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5556 !recordingAllowed()) { 5557 lStatus = PERMISSION_DENIED; 5558 goto Exit; 5559 } 5560 5561 // return effect descriptor 5562 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5563 5564 // If output is not specified try to find a matching audio session ID in one of the 5565 // output threads. 5566 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5567 // because of code checking output when entering the function. 5568 // Note: io is never 0 when creating an effect on an input 5569 if (io == 0) { 5570 // look for the thread where the specified audio session is present 5571 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5572 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5573 io = mPlaybackThreads.keyAt(i); 5574 break; 5575 } 5576 } 5577 if (io == 0) { 5578 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5579 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5580 io = mRecordThreads.keyAt(i); 5581 break; 5582 } 5583 } 5584 } 5585 // If no output thread contains the requested session ID, default to 5586 // first output. The effect chain will be moved to the correct output 5587 // thread when a track with the same session ID is created 5588 if (io == 0 && mPlaybackThreads.size()) { 5589 io = mPlaybackThreads.keyAt(0); 5590 } 5591 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5592 } 5593 ThreadBase *thread = checkRecordThread_l(io); 5594 if (thread == NULL) { 5595 thread = checkPlaybackThread_l(io); 5596 if (thread == NULL) { 5597 ALOGE("createEffect() unknown output thread"); 5598 lStatus = BAD_VALUE; 5599 goto Exit; 5600 } 5601 } 5602 5603 wclient = mClients.valueFor(pid); 5604 5605 if (wclient != NULL) { 5606 client = wclient.promote(); 5607 } else { 5608 client = new Client(this, pid); 5609 mClients.add(pid, client); 5610 } 5611 5612 // create effect on selected output thread 5613 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5614 &desc, enabled, &lStatus); 5615 if (handle != 0 && id != NULL) { 5616 *id = handle->id(); 5617 } 5618 } 5619 5620Exit: 5621 if(status) { 5622 *status = lStatus; 5623 } 5624 return handle; 5625} 5626 5627status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5628{ 5629 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5630 sessionId, srcOutput, dstOutput); 5631 Mutex::Autolock _l(mLock); 5632 if (srcOutput == dstOutput) { 5633 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5634 return NO_ERROR; 5635 } 5636 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5637 if (srcThread == NULL) { 5638 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5639 return BAD_VALUE; 5640 } 5641 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5642 if (dstThread == NULL) { 5643 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5644 return BAD_VALUE; 5645 } 5646 5647 Mutex::Autolock _dl(dstThread->mLock); 5648 Mutex::Autolock _sl(srcThread->mLock); 5649 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5650 5651 return NO_ERROR; 5652} 5653 5654// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5655status_t AudioFlinger::moveEffectChain_l(int sessionId, 5656 AudioFlinger::PlaybackThread *srcThread, 5657 AudioFlinger::PlaybackThread *dstThread, 5658 bool reRegister) 5659{ 5660 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5661 sessionId, srcThread, dstThread); 5662 5663 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5664 if (chain == 0) { 5665 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5666 sessionId, srcThread); 5667 return INVALID_OPERATION; 5668 } 5669 5670 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5671 // so that a new chain is created with correct parameters when first effect is added. This is 5672 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5673 // removed. 5674 srcThread->removeEffectChain_l(chain); 5675 5676 // transfer all effects one by one so that new effect chain is created on new thread with 5677 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5678 int dstOutput = dstThread->id(); 5679 sp<EffectChain> dstChain; 5680 uint32_t strategy = 0; // prevent compiler warning 5681 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5682 while (effect != 0) { 5683 srcThread->removeEffect_l(effect); 5684 dstThread->addEffect_l(effect); 5685 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5686 if (effect->state() == EffectModule::ACTIVE || 5687 effect->state() == EffectModule::STOPPING) { 5688 effect->start(); 5689 } 5690 // if the move request is not received from audio policy manager, the effect must be 5691 // re-registered with the new strategy and output 5692 if (dstChain == 0) { 5693 dstChain = effect->chain().promote(); 5694 if (dstChain == 0) { 5695 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5696 srcThread->addEffect_l(effect); 5697 return NO_INIT; 5698 } 5699 strategy = dstChain->strategy(); 5700 } 5701 if (reRegister) { 5702 AudioSystem::unregisterEffect(effect->id()); 5703 AudioSystem::registerEffect(&effect->desc(), 5704 dstOutput, 5705 strategy, 5706 sessionId, 5707 effect->id()); 5708 } 5709 effect = chain->getEffectFromId_l(0); 5710 } 5711 5712 return NO_ERROR; 5713} 5714 5715 5716// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5717sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5718 const sp<AudioFlinger::Client>& client, 5719 const sp<IEffectClient>& effectClient, 5720 int32_t priority, 5721 int sessionId, 5722 effect_descriptor_t *desc, 5723 int *enabled, 5724 status_t *status 5725 ) 5726{ 5727 sp<EffectModule> effect; 5728 sp<EffectHandle> handle; 5729 status_t lStatus; 5730 sp<EffectChain> chain; 5731 bool chainCreated = false; 5732 bool effectCreated = false; 5733 bool effectRegistered = false; 5734 5735 lStatus = initCheck(); 5736 if (lStatus != NO_ERROR) { 5737 ALOGW("createEffect_l() Audio driver not initialized."); 5738 goto Exit; 5739 } 5740 5741 // Do not allow effects with session ID 0 on direct output or duplicating threads 5742 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5743 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5744 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5745 desc->name, sessionId); 5746 lStatus = BAD_VALUE; 5747 goto Exit; 5748 } 5749 // Only Pre processor effects are allowed on input threads and only on input threads 5750 if ((mType == RECORD && 5751 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5752 (mType != RECORD && 5753 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5754 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5755 desc->name, desc->flags, mType); 5756 lStatus = BAD_VALUE; 5757 goto Exit; 5758 } 5759 5760 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5761 5762 { // scope for mLock 5763 Mutex::Autolock _l(mLock); 5764 5765 // check for existing effect chain with the requested audio session 5766 chain = getEffectChain_l(sessionId); 5767 if (chain == 0) { 5768 // create a new chain for this session 5769 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5770 chain = new EffectChain(this, sessionId); 5771 addEffectChain_l(chain); 5772 chain->setStrategy(getStrategyForSession_l(sessionId)); 5773 chainCreated = true; 5774 } else { 5775 effect = chain->getEffectFromDesc_l(desc); 5776 } 5777 5778 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5779 5780 if (effect == 0) { 5781 int id = mAudioFlinger->nextUniqueId(); 5782 // Check CPU and memory usage 5783 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5784 if (lStatus != NO_ERROR) { 5785 goto Exit; 5786 } 5787 effectRegistered = true; 5788 // create a new effect module if none present in the chain 5789 effect = new EffectModule(this, chain, desc, id, sessionId); 5790 lStatus = effect->status(); 5791 if (lStatus != NO_ERROR) { 5792 goto Exit; 5793 } 5794 lStatus = chain->addEffect_l(effect); 5795 if (lStatus != NO_ERROR) { 5796 goto Exit; 5797 } 5798 effectCreated = true; 5799 5800 effect->setDevice(mDevice); 5801 effect->setMode(mAudioFlinger->getMode()); 5802 } 5803 // create effect handle and connect it to effect module 5804 handle = new EffectHandle(effect, client, effectClient, priority); 5805 lStatus = effect->addHandle(handle); 5806 if (enabled != NULL) { 5807 *enabled = (int)effect->isEnabled(); 5808 } 5809 } 5810 5811Exit: 5812 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5813 Mutex::Autolock _l(mLock); 5814 if (effectCreated) { 5815 chain->removeEffect_l(effect); 5816 } 5817 if (effectRegistered) { 5818 AudioSystem::unregisterEffect(effect->id()); 5819 } 5820 if (chainCreated) { 5821 removeEffectChain_l(chain); 5822 } 5823 handle.clear(); 5824 } 5825 5826 if(status) { 5827 *status = lStatus; 5828 } 5829 return handle; 5830} 5831 5832sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5833{ 5834 sp<EffectModule> effect; 5835 5836 sp<EffectChain> chain = getEffectChain_l(sessionId); 5837 if (chain != 0) { 5838 effect = chain->getEffectFromId_l(effectId); 5839 } 5840 return effect; 5841} 5842 5843// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5844// PlaybackThread::mLock held 5845status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5846{ 5847 // check for existing effect chain with the requested audio session 5848 int sessionId = effect->sessionId(); 5849 sp<EffectChain> chain = getEffectChain_l(sessionId); 5850 bool chainCreated = false; 5851 5852 if (chain == 0) { 5853 // create a new chain for this session 5854 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5855 chain = new EffectChain(this, sessionId); 5856 addEffectChain_l(chain); 5857 chain->setStrategy(getStrategyForSession_l(sessionId)); 5858 chainCreated = true; 5859 } 5860 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5861 5862 if (chain->getEffectFromId_l(effect->id()) != 0) { 5863 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5864 this, effect->desc().name, chain.get()); 5865 return BAD_VALUE; 5866 } 5867 5868 status_t status = chain->addEffect_l(effect); 5869 if (status != NO_ERROR) { 5870 if (chainCreated) { 5871 removeEffectChain_l(chain); 5872 } 5873 return status; 5874 } 5875 5876 effect->setDevice(mDevice); 5877 effect->setMode(mAudioFlinger->getMode()); 5878 return NO_ERROR; 5879} 5880 5881void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5882 5883 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5884 effect_descriptor_t desc = effect->desc(); 5885 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5886 detachAuxEffect_l(effect->id()); 5887 } 5888 5889 sp<EffectChain> chain = effect->chain().promote(); 5890 if (chain != 0) { 5891 // remove effect chain if removing last effect 5892 if (chain->removeEffect_l(effect) == 0) { 5893 removeEffectChain_l(chain); 5894 } 5895 } else { 5896 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5897 } 5898} 5899 5900void AudioFlinger::ThreadBase::lockEffectChains_l( 5901 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5902{ 5903 effectChains = mEffectChains; 5904 for (size_t i = 0; i < mEffectChains.size(); i++) { 5905 mEffectChains[i]->lock(); 5906 } 5907} 5908 5909void AudioFlinger::ThreadBase::unlockEffectChains( 5910 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5911{ 5912 for (size_t i = 0; i < effectChains.size(); i++) { 5913 effectChains[i]->unlock(); 5914 } 5915} 5916 5917sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5918{ 5919 Mutex::Autolock _l(mLock); 5920 return getEffectChain_l(sessionId); 5921} 5922 5923sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5924{ 5925 sp<EffectChain> chain; 5926 5927 size_t size = mEffectChains.size(); 5928 for (size_t i = 0; i < size; i++) { 5929 if (mEffectChains[i]->sessionId() == sessionId) { 5930 chain = mEffectChains[i]; 5931 break; 5932 } 5933 } 5934 return chain; 5935} 5936 5937void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5938{ 5939 Mutex::Autolock _l(mLock); 5940 size_t size = mEffectChains.size(); 5941 for (size_t i = 0; i < size; i++) { 5942 mEffectChains[i]->setMode_l(mode); 5943 } 5944} 5945 5946void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5947 const wp<EffectHandle>& handle, 5948 bool unpiniflast) { 5949 5950 Mutex::Autolock _l(mLock); 5951 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5952 // delete the effect module if removing last handle on it 5953 if (effect->removeHandle(handle) == 0) { 5954 if (!effect->isPinned() || unpiniflast) { 5955 removeEffect_l(effect); 5956 AudioSystem::unregisterEffect(effect->id()); 5957 } 5958 } 5959} 5960 5961status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5962{ 5963 int session = chain->sessionId(); 5964 int16_t *buffer = mMixBuffer; 5965 bool ownsBuffer = false; 5966 5967 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5968 if (session > 0) { 5969 // Only one effect chain can be present in direct output thread and it uses 5970 // the mix buffer as input 5971 if (mType != DIRECT) { 5972 size_t numSamples = mFrameCount * mChannelCount; 5973 buffer = new int16_t[numSamples]; 5974 memset(buffer, 0, numSamples * sizeof(int16_t)); 5975 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5976 ownsBuffer = true; 5977 } 5978 5979 // Attach all tracks with same session ID to this chain. 5980 for (size_t i = 0; i < mTracks.size(); ++i) { 5981 sp<Track> track = mTracks[i]; 5982 if (session == track->sessionId()) { 5983 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5984 track->setMainBuffer(buffer); 5985 chain->incTrackCnt(); 5986 } 5987 } 5988 5989 // indicate all active tracks in the chain 5990 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5991 sp<Track> track = mActiveTracks[i].promote(); 5992 if (track == 0) continue; 5993 if (session == track->sessionId()) { 5994 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5995 chain->incActiveTrackCnt(); 5996 } 5997 } 5998 } 5999 6000 chain->setInBuffer(buffer, ownsBuffer); 6001 chain->setOutBuffer(mMixBuffer); 6002 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6003 // chains list in order to be processed last as it contains output stage effects 6004 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6005 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6006 // after track specific effects and before output stage 6007 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6008 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6009 // Effect chain for other sessions are inserted at beginning of effect 6010 // chains list to be processed before output mix effects. Relative order between other 6011 // sessions is not important 6012 size_t size = mEffectChains.size(); 6013 size_t i = 0; 6014 for (i = 0; i < size; i++) { 6015 if (mEffectChains[i]->sessionId() < session) break; 6016 } 6017 mEffectChains.insertAt(chain, i); 6018 checkSuspendOnAddEffectChain_l(chain); 6019 6020 return NO_ERROR; 6021} 6022 6023size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6024{ 6025 int session = chain->sessionId(); 6026 6027 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6028 6029 for (size_t i = 0; i < mEffectChains.size(); i++) { 6030 if (chain == mEffectChains[i]) { 6031 mEffectChains.removeAt(i); 6032 // detach all active tracks from the chain 6033 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6034 sp<Track> track = mActiveTracks[i].promote(); 6035 if (track == 0) continue; 6036 if (session == track->sessionId()) { 6037 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6038 chain.get(), session); 6039 chain->decActiveTrackCnt(); 6040 } 6041 } 6042 6043 // detach all tracks with same session ID from this chain 6044 for (size_t i = 0; i < mTracks.size(); ++i) { 6045 sp<Track> track = mTracks[i]; 6046 if (session == track->sessionId()) { 6047 track->setMainBuffer(mMixBuffer); 6048 chain->decTrackCnt(); 6049 } 6050 } 6051 break; 6052 } 6053 } 6054 return mEffectChains.size(); 6055} 6056 6057status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6058 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6059{ 6060 Mutex::Autolock _l(mLock); 6061 return attachAuxEffect_l(track, EffectId); 6062} 6063 6064status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6065 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6066{ 6067 status_t status = NO_ERROR; 6068 6069 if (EffectId == 0) { 6070 track->setAuxBuffer(0, NULL); 6071 } else { 6072 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6073 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6074 if (effect != 0) { 6075 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6076 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6077 } else { 6078 status = INVALID_OPERATION; 6079 } 6080 } else { 6081 status = BAD_VALUE; 6082 } 6083 } 6084 return status; 6085} 6086 6087void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6088{ 6089 for (size_t i = 0; i < mTracks.size(); ++i) { 6090 sp<Track> track = mTracks[i]; 6091 if (track->auxEffectId() == effectId) { 6092 attachAuxEffect_l(track, 0); 6093 } 6094 } 6095} 6096 6097status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6098{ 6099 // only one chain per input thread 6100 if (mEffectChains.size() != 0) { 6101 return INVALID_OPERATION; 6102 } 6103 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6104 6105 chain->setInBuffer(NULL); 6106 chain->setOutBuffer(NULL); 6107 6108 checkSuspendOnAddEffectChain_l(chain); 6109 6110 mEffectChains.add(chain); 6111 6112 return NO_ERROR; 6113} 6114 6115size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6116{ 6117 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6118 ALOGW_IF(mEffectChains.size() != 1, 6119 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6120 chain.get(), mEffectChains.size(), this); 6121 if (mEffectChains.size() == 1) { 6122 mEffectChains.removeAt(0); 6123 } 6124 return 0; 6125} 6126 6127// ---------------------------------------------------------------------------- 6128// EffectModule implementation 6129// ---------------------------------------------------------------------------- 6130 6131#undef LOG_TAG 6132#define LOG_TAG "AudioFlinger::EffectModule" 6133 6134AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6135 const wp<AudioFlinger::EffectChain>& chain, 6136 effect_descriptor_t *desc, 6137 int id, 6138 int sessionId) 6139 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6140 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6141{ 6142 ALOGV("Constructor %p", this); 6143 int lStatus; 6144 sp<ThreadBase> thread = mThread.promote(); 6145 if (thread == 0) { 6146 return; 6147 } 6148 6149 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6150 6151 // create effect engine from effect factory 6152 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6153 6154 if (mStatus != NO_ERROR) { 6155 return; 6156 } 6157 lStatus = init(); 6158 if (lStatus < 0) { 6159 mStatus = lStatus; 6160 goto Error; 6161 } 6162 6163 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6164 mPinned = true; 6165 } 6166 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6167 return; 6168Error: 6169 EffectRelease(mEffectInterface); 6170 mEffectInterface = NULL; 6171 ALOGV("Constructor Error %d", mStatus); 6172} 6173 6174AudioFlinger::EffectModule::~EffectModule() 6175{ 6176 ALOGV("Destructor %p", this); 6177 if (mEffectInterface != NULL) { 6178 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6179 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6180 sp<ThreadBase> thread = mThread.promote(); 6181 if (thread != 0) { 6182 audio_stream_t *stream = thread->stream(); 6183 if (stream != NULL) { 6184 stream->remove_audio_effect(stream, mEffectInterface); 6185 } 6186 } 6187 } 6188 // release effect engine 6189 EffectRelease(mEffectInterface); 6190 } 6191} 6192 6193status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6194{ 6195 status_t status; 6196 6197 Mutex::Autolock _l(mLock); 6198 // First handle in mHandles has highest priority and controls the effect module 6199 int priority = handle->priority(); 6200 size_t size = mHandles.size(); 6201 sp<EffectHandle> h; 6202 size_t i; 6203 for (i = 0; i < size; i++) { 6204 h = mHandles[i].promote(); 6205 if (h == 0) continue; 6206 if (h->priority() <= priority) break; 6207 } 6208 // if inserted in first place, move effect control from previous owner to this handle 6209 if (i == 0) { 6210 bool enabled = false; 6211 if (h != 0) { 6212 enabled = h->enabled(); 6213 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6214 } 6215 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6216 status = NO_ERROR; 6217 } else { 6218 status = ALREADY_EXISTS; 6219 } 6220 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6221 mHandles.insertAt(handle, i); 6222 return status; 6223} 6224 6225size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6226{ 6227 Mutex::Autolock _l(mLock); 6228 size_t size = mHandles.size(); 6229 size_t i; 6230 for (i = 0; i < size; i++) { 6231 if (mHandles[i] == handle) break; 6232 } 6233 if (i == size) { 6234 return size; 6235 } 6236 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6237 6238 bool enabled = false; 6239 EffectHandle *hdl = handle.unsafe_get(); 6240 if (hdl != NULL) { 6241 ALOGV("removeHandle() unsafe_get OK"); 6242 enabled = hdl->enabled(); 6243 } 6244 mHandles.removeAt(i); 6245 size = mHandles.size(); 6246 // if removed from first place, move effect control from this handle to next in line 6247 if (i == 0 && size != 0) { 6248 sp<EffectHandle> h = mHandles[0].promote(); 6249 if (h != 0) { 6250 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6251 } 6252 } 6253 6254 // Prevent calls to process() and other functions on effect interface from now on. 6255 // The effect engine will be released by the destructor when the last strong reference on 6256 // this object is released which can happen after next process is called. 6257 if (size == 0 && !mPinned) { 6258 mState = DESTROYED; 6259 } 6260 6261 return size; 6262} 6263 6264sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6265{ 6266 Mutex::Autolock _l(mLock); 6267 sp<EffectHandle> handle; 6268 if (mHandles.size() != 0) { 6269 handle = mHandles[0].promote(); 6270 } 6271 return handle; 6272} 6273 6274void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6275{ 6276 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6277 // keep a strong reference on this EffectModule to avoid calling the 6278 // destructor before we exit 6279 sp<EffectModule> keep(this); 6280 { 6281 sp<ThreadBase> thread = mThread.promote(); 6282 if (thread != 0) { 6283 thread->disconnectEffect(keep, handle, unpiniflast); 6284 } 6285 } 6286} 6287 6288void AudioFlinger::EffectModule::updateState() { 6289 Mutex::Autolock _l(mLock); 6290 6291 switch (mState) { 6292 case RESTART: 6293 reset_l(); 6294 // FALL THROUGH 6295 6296 case STARTING: 6297 // clear auxiliary effect input buffer for next accumulation 6298 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6299 memset(mConfig.inputCfg.buffer.raw, 6300 0, 6301 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6302 } 6303 start_l(); 6304 mState = ACTIVE; 6305 break; 6306 case STOPPING: 6307 stop_l(); 6308 mDisableWaitCnt = mMaxDisableWaitCnt; 6309 mState = STOPPED; 6310 break; 6311 case STOPPED: 6312 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6313 // turn off sequence. 6314 if (--mDisableWaitCnt == 0) { 6315 reset_l(); 6316 mState = IDLE; 6317 } 6318 break; 6319 default: //IDLE , ACTIVE, DESTROYED 6320 break; 6321 } 6322} 6323 6324void AudioFlinger::EffectModule::process() 6325{ 6326 Mutex::Autolock _l(mLock); 6327 6328 if (mState == DESTROYED || mEffectInterface == NULL || 6329 mConfig.inputCfg.buffer.raw == NULL || 6330 mConfig.outputCfg.buffer.raw == NULL) { 6331 return; 6332 } 6333 6334 if (isProcessEnabled()) { 6335 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6336 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6337 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6338 mConfig.inputCfg.buffer.s32, 6339 mConfig.inputCfg.buffer.frameCount/2); 6340 } 6341 6342 // do the actual processing in the effect engine 6343 int ret = (*mEffectInterface)->process(mEffectInterface, 6344 &mConfig.inputCfg.buffer, 6345 &mConfig.outputCfg.buffer); 6346 6347 // force transition to IDLE state when engine is ready 6348 if (mState == STOPPED && ret == -ENODATA) { 6349 mDisableWaitCnt = 1; 6350 } 6351 6352 // clear auxiliary effect input buffer for next accumulation 6353 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6354 memset(mConfig.inputCfg.buffer.raw, 0, 6355 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6356 } 6357 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6358 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6359 // If an insert effect is idle and input buffer is different from output buffer, 6360 // accumulate input onto output 6361 sp<EffectChain> chain = mChain.promote(); 6362 if (chain != 0 && chain->activeTrackCnt() != 0) { 6363 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6364 int16_t *in = mConfig.inputCfg.buffer.s16; 6365 int16_t *out = mConfig.outputCfg.buffer.s16; 6366 for (size_t i = 0; i < frameCnt; i++) { 6367 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6368 } 6369 } 6370 } 6371} 6372 6373void AudioFlinger::EffectModule::reset_l() 6374{ 6375 if (mEffectInterface == NULL) { 6376 return; 6377 } 6378 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6379} 6380 6381status_t AudioFlinger::EffectModule::configure() 6382{ 6383 uint32_t channels; 6384 if (mEffectInterface == NULL) { 6385 return NO_INIT; 6386 } 6387 6388 sp<ThreadBase> thread = mThread.promote(); 6389 if (thread == 0) { 6390 return DEAD_OBJECT; 6391 } 6392 6393 // TODO: handle configuration of effects replacing track process 6394 if (thread->channelCount() == 1) { 6395 channels = AUDIO_CHANNEL_OUT_MONO; 6396 } else { 6397 channels = AUDIO_CHANNEL_OUT_STEREO; 6398 } 6399 6400 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6401 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6402 } else { 6403 mConfig.inputCfg.channels = channels; 6404 } 6405 mConfig.outputCfg.channels = channels; 6406 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6407 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6408 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6409 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6410 mConfig.inputCfg.bufferProvider.cookie = NULL; 6411 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6412 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6413 mConfig.outputCfg.bufferProvider.cookie = NULL; 6414 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6415 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6416 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6417 // Insert effect: 6418 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6419 // always overwrites output buffer: input buffer == output buffer 6420 // - in other sessions: 6421 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6422 // other effect: overwrites output buffer: input buffer == output buffer 6423 // Auxiliary effect: 6424 // accumulates in output buffer: input buffer != output buffer 6425 // Therefore: accumulate <=> input buffer != output buffer 6426 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6427 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6428 } else { 6429 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6430 } 6431 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6432 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6433 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6434 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6435 6436 ALOGV("configure() %p thread %p buffer %p framecount %d", 6437 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6438 6439 status_t cmdStatus; 6440 uint32_t size = sizeof(int); 6441 status_t status = (*mEffectInterface)->command(mEffectInterface, 6442 EFFECT_CMD_SET_CONFIG, 6443 sizeof(effect_config_t), 6444 &mConfig, 6445 &size, 6446 &cmdStatus); 6447 if (status == 0) { 6448 status = cmdStatus; 6449 } 6450 6451 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6452 (1000 * mConfig.outputCfg.buffer.frameCount); 6453 6454 return status; 6455} 6456 6457status_t AudioFlinger::EffectModule::init() 6458{ 6459 Mutex::Autolock _l(mLock); 6460 if (mEffectInterface == NULL) { 6461 return NO_INIT; 6462 } 6463 status_t cmdStatus; 6464 uint32_t size = sizeof(status_t); 6465 status_t status = (*mEffectInterface)->command(mEffectInterface, 6466 EFFECT_CMD_INIT, 6467 0, 6468 NULL, 6469 &size, 6470 &cmdStatus); 6471 if (status == 0) { 6472 status = cmdStatus; 6473 } 6474 return status; 6475} 6476 6477status_t AudioFlinger::EffectModule::start() 6478{ 6479 Mutex::Autolock _l(mLock); 6480 return start_l(); 6481} 6482 6483status_t AudioFlinger::EffectModule::start_l() 6484{ 6485 if (mEffectInterface == NULL) { 6486 return NO_INIT; 6487 } 6488 status_t cmdStatus; 6489 uint32_t size = sizeof(status_t); 6490 status_t status = (*mEffectInterface)->command(mEffectInterface, 6491 EFFECT_CMD_ENABLE, 6492 0, 6493 NULL, 6494 &size, 6495 &cmdStatus); 6496 if (status == 0) { 6497 status = cmdStatus; 6498 } 6499 if (status == 0 && 6500 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6501 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6502 sp<ThreadBase> thread = mThread.promote(); 6503 if (thread != 0) { 6504 audio_stream_t *stream = thread->stream(); 6505 if (stream != NULL) { 6506 stream->add_audio_effect(stream, mEffectInterface); 6507 } 6508 } 6509 } 6510 return status; 6511} 6512 6513status_t AudioFlinger::EffectModule::stop() 6514{ 6515 Mutex::Autolock _l(mLock); 6516 return stop_l(); 6517} 6518 6519status_t AudioFlinger::EffectModule::stop_l() 6520{ 6521 if (mEffectInterface == NULL) { 6522 return NO_INIT; 6523 } 6524 status_t cmdStatus; 6525 uint32_t size = sizeof(status_t); 6526 status_t status = (*mEffectInterface)->command(mEffectInterface, 6527 EFFECT_CMD_DISABLE, 6528 0, 6529 NULL, 6530 &size, 6531 &cmdStatus); 6532 if (status == 0) { 6533 status = cmdStatus; 6534 } 6535 if (status == 0 && 6536 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6537 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6538 sp<ThreadBase> thread = mThread.promote(); 6539 if (thread != 0) { 6540 audio_stream_t *stream = thread->stream(); 6541 if (stream != NULL) { 6542 stream->remove_audio_effect(stream, mEffectInterface); 6543 } 6544 } 6545 } 6546 return status; 6547} 6548 6549status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6550 uint32_t cmdSize, 6551 void *pCmdData, 6552 uint32_t *replySize, 6553 void *pReplyData) 6554{ 6555 Mutex::Autolock _l(mLock); 6556// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6557 6558 if (mState == DESTROYED || mEffectInterface == NULL) { 6559 return NO_INIT; 6560 } 6561 status_t status = (*mEffectInterface)->command(mEffectInterface, 6562 cmdCode, 6563 cmdSize, 6564 pCmdData, 6565 replySize, 6566 pReplyData); 6567 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6568 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6569 for (size_t i = 1; i < mHandles.size(); i++) { 6570 sp<EffectHandle> h = mHandles[i].promote(); 6571 if (h != 0) { 6572 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6573 } 6574 } 6575 } 6576 return status; 6577} 6578 6579status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6580{ 6581 6582 Mutex::Autolock _l(mLock); 6583 ALOGV("setEnabled %p enabled %d", this, enabled); 6584 6585 if (enabled != isEnabled()) { 6586 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6587 if (enabled && status != NO_ERROR) { 6588 return status; 6589 } 6590 6591 switch (mState) { 6592 // going from disabled to enabled 6593 case IDLE: 6594 mState = STARTING; 6595 break; 6596 case STOPPED: 6597 mState = RESTART; 6598 break; 6599 case STOPPING: 6600 mState = ACTIVE; 6601 break; 6602 6603 // going from enabled to disabled 6604 case RESTART: 6605 mState = STOPPED; 6606 break; 6607 case STARTING: 6608 mState = IDLE; 6609 break; 6610 case ACTIVE: 6611 mState = STOPPING; 6612 break; 6613 case DESTROYED: 6614 return NO_ERROR; // simply ignore as we are being destroyed 6615 } 6616 for (size_t i = 1; i < mHandles.size(); i++) { 6617 sp<EffectHandle> h = mHandles[i].promote(); 6618 if (h != 0) { 6619 h->setEnabled(enabled); 6620 } 6621 } 6622 } 6623 return NO_ERROR; 6624} 6625 6626bool AudioFlinger::EffectModule::isEnabled() 6627{ 6628 switch (mState) { 6629 case RESTART: 6630 case STARTING: 6631 case ACTIVE: 6632 return true; 6633 case IDLE: 6634 case STOPPING: 6635 case STOPPED: 6636 case DESTROYED: 6637 default: 6638 return false; 6639 } 6640} 6641 6642bool AudioFlinger::EffectModule::isProcessEnabled() 6643{ 6644 switch (mState) { 6645 case RESTART: 6646 case ACTIVE: 6647 case STOPPING: 6648 case STOPPED: 6649 return true; 6650 case IDLE: 6651 case STARTING: 6652 case DESTROYED: 6653 default: 6654 return false; 6655 } 6656} 6657 6658status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6659{ 6660 Mutex::Autolock _l(mLock); 6661 status_t status = NO_ERROR; 6662 6663 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6664 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6665 if (isProcessEnabled() && 6666 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6667 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6668 status_t cmdStatus; 6669 uint32_t volume[2]; 6670 uint32_t *pVolume = NULL; 6671 uint32_t size = sizeof(volume); 6672 volume[0] = *left; 6673 volume[1] = *right; 6674 if (controller) { 6675 pVolume = volume; 6676 } 6677 status = (*mEffectInterface)->command(mEffectInterface, 6678 EFFECT_CMD_SET_VOLUME, 6679 size, 6680 volume, 6681 &size, 6682 pVolume); 6683 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6684 *left = volume[0]; 6685 *right = volume[1]; 6686 } 6687 } 6688 return status; 6689} 6690 6691status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6692{ 6693 Mutex::Autolock _l(mLock); 6694 status_t status = NO_ERROR; 6695 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6696 // audio pre processing modules on RecordThread can receive both output and 6697 // input device indication in the same call 6698 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6699 if (dev) { 6700 status_t cmdStatus; 6701 uint32_t size = sizeof(status_t); 6702 6703 status = (*mEffectInterface)->command(mEffectInterface, 6704 EFFECT_CMD_SET_DEVICE, 6705 sizeof(uint32_t), 6706 &dev, 6707 &size, 6708 &cmdStatus); 6709 if (status == NO_ERROR) { 6710 status = cmdStatus; 6711 } 6712 } 6713 dev = device & AUDIO_DEVICE_IN_ALL; 6714 if (dev) { 6715 status_t cmdStatus; 6716 uint32_t size = sizeof(status_t); 6717 6718 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6719 EFFECT_CMD_SET_INPUT_DEVICE, 6720 sizeof(uint32_t), 6721 &dev, 6722 &size, 6723 &cmdStatus); 6724 if (status2 == NO_ERROR) { 6725 status2 = cmdStatus; 6726 } 6727 if (status == NO_ERROR) { 6728 status = status2; 6729 } 6730 } 6731 } 6732 return status; 6733} 6734 6735status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6736{ 6737 Mutex::Autolock _l(mLock); 6738 status_t status = NO_ERROR; 6739 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6740 status_t cmdStatus; 6741 uint32_t size = sizeof(status_t); 6742 status = (*mEffectInterface)->command(mEffectInterface, 6743 EFFECT_CMD_SET_AUDIO_MODE, 6744 sizeof(audio_mode_t), 6745 &mode, 6746 &size, 6747 &cmdStatus); 6748 if (status == NO_ERROR) { 6749 status = cmdStatus; 6750 } 6751 } 6752 return status; 6753} 6754 6755void AudioFlinger::EffectModule::setSuspended(bool suspended) 6756{ 6757 Mutex::Autolock _l(mLock); 6758 mSuspended = suspended; 6759} 6760 6761bool AudioFlinger::EffectModule::suspended() const 6762{ 6763 Mutex::Autolock _l(mLock); 6764 return mSuspended; 6765} 6766 6767status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6768{ 6769 const size_t SIZE = 256; 6770 char buffer[SIZE]; 6771 String8 result; 6772 6773 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6774 result.append(buffer); 6775 6776 bool locked = tryLock(mLock); 6777 // failed to lock - AudioFlinger is probably deadlocked 6778 if (!locked) { 6779 result.append("\t\tCould not lock Fx mutex:\n"); 6780 } 6781 6782 result.append("\t\tSession Status State Engine:\n"); 6783 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6784 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6785 result.append(buffer); 6786 6787 result.append("\t\tDescriptor:\n"); 6788 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6789 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6790 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6791 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6792 result.append(buffer); 6793 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6794 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6795 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6796 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6797 result.append(buffer); 6798 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6799 mDescriptor.apiVersion, 6800 mDescriptor.flags); 6801 result.append(buffer); 6802 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6803 mDescriptor.name); 6804 result.append(buffer); 6805 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6806 mDescriptor.implementor); 6807 result.append(buffer); 6808 6809 result.append("\t\t- Input configuration:\n"); 6810 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6811 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6812 (uint32_t)mConfig.inputCfg.buffer.raw, 6813 mConfig.inputCfg.buffer.frameCount, 6814 mConfig.inputCfg.samplingRate, 6815 mConfig.inputCfg.channels, 6816 mConfig.inputCfg.format); 6817 result.append(buffer); 6818 6819 result.append("\t\t- Output configuration:\n"); 6820 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6821 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6822 (uint32_t)mConfig.outputCfg.buffer.raw, 6823 mConfig.outputCfg.buffer.frameCount, 6824 mConfig.outputCfg.samplingRate, 6825 mConfig.outputCfg.channels, 6826 mConfig.outputCfg.format); 6827 result.append(buffer); 6828 6829 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6830 result.append(buffer); 6831 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6832 for (size_t i = 0; i < mHandles.size(); ++i) { 6833 sp<EffectHandle> handle = mHandles[i].promote(); 6834 if (handle != 0) { 6835 handle->dump(buffer, SIZE); 6836 result.append(buffer); 6837 } 6838 } 6839 6840 result.append("\n"); 6841 6842 write(fd, result.string(), result.length()); 6843 6844 if (locked) { 6845 mLock.unlock(); 6846 } 6847 6848 return NO_ERROR; 6849} 6850 6851// ---------------------------------------------------------------------------- 6852// EffectHandle implementation 6853// ---------------------------------------------------------------------------- 6854 6855#undef LOG_TAG 6856#define LOG_TAG "AudioFlinger::EffectHandle" 6857 6858AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6859 const sp<AudioFlinger::Client>& client, 6860 const sp<IEffectClient>& effectClient, 6861 int32_t priority) 6862 : BnEffect(), 6863 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6864 mPriority(priority), mHasControl(false), mEnabled(false) 6865{ 6866 ALOGV("constructor %p", this); 6867 6868 if (client == 0) { 6869 return; 6870 } 6871 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6872 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6873 if (mCblkMemory != 0) { 6874 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6875 6876 if (mCblk != NULL) { 6877 new(mCblk) effect_param_cblk_t(); 6878 mBuffer = (uint8_t *)mCblk + bufOffset; 6879 } 6880 } else { 6881 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6882 return; 6883 } 6884} 6885 6886AudioFlinger::EffectHandle::~EffectHandle() 6887{ 6888 ALOGV("Destructor %p", this); 6889 disconnect(false); 6890 ALOGV("Destructor DONE %p", this); 6891} 6892 6893status_t AudioFlinger::EffectHandle::enable() 6894{ 6895 ALOGV("enable %p", this); 6896 if (!mHasControl) return INVALID_OPERATION; 6897 if (mEffect == 0) return DEAD_OBJECT; 6898 6899 if (mEnabled) { 6900 return NO_ERROR; 6901 } 6902 6903 mEnabled = true; 6904 6905 sp<ThreadBase> thread = mEffect->thread().promote(); 6906 if (thread != 0) { 6907 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6908 } 6909 6910 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6911 if (mEffect->suspended()) { 6912 return NO_ERROR; 6913 } 6914 6915 status_t status = mEffect->setEnabled(true); 6916 if (status != NO_ERROR) { 6917 if (thread != 0) { 6918 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6919 } 6920 mEnabled = false; 6921 } 6922 return status; 6923} 6924 6925status_t AudioFlinger::EffectHandle::disable() 6926{ 6927 ALOGV("disable %p", this); 6928 if (!mHasControl) return INVALID_OPERATION; 6929 if (mEffect == 0) return DEAD_OBJECT; 6930 6931 if (!mEnabled) { 6932 return NO_ERROR; 6933 } 6934 mEnabled = false; 6935 6936 if (mEffect->suspended()) { 6937 return NO_ERROR; 6938 } 6939 6940 status_t status = mEffect->setEnabled(false); 6941 6942 sp<ThreadBase> thread = mEffect->thread().promote(); 6943 if (thread != 0) { 6944 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6945 } 6946 6947 return status; 6948} 6949 6950void AudioFlinger::EffectHandle::disconnect() 6951{ 6952 disconnect(true); 6953} 6954 6955void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6956{ 6957 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6958 if (mEffect == 0) { 6959 return; 6960 } 6961 mEffect->disconnect(this, unpiniflast); 6962 6963 if (mHasControl && mEnabled) { 6964 sp<ThreadBase> thread = mEffect->thread().promote(); 6965 if (thread != 0) { 6966 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6967 } 6968 } 6969 6970 // release sp on module => module destructor can be called now 6971 mEffect.clear(); 6972 if (mClient != 0) { 6973 if (mCblk != NULL) { 6974 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 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 == NULL) ? 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