AudioFlinger.cpp revision 5c0ad10b14ec2287f90f95912d98e66eef006e2a
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), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 164 mBtNrecIsOff(false) 165{ 166} 167 168void AudioFlinger::onFirstRef() 169{ 170 int rc = 0; 171 172 Mutex::Autolock _l(mLock); 173 174 /* TODO: move all this work into an Init() function */ 175 mHardwareStatus = AUDIO_HW_IDLE; 176 177 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 178 const hw_module_t *mod; 179 audio_hw_device_t *dev; 180 181 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 182 if (rc) 183 continue; 184 185 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 186 mod->name, mod->id); 187 mAudioHwDevs.push(dev); 188 189 if (!mPrimaryHardwareDev) { 190 mPrimaryHardwareDev = dev; 191 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 192 mod->name, mod->id, audio_interfaces[i]); 193 } 194 } 195 196 mHardwareStatus = AUDIO_HW_INIT; 197 198 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 199 ALOGE("Primary audio interface not found"); 200 return; 201 } 202 203 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 204 audio_hw_device_t *dev = mAudioHwDevs[i]; 205 206 mHardwareStatus = AUDIO_HW_INIT; 207 rc = dev->init_check(dev); 208 if (rc == 0) { 209 AutoMutex lock(mHardwareLock); 210 211 mMode = AUDIO_MODE_NORMAL; 212 mHardwareStatus = AUDIO_HW_SET_MODE; 213 dev->set_mode(dev, mMode); 214 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 215 dev->set_master_volume(dev, 1.0f); 216 mHardwareStatus = AUDIO_HW_IDLE; 217 } 218 } 219} 220 221status_t AudioFlinger::initCheck() const 222{ 223 Mutex::Autolock _l(mLock); 224 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 225 return NO_INIT; 226 return NO_ERROR; 227} 228 229AudioFlinger::~AudioFlinger() 230{ 231 int num_devs = mAudioHwDevs.size(); 232 233 while (!mRecordThreads.isEmpty()) { 234 // closeInput() will remove first entry from mRecordThreads 235 closeInput(mRecordThreads.keyAt(0)); 236 } 237 while (!mPlaybackThreads.isEmpty()) { 238 // closeOutput() will remove first entry from mPlaybackThreads 239 closeOutput(mPlaybackThreads.keyAt(0)); 240 } 241 242 for (int i = 0; i < num_devs; i++) { 243 audio_hw_device_t *dev = mAudioHwDevs[i]; 244 audio_hw_device_close(dev); 245 } 246 mAudioHwDevs.clear(); 247} 248 249audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 250{ 251 /* first matching HW device is returned */ 252 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 253 audio_hw_device_t *dev = mAudioHwDevs[i]; 254 if ((dev->get_supported_devices(dev) & devices) == devices) 255 return dev; 256 } 257 return NULL; 258} 259 260status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 261{ 262 const size_t SIZE = 256; 263 char buffer[SIZE]; 264 String8 result; 265 266 result.append("Clients:\n"); 267 for (size_t i = 0; i < mClients.size(); ++i) { 268 wp<Client> wClient = mClients.valueAt(i); 269 if (wClient != 0) { 270 sp<Client> client = wClient.promote(); 271 if (client != 0) { 272 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 273 result.append(buffer); 274 } 275 } 276 } 277 278 result.append("Global session refs:\n"); 279 result.append(" session pid cnt\n"); 280 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 281 AudioSessionRef *r = mAudioSessionRefs[i]; 282 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 283 result.append(buffer); 284 } 285 write(fd, result.string(), result.size()); 286 return NO_ERROR; 287} 288 289 290status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 291{ 292 const size_t SIZE = 256; 293 char buffer[SIZE]; 294 String8 result; 295 hardware_call_state hardwareStatus = mHardwareStatus; 296 297 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 298 result.append(buffer); 299 write(fd, result.string(), result.size()); 300 return NO_ERROR; 301} 302 303status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 304{ 305 const size_t SIZE = 256; 306 char buffer[SIZE]; 307 String8 result; 308 snprintf(buffer, SIZE, "Permission Denial: " 309 "can't dump AudioFlinger from pid=%d, uid=%d\n", 310 IPCThreadState::self()->getCallingPid(), 311 IPCThreadState::self()->getCallingUid()); 312 result.append(buffer); 313 write(fd, result.string(), result.size()); 314 return NO_ERROR; 315} 316 317static bool tryLock(Mutex& mutex) 318{ 319 bool locked = false; 320 for (int i = 0; i < kDumpLockRetries; ++i) { 321 if (mutex.tryLock() == NO_ERROR) { 322 locked = true; 323 break; 324 } 325 usleep(kDumpLockSleepUs); 326 } 327 return locked; 328} 329 330status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 331{ 332 if (!checkCallingPermission(String16("android.permission.DUMP"))) { 333 dumpPermissionDenial(fd, args); 334 } else { 335 // get state of hardware lock 336 bool hardwareLocked = tryLock(mHardwareLock); 337 if (!hardwareLocked) { 338 String8 result(kHardwareLockedString); 339 write(fd, result.string(), result.size()); 340 } else { 341 mHardwareLock.unlock(); 342 } 343 344 bool locked = tryLock(mLock); 345 346 // failed to lock - AudioFlinger is probably deadlocked 347 if (!locked) { 348 String8 result(kDeadlockedString); 349 write(fd, result.string(), result.size()); 350 } 351 352 dumpClients(fd, args); 353 dumpInternals(fd, args); 354 355 // dump playback threads 356 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 357 mPlaybackThreads.valueAt(i)->dump(fd, args); 358 } 359 360 // dump record threads 361 for (size_t i = 0; i < mRecordThreads.size(); i++) { 362 mRecordThreads.valueAt(i)->dump(fd, args); 363 } 364 365 // dump all hardware devs 366 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 367 audio_hw_device_t *dev = mAudioHwDevs[i]; 368 dev->dump(dev, fd); 369 } 370 if (locked) mLock.unlock(); 371 } 372 return NO_ERROR; 373} 374 375 376// IAudioFlinger interface 377 378 379sp<IAudioTrack> AudioFlinger::createTrack( 380 pid_t pid, 381 audio_stream_type_t streamType, 382 uint32_t sampleRate, 383 audio_format_t format, 384 uint32_t channelMask, 385 int frameCount, 386 uint32_t flags, 387 const sp<IMemory>& sharedBuffer, 388 int output, 389 int *sessionId, 390 status_t *status) 391{ 392 sp<PlaybackThread::Track> track; 393 sp<TrackHandle> trackHandle; 394 sp<Client> client; 395 wp<Client> wclient; 396 status_t lStatus; 397 int lSessionId; 398 399 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 400 // but if someone uses binder directly they could bypass that and cause us to crash 401 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 402 ALOGE("createTrack() invalid stream type %d", streamType); 403 lStatus = BAD_VALUE; 404 goto Exit; 405 } 406 407 { 408 Mutex::Autolock _l(mLock); 409 PlaybackThread *thread = checkPlaybackThread_l(output); 410 PlaybackThread *effectThread = NULL; 411 if (thread == NULL) { 412 ALOGE("unknown output thread"); 413 lStatus = BAD_VALUE; 414 goto Exit; 415 } 416 417 wclient = mClients.valueFor(pid); 418 419 if (wclient != NULL) { 420 client = wclient.promote(); 421 } else { 422 client = new Client(this, pid); 423 mClients.add(pid, client); 424 } 425 426 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 427 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 428 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 429 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 430 if (mPlaybackThreads.keyAt(i) != output) { 431 // prevent same audio session on different output threads 432 uint32_t sessions = t->hasAudioSession(*sessionId); 433 if (sessions & PlaybackThread::TRACK_SESSION) { 434 ALOGE("createTrack() session ID %d already in use", *sessionId); 435 lStatus = BAD_VALUE; 436 goto Exit; 437 } 438 // check if an effect with same session ID is waiting for a track to be created 439 if (sessions & PlaybackThread::EFFECT_SESSION) { 440 effectThread = t.get(); 441 } 442 } 443 } 444 lSessionId = *sessionId; 445 } else { 446 // if no audio session id is provided, create one here 447 lSessionId = nextUniqueId(); 448 if (sessionId != NULL) { 449 *sessionId = lSessionId; 450 } 451 } 452 ALOGV("createTrack() lSessionId: %d", lSessionId); 453 454 track = thread->createTrack_l(client, streamType, sampleRate, format, 455 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 456 457 // move effect chain to this output thread if an effect on same session was waiting 458 // for a track to be created 459 if (lStatus == NO_ERROR && effectThread != NULL) { 460 Mutex::Autolock _dl(thread->mLock); 461 Mutex::Autolock _sl(effectThread->mLock); 462 moveEffectChain_l(lSessionId, effectThread, thread, true); 463 } 464 } 465 if (lStatus == NO_ERROR) { 466 trackHandle = new TrackHandle(track); 467 } else { 468 // remove local strong reference to Client before deleting the Track so that the Client 469 // destructor is called by the TrackBase destructor with mLock held 470 client.clear(); 471 track.clear(); 472 } 473 474Exit: 475 if(status) { 476 *status = lStatus; 477 } 478 return trackHandle; 479} 480 481uint32_t AudioFlinger::sampleRate(int output) const 482{ 483 Mutex::Autolock _l(mLock); 484 PlaybackThread *thread = checkPlaybackThread_l(output); 485 if (thread == NULL) { 486 ALOGW("sampleRate() unknown thread %d", output); 487 return 0; 488 } 489 return thread->sampleRate(); 490} 491 492int AudioFlinger::channelCount(int output) const 493{ 494 Mutex::Autolock _l(mLock); 495 PlaybackThread *thread = checkPlaybackThread_l(output); 496 if (thread == NULL) { 497 ALOGW("channelCount() unknown thread %d", output); 498 return 0; 499 } 500 return thread->channelCount(); 501} 502 503audio_format_t AudioFlinger::format(int output) const 504{ 505 Mutex::Autolock _l(mLock); 506 PlaybackThread *thread = checkPlaybackThread_l(output); 507 if (thread == NULL) { 508 ALOGW("format() unknown thread %d", output); 509 return AUDIO_FORMAT_INVALID; 510 } 511 return thread->format(); 512} 513 514size_t AudioFlinger::frameCount(int output) const 515{ 516 Mutex::Autolock _l(mLock); 517 PlaybackThread *thread = checkPlaybackThread_l(output); 518 if (thread == NULL) { 519 ALOGW("frameCount() unknown thread %d", output); 520 return 0; 521 } 522 return thread->frameCount(); 523} 524 525uint32_t AudioFlinger::latency(int output) const 526{ 527 Mutex::Autolock _l(mLock); 528 PlaybackThread *thread = checkPlaybackThread_l(output); 529 if (thread == NULL) { 530 ALOGW("latency() unknown thread %d", output); 531 return 0; 532 } 533 return thread->latency(); 534} 535 536status_t AudioFlinger::setMasterVolume(float value) 537{ 538 status_t ret = initCheck(); 539 if (ret != NO_ERROR) { 540 return ret; 541 } 542 543 // check calling permissions 544 if (!settingsAllowed()) { 545 return PERMISSION_DENIED; 546 } 547 548 // when hw supports master volume, don't scale in sw mixer 549 { // scope for the lock 550 AutoMutex lock(mHardwareLock); 551 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 552 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 553 value = 1.0f; 554 } 555 mHardwareStatus = AUDIO_HW_IDLE; 556 } 557 558 Mutex::Autolock _l(mLock); 559 mMasterVolume = value; 560 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 561 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 562 563 return NO_ERROR; 564} 565 566status_t AudioFlinger::setMode(audio_mode_t mode) 567{ 568 status_t ret = initCheck(); 569 if (ret != NO_ERROR) { 570 return ret; 571 } 572 573 // check calling permissions 574 if (!settingsAllowed()) { 575 return PERMISSION_DENIED; 576 } 577 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 578 ALOGW("Illegal value: setMode(%d)", mode); 579 return BAD_VALUE; 580 } 581 582 { // scope for the lock 583 AutoMutex lock(mHardwareLock); 584 mHardwareStatus = AUDIO_HW_SET_MODE; 585 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 586 mHardwareStatus = AUDIO_HW_IDLE; 587 } 588 589 if (NO_ERROR == ret) { 590 Mutex::Autolock _l(mLock); 591 mMode = mode; 592 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 593 mPlaybackThreads.valueAt(i)->setMode(mode); 594 } 595 596 return ret; 597} 598 599status_t AudioFlinger::setMicMute(bool state) 600{ 601 status_t ret = initCheck(); 602 if (ret != NO_ERROR) { 603 return ret; 604 } 605 606 // check calling permissions 607 if (!settingsAllowed()) { 608 return PERMISSION_DENIED; 609 } 610 611 AutoMutex lock(mHardwareLock); 612 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 613 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 614 mHardwareStatus = AUDIO_HW_IDLE; 615 return ret; 616} 617 618bool AudioFlinger::getMicMute() const 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return false; 623 } 624 625 bool state = AUDIO_MODE_INVALID; 626 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 627 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 628 mHardwareStatus = AUDIO_HW_IDLE; 629 return state; 630} 631 632status_t AudioFlinger::setMasterMute(bool muted) 633{ 634 // check calling permissions 635 if (!settingsAllowed()) { 636 return PERMISSION_DENIED; 637 } 638 639 Mutex::Autolock _l(mLock); 640 mMasterMute = muted; 641 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 642 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 643 644 return NO_ERROR; 645} 646 647float AudioFlinger::masterVolume() const 648{ 649 Mutex::Autolock _l(mLock); 650 return masterVolume_l(); 651} 652 653bool AudioFlinger::masterMute() const 654{ 655 Mutex::Autolock _l(mLock); 656 return masterMute_l(); 657} 658 659status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output) 660{ 661 // check calling permissions 662 if (!settingsAllowed()) { 663 return PERMISSION_DENIED; 664 } 665 666 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 667 ALOGE("setStreamVolume() invalid stream %d", stream); 668 return BAD_VALUE; 669 } 670 671 AutoMutex lock(mLock); 672 PlaybackThread *thread = NULL; 673 if (output) { 674 thread = checkPlaybackThread_l(output); 675 if (thread == NULL) { 676 return BAD_VALUE; 677 } 678 } 679 680 mStreamTypes[stream].volume = value; 681 682 if (thread == NULL) { 683 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 684 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 685 } 686 } else { 687 thread->setStreamVolume(stream, value); 688 } 689 690 return NO_ERROR; 691} 692 693status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 694{ 695 // check calling permissions 696 if (!settingsAllowed()) { 697 return PERMISSION_DENIED; 698 } 699 700 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 701 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 702 ALOGE("setStreamMute() invalid stream %d", stream); 703 return BAD_VALUE; 704 } 705 706 AutoMutex lock(mLock); 707 mStreamTypes[stream].mute = muted; 708 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 709 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 710 711 return NO_ERROR; 712} 713 714float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const 715{ 716 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 717 return 0.0f; 718 } 719 720 AutoMutex lock(mLock); 721 float volume; 722 if (output) { 723 PlaybackThread *thread = checkPlaybackThread_l(output); 724 if (thread == NULL) { 725 return 0.0f; 726 } 727 volume = thread->streamVolume(stream); 728 } else { 729 volume = mStreamTypes[stream].volume; 730 } 731 732 return volume; 733} 734 735bool AudioFlinger::streamMute(audio_stream_type_t stream) const 736{ 737 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 738 return true; 739 } 740 741 return mStreamTypes[stream].mute; 742} 743 744status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 745{ 746 status_t result; 747 748 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 749 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 750 // check calling permissions 751 if (!settingsAllowed()) { 752 return PERMISSION_DENIED; 753 } 754 755 // ioHandle == 0 means the parameters are global to the audio hardware interface 756 if (ioHandle == 0) { 757 AutoMutex lock(mHardwareLock); 758 mHardwareStatus = AUDIO_SET_PARAMETER; 759 status_t final_result = NO_ERROR; 760 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 761 audio_hw_device_t *dev = mAudioHwDevs[i]; 762 result = dev->set_parameters(dev, keyValuePairs.string()); 763 final_result = result ?: final_result; 764 } 765 mHardwareStatus = AUDIO_HW_IDLE; 766 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 767 AudioParameter param = AudioParameter(keyValuePairs); 768 String8 value; 769 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 770 Mutex::Autolock _l(mLock); 771 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 772 if (mBtNrecIsOff != btNrecIsOff) { 773 for (size_t i = 0; i < mRecordThreads.size(); i++) { 774 sp<RecordThread> thread = mRecordThreads.valueAt(i); 775 RecordThread::RecordTrack *track = thread->track(); 776 if (track != NULL) { 777 audio_devices_t device = (audio_devices_t)( 778 thread->device() & AUDIO_DEVICE_IN_ALL); 779 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 780 thread->setEffectSuspended(FX_IID_AEC, 781 suspend, 782 track->sessionId()); 783 thread->setEffectSuspended(FX_IID_NS, 784 suspend, 785 track->sessionId()); 786 } 787 } 788 mBtNrecIsOff = btNrecIsOff; 789 } 790 } 791 return final_result; 792 } 793 794 // hold a strong ref on thread in case closeOutput() or closeInput() is called 795 // and the thread is exited once the lock is released 796 sp<ThreadBase> thread; 797 { 798 Mutex::Autolock _l(mLock); 799 thread = checkPlaybackThread_l(ioHandle); 800 if (thread == NULL) { 801 thread = checkRecordThread_l(ioHandle); 802 } else if (thread == primaryPlaybackThread_l()) { 803 // indicate output device change to all input threads for pre processing 804 AudioParameter param = AudioParameter(keyValuePairs); 805 int value; 806 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 807 for (size_t i = 0; i < mRecordThreads.size(); i++) { 808 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 809 } 810 } 811 } 812 } 813 if (thread != NULL) { 814 result = thread->setParameters(keyValuePairs); 815 return result; 816 } 817 return BAD_VALUE; 818} 819 820String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 821{ 822// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 823// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 824 825 if (ioHandle == 0) { 826 String8 out_s8; 827 828 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 829 audio_hw_device_t *dev = mAudioHwDevs[i]; 830 char *s = dev->get_parameters(dev, keys.string()); 831 out_s8 += String8(s); 832 free(s); 833 } 834 return out_s8; 835 } 836 837 Mutex::Autolock _l(mLock); 838 839 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 840 if (playbackThread != NULL) { 841 return playbackThread->getParameters(keys); 842 } 843 RecordThread *recordThread = checkRecordThread_l(ioHandle); 844 if (recordThread != NULL) { 845 return recordThread->getParameters(keys); 846 } 847 return String8(""); 848} 849 850size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) 851{ 852 status_t ret = initCheck(); 853 if (ret != NO_ERROR) { 854 return 0; 855 } 856 857 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 858} 859 860unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 861{ 862 if (ioHandle == 0) { 863 return 0; 864 } 865 866 Mutex::Autolock _l(mLock); 867 868 RecordThread *recordThread = checkRecordThread_l(ioHandle); 869 if (recordThread != NULL) { 870 return recordThread->getInputFramesLost(); 871 } 872 return 0; 873} 874 875status_t AudioFlinger::setVoiceVolume(float value) 876{ 877 status_t ret = initCheck(); 878 if (ret != NO_ERROR) { 879 return ret; 880 } 881 882 // check calling permissions 883 if (!settingsAllowed()) { 884 return PERMISSION_DENIED; 885 } 886 887 AutoMutex lock(mHardwareLock); 888 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 889 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 890 mHardwareStatus = AUDIO_HW_IDLE; 891 892 return ret; 893} 894 895status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 896{ 897 status_t status; 898 899 Mutex::Autolock _l(mLock); 900 901 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 902 if (playbackThread != NULL) { 903 return playbackThread->getRenderPosition(halFrames, dspFrames); 904 } 905 906 return BAD_VALUE; 907} 908 909void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 910{ 911 912 Mutex::Autolock _l(mLock); 913 914 int pid = IPCThreadState::self()->getCallingPid(); 915 if (mNotificationClients.indexOfKey(pid) < 0) { 916 sp<NotificationClient> notificationClient = new NotificationClient(this, 917 client, 918 pid); 919 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 920 921 mNotificationClients.add(pid, notificationClient); 922 923 sp<IBinder> binder = client->asBinder(); 924 binder->linkToDeath(notificationClient); 925 926 // the config change is always sent from playback or record threads to avoid deadlock 927 // with AudioSystem::gLock 928 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 929 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 930 } 931 932 for (size_t i = 0; i < mRecordThreads.size(); i++) { 933 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 934 } 935 } 936} 937 938void AudioFlinger::removeNotificationClient(pid_t pid) 939{ 940 Mutex::Autolock _l(mLock); 941 942 int index = mNotificationClients.indexOfKey(pid); 943 if (index >= 0) { 944 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 945 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 946 mNotificationClients.removeItem(pid); 947 } 948 949 ALOGV("%d died, releasing its sessions", pid); 950 int num = mAudioSessionRefs.size(); 951 bool removed = false; 952 for (int i = 0; i< num; i++) { 953 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 954 ALOGV(" pid %d @ %d", ref->pid, i); 955 if (ref->pid == pid) { 956 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 957 mAudioSessionRefs.removeAt(i); 958 delete ref; 959 removed = true; 960 i--; 961 num--; 962 } 963 } 964 if (removed) { 965 purgeStaleEffects_l(); 966 } 967} 968 969// audioConfigChanged_l() must be called with AudioFlinger::mLock held 970void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 971{ 972 size_t size = mNotificationClients.size(); 973 for (size_t i = 0; i < size; i++) { 974 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, 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), mChannelCount(0), 993 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false), 994 mDevice(device) 995{ 996 mDeathRecipient = new PMDeathRecipient(this); 997} 998 999AudioFlinger::ThreadBase::~ThreadBase() 1000{ 1001 mParamCond.broadcast(); 1002 // do not lock the mutex in destructor 1003 releaseWakeLock_l(); 1004 if (mPowerManager != 0) { 1005 sp<IBinder> binder = mPowerManager->asBinder(); 1006 binder->unlinkToDeath(mDeathRecipient); 1007 } 1008} 1009 1010void AudioFlinger::ThreadBase::exit() 1011{ 1012 // keep a strong ref on ourself so that we won't get 1013 // destroyed in the middle of requestExitAndWait() 1014 sp <ThreadBase> strongMe = this; 1015 1016 ALOGV("ThreadBase::exit"); 1017 { 1018 AutoMutex lock(mLock); 1019 mExiting = true; 1020 requestExit(); 1021 mWaitWorkCV.signal(); 1022 } 1023 requestExitAndWait(); 1024} 1025 1026uint32_t AudioFlinger::ThreadBase::sampleRate() const 1027{ 1028 return mSampleRate; 1029} 1030 1031int AudioFlinger::ThreadBase::channelCount() const 1032{ 1033 return (int)mChannelCount; 1034} 1035 1036audio_format_t AudioFlinger::ThreadBase::format() const 1037{ 1038 return mFormat; 1039} 1040 1041size_t AudioFlinger::ThreadBase::frameCount() const 1042{ 1043 return mFrameCount; 1044} 1045 1046status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1047{ 1048 status_t status; 1049 1050 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1051 Mutex::Autolock _l(mLock); 1052 1053 mNewParameters.add(keyValuePairs); 1054 mWaitWorkCV.signal(); 1055 // wait condition with timeout in case the thread loop has exited 1056 // before the request could be processed 1057 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1058 status = mParamStatus; 1059 mWaitWorkCV.signal(); 1060 } else { 1061 status = TIMED_OUT; 1062 } 1063 return status; 1064} 1065 1066void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1067{ 1068 Mutex::Autolock _l(mLock); 1069 sendConfigEvent_l(event, param); 1070} 1071 1072// sendConfigEvent_l() must be called with ThreadBase::mLock held 1073void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1074{ 1075 ConfigEvent configEvent; 1076 configEvent.mEvent = event; 1077 configEvent.mParam = param; 1078 mConfigEvents.add(configEvent); 1079 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1080 mWaitWorkCV.signal(); 1081} 1082 1083void AudioFlinger::ThreadBase::processConfigEvents() 1084{ 1085 mLock.lock(); 1086 while(!mConfigEvents.isEmpty()) { 1087 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1088 ConfigEvent configEvent = mConfigEvents[0]; 1089 mConfigEvents.removeAt(0); 1090 // release mLock before locking AudioFlinger mLock: lock order is always 1091 // AudioFlinger then ThreadBase to avoid cross deadlock 1092 mLock.unlock(); 1093 mAudioFlinger->mLock.lock(); 1094 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1095 mAudioFlinger->mLock.unlock(); 1096 mLock.lock(); 1097 } 1098 mLock.unlock(); 1099} 1100 1101status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1102{ 1103 const size_t SIZE = 256; 1104 char buffer[SIZE]; 1105 String8 result; 1106 1107 bool locked = tryLock(mLock); 1108 if (!locked) { 1109 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1110 write(fd, buffer, strlen(buffer)); 1111 } 1112 1113 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1114 result.append(buffer); 1115 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1116 result.append(buffer); 1117 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1122 result.append(buffer); 1123 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1124 result.append(buffer); 1125 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1126 result.append(buffer); 1127 1128 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1129 result.append(buffer); 1130 result.append(" Index Command"); 1131 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1132 snprintf(buffer, SIZE, "\n %02d ", i); 1133 result.append(buffer); 1134 result.append(mNewParameters[i]); 1135 } 1136 1137 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1138 result.append(buffer); 1139 snprintf(buffer, SIZE, " Index event param\n"); 1140 result.append(buffer); 1141 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1142 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1143 result.append(buffer); 1144 } 1145 result.append("\n"); 1146 1147 write(fd, result.string(), result.size()); 1148 1149 if (locked) { 1150 mLock.unlock(); 1151 } 1152 return NO_ERROR; 1153} 1154 1155status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1156{ 1157 const size_t SIZE = 256; 1158 char buffer[SIZE]; 1159 String8 result; 1160 1161 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1162 write(fd, buffer, strlen(buffer)); 1163 1164 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1165 sp<EffectChain> chain = mEffectChains[i]; 1166 if (chain != 0) { 1167 chain->dump(fd, args); 1168 } 1169 } 1170 return NO_ERROR; 1171} 1172 1173void AudioFlinger::ThreadBase::acquireWakeLock() 1174{ 1175 Mutex::Autolock _l(mLock); 1176 acquireWakeLock_l(); 1177} 1178 1179void AudioFlinger::ThreadBase::acquireWakeLock_l() 1180{ 1181 if (mPowerManager == 0) { 1182 // use checkService() to avoid blocking if power service is not up yet 1183 sp<IBinder> binder = 1184 defaultServiceManager()->checkService(String16("power")); 1185 if (binder == 0) { 1186 ALOGW("Thread %s cannot connect to the power manager service", mName); 1187 } else { 1188 mPowerManager = interface_cast<IPowerManager>(binder); 1189 binder->linkToDeath(mDeathRecipient); 1190 } 1191 } 1192 if (mPowerManager != 0) { 1193 sp<IBinder> binder = new BBinder(); 1194 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1195 binder, 1196 String16(mName)); 1197 if (status == NO_ERROR) { 1198 mWakeLockToken = binder; 1199 } 1200 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1201 } 1202} 1203 1204void AudioFlinger::ThreadBase::releaseWakeLock() 1205{ 1206 Mutex::Autolock _l(mLock); 1207 releaseWakeLock_l(); 1208} 1209 1210void AudioFlinger::ThreadBase::releaseWakeLock_l() 1211{ 1212 if (mWakeLockToken != 0) { 1213 ALOGV("releaseWakeLock_l() %s", mName); 1214 if (mPowerManager != 0) { 1215 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1216 } 1217 mWakeLockToken.clear(); 1218 } 1219} 1220 1221void AudioFlinger::ThreadBase::clearPowerManager() 1222{ 1223 Mutex::Autolock _l(mLock); 1224 releaseWakeLock_l(); 1225 mPowerManager.clear(); 1226} 1227 1228void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1229{ 1230 sp<ThreadBase> thread = mThread.promote(); 1231 if (thread != 0) { 1232 thread->clearPowerManager(); 1233 } 1234 ALOGW("power manager service died !!!"); 1235} 1236 1237void AudioFlinger::ThreadBase::setEffectSuspended( 1238 const effect_uuid_t *type, bool suspend, int sessionId) 1239{ 1240 Mutex::Autolock _l(mLock); 1241 setEffectSuspended_l(type, suspend, sessionId); 1242} 1243 1244void AudioFlinger::ThreadBase::setEffectSuspended_l( 1245 const effect_uuid_t *type, bool suspend, int sessionId) 1246{ 1247 sp<EffectChain> chain; 1248 chain = getEffectChain_l(sessionId); 1249 if (chain != 0) { 1250 if (type != NULL) { 1251 chain->setEffectSuspended_l(type, suspend); 1252 } else { 1253 chain->setEffectSuspendedAll_l(suspend); 1254 } 1255 } 1256 1257 updateSuspendedSessions_l(type, suspend, sessionId); 1258} 1259 1260void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1261{ 1262 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1263 if (index < 0) { 1264 return; 1265 } 1266 1267 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1268 mSuspendedSessions.editValueAt(index); 1269 1270 for (size_t i = 0; i < sessionEffects.size(); i++) { 1271 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1272 for (int j = 0; j < desc->mRefCount; j++) { 1273 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1274 chain->setEffectSuspendedAll_l(true); 1275 } else { 1276 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1277 desc->mType.timeLow); 1278 chain->setEffectSuspended_l(&desc->mType, true); 1279 } 1280 } 1281 } 1282} 1283 1284void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1285 bool suspend, 1286 int sessionId) 1287{ 1288 int index = mSuspendedSessions.indexOfKey(sessionId); 1289 1290 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1291 1292 if (suspend) { 1293 if (index >= 0) { 1294 sessionEffects = mSuspendedSessions.editValueAt(index); 1295 } else { 1296 mSuspendedSessions.add(sessionId, sessionEffects); 1297 } 1298 } else { 1299 if (index < 0) { 1300 return; 1301 } 1302 sessionEffects = mSuspendedSessions.editValueAt(index); 1303 } 1304 1305 1306 int key = EffectChain::kKeyForSuspendAll; 1307 if (type != NULL) { 1308 key = type->timeLow; 1309 } 1310 index = sessionEffects.indexOfKey(key); 1311 1312 sp <SuspendedSessionDesc> desc; 1313 if (suspend) { 1314 if (index >= 0) { 1315 desc = sessionEffects.valueAt(index); 1316 } else { 1317 desc = new SuspendedSessionDesc(); 1318 if (type != NULL) { 1319 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1320 } 1321 sessionEffects.add(key, desc); 1322 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1323 } 1324 desc->mRefCount++; 1325 } else { 1326 if (index < 0) { 1327 return; 1328 } 1329 desc = sessionEffects.valueAt(index); 1330 if (--desc->mRefCount == 0) { 1331 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1332 sessionEffects.removeItemsAt(index); 1333 if (sessionEffects.isEmpty()) { 1334 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1335 sessionId); 1336 mSuspendedSessions.removeItem(sessionId); 1337 } 1338 } 1339 } 1340 if (!sessionEffects.isEmpty()) { 1341 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1342 } 1343} 1344 1345void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1346 bool enabled, 1347 int sessionId) 1348{ 1349 Mutex::Autolock _l(mLock); 1350 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1351} 1352 1353void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1354 bool enabled, 1355 int sessionId) 1356{ 1357 if (mType != RECORD) { 1358 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1359 // another session. This gives the priority to well behaved effect control panels 1360 // and applications not using global effects. 1361 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1362 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1363 } 1364 } 1365 1366 sp<EffectChain> chain = getEffectChain_l(sessionId); 1367 if (chain != 0) { 1368 chain->checkSuspendOnEffectEnabled(effect, enabled); 1369 } 1370} 1371 1372// ---------------------------------------------------------------------------- 1373 1374AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1375 AudioStreamOut* output, 1376 int id, 1377 uint32_t device, 1378 type_t type) 1379 : ThreadBase(audioFlinger, id, device, type), 1380 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1381 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1382{ 1383 snprintf(mName, kNameLength, "AudioOut_%d", id); 1384 1385 readOutputParameters(); 1386 1387 // Assumes constructor is called by AudioFlinger with it's mLock held, 1388 // but it would be safer to explicitly pass these as parameters 1389 mMasterVolume = mAudioFlinger->masterVolume_l(); 1390 mMasterMute = mAudioFlinger->masterMute_l(); 1391 1392 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1393 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1394 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1395 stream = (audio_stream_type_t) (stream + 1)) { 1396 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1397 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1398 // initialized by stream_type_t default constructor 1399 // mStreamTypes[stream].valid = true; 1400 } 1401} 1402 1403AudioFlinger::PlaybackThread::~PlaybackThread() 1404{ 1405 delete [] mMixBuffer; 1406} 1407 1408status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1409{ 1410 dumpInternals(fd, args); 1411 dumpTracks(fd, args); 1412 dumpEffectChains(fd, args); 1413 return NO_ERROR; 1414} 1415 1416status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1417{ 1418 const size_t SIZE = 256; 1419 char buffer[SIZE]; 1420 String8 result; 1421 1422 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1423 result.append(buffer); 1424 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1425 for (size_t i = 0; i < mTracks.size(); ++i) { 1426 sp<Track> track = mTracks[i]; 1427 if (track != 0) { 1428 track->dump(buffer, SIZE); 1429 result.append(buffer); 1430 } 1431 } 1432 1433 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1434 result.append(buffer); 1435 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1436 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1437 wp<Track> wTrack = mActiveTracks[i]; 1438 if (wTrack != 0) { 1439 sp<Track> track = wTrack.promote(); 1440 if (track != 0) { 1441 track->dump(buffer, SIZE); 1442 result.append(buffer); 1443 } 1444 } 1445 } 1446 write(fd, result.string(), result.size()); 1447 return NO_ERROR; 1448} 1449 1450status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1451{ 1452 const size_t SIZE = 256; 1453 char buffer[SIZE]; 1454 String8 result; 1455 1456 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1457 result.append(buffer); 1458 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1459 result.append(buffer); 1460 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1461 result.append(buffer); 1462 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1463 result.append(buffer); 1464 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1465 result.append(buffer); 1466 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1467 result.append(buffer); 1468 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1469 result.append(buffer); 1470 write(fd, result.string(), result.size()); 1471 1472 dumpBase(fd, args); 1473 1474 return NO_ERROR; 1475} 1476 1477// Thread virtuals 1478status_t AudioFlinger::PlaybackThread::readyToRun() 1479{ 1480 status_t status = initCheck(); 1481 if (status == NO_ERROR) { 1482 ALOGI("AudioFlinger's thread %p ready to run", this); 1483 } else { 1484 ALOGE("No working audio driver found."); 1485 } 1486 return status; 1487} 1488 1489void AudioFlinger::PlaybackThread::onFirstRef() 1490{ 1491 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1492} 1493 1494// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1495sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1496 const sp<AudioFlinger::Client>& client, 1497 audio_stream_type_t streamType, 1498 uint32_t sampleRate, 1499 audio_format_t format, 1500 uint32_t channelMask, 1501 int frameCount, 1502 const sp<IMemory>& sharedBuffer, 1503 int sessionId, 1504 status_t *status) 1505{ 1506 sp<Track> track; 1507 status_t lStatus; 1508 1509 if (mType == DIRECT) { 1510 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1511 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1512 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1513 "for output %p with format %d", 1514 sampleRate, format, channelMask, mOutput, mFormat); 1515 lStatus = BAD_VALUE; 1516 goto Exit; 1517 } 1518 } 1519 } else { 1520 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1521 if (sampleRate > mSampleRate*2) { 1522 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1523 lStatus = BAD_VALUE; 1524 goto Exit; 1525 } 1526 } 1527 1528 lStatus = initCheck(); 1529 if (lStatus != NO_ERROR) { 1530 ALOGE("Audio driver not initialized."); 1531 goto Exit; 1532 } 1533 1534 { // scope for mLock 1535 Mutex::Autolock _l(mLock); 1536 1537 // all tracks in same audio session must share the same routing strategy otherwise 1538 // conflicts will happen when tracks are moved from one output to another by audio policy 1539 // manager 1540 uint32_t strategy = 1541 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1542 for (size_t i = 0; i < mTracks.size(); ++i) { 1543 sp<Track> t = mTracks[i]; 1544 if (t != 0) { 1545 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1546 if (sessionId == t->sessionId() && strategy != actual) { 1547 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1548 strategy, actual); 1549 lStatus = BAD_VALUE; 1550 goto Exit; 1551 } 1552 } 1553 } 1554 1555 track = new Track(this, client, streamType, sampleRate, format, 1556 channelMask, frameCount, sharedBuffer, sessionId); 1557 if (track->getCblk() == NULL || track->name() < 0) { 1558 lStatus = NO_MEMORY; 1559 goto Exit; 1560 } 1561 mTracks.add(track); 1562 1563 sp<EffectChain> chain = getEffectChain_l(sessionId); 1564 if (chain != 0) { 1565 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1566 track->setMainBuffer(chain->inBuffer()); 1567 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1568 chain->incTrackCnt(); 1569 } 1570 1571 // invalidate track immediately if the stream type was moved to another thread since 1572 // createTrack() was called by the client process. 1573 if (!mStreamTypes[streamType].valid) { 1574 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1575 this, streamType); 1576 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1577 } 1578 } 1579 lStatus = NO_ERROR; 1580 1581Exit: 1582 if(status) { 1583 *status = lStatus; 1584 } 1585 return track; 1586} 1587 1588uint32_t AudioFlinger::PlaybackThread::latency() const 1589{ 1590 Mutex::Autolock _l(mLock); 1591 if (initCheck() == NO_ERROR) { 1592 return mOutput->stream->get_latency(mOutput->stream); 1593 } else { 1594 return 0; 1595 } 1596} 1597 1598status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1599{ 1600 mMasterVolume = value; 1601 return NO_ERROR; 1602} 1603 1604status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1605{ 1606 mMasterMute = muted; 1607 return NO_ERROR; 1608} 1609 1610float AudioFlinger::PlaybackThread::masterVolume() const 1611{ 1612 return mMasterVolume; 1613} 1614 1615bool AudioFlinger::PlaybackThread::masterMute() const 1616{ 1617 return mMasterMute; 1618} 1619 1620status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1621{ 1622 mStreamTypes[stream].volume = value; 1623 return NO_ERROR; 1624} 1625 1626status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1627{ 1628 mStreamTypes[stream].mute = muted; 1629 return NO_ERROR; 1630} 1631 1632float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1633{ 1634 return mStreamTypes[stream].volume; 1635} 1636 1637bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1638{ 1639 return mStreamTypes[stream].mute; 1640} 1641 1642// addTrack_l() must be called with ThreadBase::mLock held 1643status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1644{ 1645 status_t status = ALREADY_EXISTS; 1646 1647 // set retry count for buffer fill 1648 track->mRetryCount = kMaxTrackStartupRetries; 1649 if (mActiveTracks.indexOf(track) < 0) { 1650 // the track is newly added, make sure it fills up all its 1651 // buffers before playing. This is to ensure the client will 1652 // effectively get the latency it requested. 1653 track->mFillingUpStatus = Track::FS_FILLING; 1654 track->mResetDone = false; 1655 mActiveTracks.add(track); 1656 if (track->mainBuffer() != mMixBuffer) { 1657 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1658 if (chain != 0) { 1659 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1660 chain->incActiveTrackCnt(); 1661 } 1662 } 1663 1664 status = NO_ERROR; 1665 } 1666 1667 ALOGV("mWaitWorkCV.broadcast"); 1668 mWaitWorkCV.broadcast(); 1669 1670 return status; 1671} 1672 1673// destroyTrack_l() must be called with ThreadBase::mLock held 1674void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1675{ 1676 track->mState = TrackBase::TERMINATED; 1677 if (mActiveTracks.indexOf(track) < 0) { 1678 removeTrack_l(track); 1679 } 1680} 1681 1682void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1683{ 1684 mTracks.remove(track); 1685 deleteTrackName_l(track->name()); 1686 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1687 if (chain != 0) { 1688 chain->decTrackCnt(); 1689 } 1690} 1691 1692String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1693{ 1694 String8 out_s8 = String8(""); 1695 char *s; 1696 1697 Mutex::Autolock _l(mLock); 1698 if (initCheck() != NO_ERROR) { 1699 return out_s8; 1700 } 1701 1702 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1703 out_s8 = String8(s); 1704 free(s); 1705 return out_s8; 1706} 1707 1708// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1709void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1710 AudioSystem::OutputDescriptor desc; 1711 void *param2 = 0; 1712 1713 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1714 1715 switch (event) { 1716 case AudioSystem::OUTPUT_OPENED: 1717 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1718 desc.channels = mChannelMask; 1719 desc.samplingRate = mSampleRate; 1720 desc.format = mFormat; 1721 desc.frameCount = mFrameCount; 1722 desc.latency = latency(); 1723 param2 = &desc; 1724 break; 1725 1726 case AudioSystem::STREAM_CONFIG_CHANGED: 1727 param2 = ¶m; 1728 case AudioSystem::OUTPUT_CLOSED: 1729 default: 1730 break; 1731 } 1732 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1733} 1734 1735void AudioFlinger::PlaybackThread::readOutputParameters() 1736{ 1737 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1738 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1739 mChannelCount = (uint16_t)popcount(mChannelMask); 1740 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1741 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1742 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1743 1744 // FIXME - Current mixer implementation only supports stereo output: Always 1745 // Allocate a stereo buffer even if HW output is mono. 1746 delete[] mMixBuffer; 1747 mMixBuffer = new int16_t[mFrameCount * 2]; 1748 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1749 1750 // force reconfiguration of effect chains and engines to take new buffer size and audio 1751 // parameters into account 1752 // Note that mLock is not held when readOutputParameters() is called from the constructor 1753 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1754 // matter. 1755 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1756 Vector< sp<EffectChain> > effectChains = mEffectChains; 1757 for (size_t i = 0; i < effectChains.size(); i ++) { 1758 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1759 } 1760} 1761 1762status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1763{ 1764 if (halFrames == 0 || dspFrames == 0) { 1765 return BAD_VALUE; 1766 } 1767 Mutex::Autolock _l(mLock); 1768 if (initCheck() != NO_ERROR) { 1769 return INVALID_OPERATION; 1770 } 1771 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1772 1773 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1774} 1775 1776uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1777{ 1778 Mutex::Autolock _l(mLock); 1779 uint32_t result = 0; 1780 if (getEffectChain_l(sessionId) != 0) { 1781 result = EFFECT_SESSION; 1782 } 1783 1784 for (size_t i = 0; i < mTracks.size(); ++i) { 1785 sp<Track> track = mTracks[i]; 1786 if (sessionId == track->sessionId() && 1787 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1788 result |= TRACK_SESSION; 1789 break; 1790 } 1791 } 1792 1793 return result; 1794} 1795 1796uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1797{ 1798 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1799 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1800 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1801 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1802 } 1803 for (size_t i = 0; i < mTracks.size(); i++) { 1804 sp<Track> track = mTracks[i]; 1805 if (sessionId == track->sessionId() && 1806 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1807 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1808 } 1809 } 1810 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1811} 1812 1813 1814AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1815{ 1816 Mutex::Autolock _l(mLock); 1817 return mOutput; 1818} 1819 1820AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1821{ 1822 Mutex::Autolock _l(mLock); 1823 AudioStreamOut *output = mOutput; 1824 mOutput = NULL; 1825 return output; 1826} 1827 1828// this method must always be called either with ThreadBase mLock held or inside the thread loop 1829audio_stream_t* AudioFlinger::PlaybackThread::stream() 1830{ 1831 if (mOutput == NULL) { 1832 return NULL; 1833 } 1834 return &mOutput->stream->common; 1835} 1836 1837uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1838{ 1839 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1840 // decoding and transfer time. So sleeping for half of the latency would likely cause 1841 // underruns 1842 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1843 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1844 } else { 1845 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1846 } 1847} 1848 1849// ---------------------------------------------------------------------------- 1850 1851AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1852 int id, uint32_t device, type_t type) 1853 : PlaybackThread(audioFlinger, output, id, device, type), 1854 mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE) 1855{ 1856 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1857 1858 // FIXME - Current mixer implementation only supports stereo output 1859 if (mChannelCount == 1) { 1860 ALOGE("Invalid audio hardware channel count"); 1861 } 1862} 1863 1864AudioFlinger::MixerThread::~MixerThread() 1865{ 1866 delete mAudioMixer; 1867} 1868 1869bool AudioFlinger::MixerThread::threadLoop() 1870{ 1871 Vector< sp<Track> > tracksToRemove; 1872 mixer_state mixerStatus = MIXER_IDLE; 1873 nsecs_t standbyTime = systemTime(); 1874 size_t mixBufferSize = mFrameCount * mFrameSize; 1875 // FIXME: Relaxed timing because of a certain device that can't meet latency 1876 // Should be reduced to 2x after the vendor fixes the driver issue 1877 // increase threshold again due to low power audio mode. The way this warning threshold is 1878 // calculated and its usefulness should be reconsidered anyway. 1879 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1880 nsecs_t lastWarning = 0; 1881 bool longStandbyExit = false; 1882 uint32_t activeSleepTime = activeSleepTimeUs(); 1883 uint32_t idleSleepTime = idleSleepTimeUs(); 1884 uint32_t sleepTime = idleSleepTime; 1885 uint32_t sleepTimeShift = 0; 1886 Vector< sp<EffectChain> > effectChains; 1887#ifdef DEBUG_CPU_USAGE 1888 ThreadCpuUsage cpu; 1889 const CentralTendencyStatistics& stats = cpu.statistics(); 1890#endif 1891 1892 acquireWakeLock(); 1893 1894 while (!exitPending()) 1895 { 1896#ifdef DEBUG_CPU_USAGE 1897 cpu.sampleAndEnable(); 1898 unsigned n = stats.n(); 1899 // cpu.elapsed() is expensive, so don't call it every loop 1900 if ((n & 127) == 1) { 1901 long long elapsed = cpu.elapsed(); 1902 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1903 double perLoop = elapsed / (double) n; 1904 double perLoop100 = perLoop * 0.01; 1905 double mean = stats.mean(); 1906 double stddev = stats.stddev(); 1907 double minimum = stats.minimum(); 1908 double maximum = stats.maximum(); 1909 cpu.resetStatistics(); 1910 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", 1911 elapsed * .000000001, n, perLoop * .000001, 1912 mean * .001, 1913 stddev * .001, 1914 minimum * .001, 1915 maximum * .001, 1916 mean / perLoop100, 1917 stddev / perLoop100, 1918 minimum / perLoop100, 1919 maximum / perLoop100); 1920 } 1921 } 1922#endif 1923 processConfigEvents(); 1924 1925 mixerStatus = MIXER_IDLE; 1926 { // scope for mLock 1927 1928 Mutex::Autolock _l(mLock); 1929 1930 if (checkForNewParameters_l()) { 1931 mixBufferSize = mFrameCount * mFrameSize; 1932 // FIXME: Relaxed timing because of a certain device that can't meet latency 1933 // Should be reduced to 2x after the vendor fixes the driver issue 1934 // increase threshold again due to low power audio mode. The way this warning 1935 // threshold is calculated and its usefulness should be reconsidered anyway. 1936 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1937 activeSleepTime = activeSleepTimeUs(); 1938 idleSleepTime = idleSleepTimeUs(); 1939 } 1940 1941 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1942 1943 // put audio hardware into standby after short delay 1944 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1945 mSuspended)) { 1946 if (!mStandby) { 1947 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1948 mOutput->stream->common.standby(&mOutput->stream->common); 1949 mStandby = true; 1950 mBytesWritten = 0; 1951 } 1952 1953 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1954 // we're about to wait, flush the binder command buffer 1955 IPCThreadState::self()->flushCommands(); 1956 1957 if (exitPending()) break; 1958 1959 releaseWakeLock_l(); 1960 // wait until we have something to do... 1961 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1962 mWaitWorkCV.wait(mLock); 1963 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1964 acquireWakeLock_l(); 1965 1966 mPrevMixerStatus = MIXER_IDLE; 1967 if (!mMasterMute) { 1968 char value[PROPERTY_VALUE_MAX]; 1969 property_get("ro.audio.silent", value, "0"); 1970 if (atoi(value)) { 1971 ALOGD("Silence is golden"); 1972 setMasterMute(true); 1973 } 1974 } 1975 1976 standbyTime = systemTime() + kStandbyTimeInNsecs; 1977 sleepTime = idleSleepTime; 1978 sleepTimeShift = 0; 1979 continue; 1980 } 1981 } 1982 1983 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1984 1985 // prevent any changes in effect chain list and in each effect chain 1986 // during mixing and effect process as the audio buffers could be deleted 1987 // or modified if an effect is created or deleted 1988 lockEffectChains_l(effectChains); 1989 } 1990 1991 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1992 // mix buffers... 1993 mAudioMixer->process(); 1994 // increase sleep time progressively when application underrun condition clears. 1995 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1996 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1997 // such that we would underrun the audio HAL. 1998 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 1999 sleepTimeShift--; 2000 } 2001 sleepTime = 0; 2002 standbyTime = systemTime() + kStandbyTimeInNsecs; 2003 //TODO: delay standby when effects have a tail 2004 } else { 2005 // If no tracks are ready, sleep once for the duration of an output 2006 // buffer size, then write 0s to the output 2007 if (sleepTime == 0) { 2008 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2009 sleepTime = activeSleepTime >> sleepTimeShift; 2010 if (sleepTime < kMinThreadSleepTimeUs) { 2011 sleepTime = kMinThreadSleepTimeUs; 2012 } 2013 // reduce sleep time in case of consecutive application underruns to avoid 2014 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2015 // duration we would end up writing less data than needed by the audio HAL if 2016 // the condition persists. 2017 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2018 sleepTimeShift++; 2019 } 2020 } else { 2021 sleepTime = idleSleepTime; 2022 } 2023 } else if (mBytesWritten != 0 || 2024 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2025 memset (mMixBuffer, 0, mixBufferSize); 2026 sleepTime = 0; 2027 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2028 } 2029 // TODO add standby time extension fct of effect tail 2030 } 2031 2032 if (mSuspended) { 2033 sleepTime = suspendSleepTimeUs(); 2034 } 2035 // sleepTime == 0 means we must write to audio hardware 2036 if (sleepTime == 0) { 2037 for (size_t i = 0; i < effectChains.size(); i ++) { 2038 effectChains[i]->process_l(); 2039 } 2040 // enable changes in effect chain 2041 unlockEffectChains(effectChains); 2042 mLastWriteTime = systemTime(); 2043 mInWrite = true; 2044 mBytesWritten += mixBufferSize; 2045 2046 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2047 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2048 mNumWrites++; 2049 mInWrite = false; 2050 nsecs_t now = systemTime(); 2051 nsecs_t delta = now - mLastWriteTime; 2052 if (!mStandby && delta > maxPeriod) { 2053 mNumDelayedWrites++; 2054 if ((now - lastWarning) > kWarningThrottleNs) { 2055 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2056 ns2ms(delta), mNumDelayedWrites, this); 2057 lastWarning = now; 2058 } 2059 if (mStandby) { 2060 longStandbyExit = true; 2061 } 2062 } 2063 mStandby = false; 2064 } else { 2065 // enable changes in effect chain 2066 unlockEffectChains(effectChains); 2067 usleep(sleepTime); 2068 } 2069 2070 // finally let go of all our tracks, without the lock held 2071 // since we can't guarantee the destructors won't acquire that 2072 // same lock. 2073 tracksToRemove.clear(); 2074 2075 // Effect chains will be actually deleted here if they were removed from 2076 // mEffectChains list during mixing or effects processing 2077 effectChains.clear(); 2078 } 2079 2080 if (!mStandby) { 2081 mOutput->stream->common.standby(&mOutput->stream->common); 2082 } 2083 2084 releaseWakeLock(); 2085 2086 ALOGV("MixerThread %p exiting", this); 2087 return false; 2088} 2089 2090// prepareTracks_l() must be called with ThreadBase::mLock held 2091AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2092 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2093{ 2094 2095 mixer_state mixerStatus = MIXER_IDLE; 2096 // find out which tracks need to be processed 2097 size_t count = activeTracks.size(); 2098 size_t mixedTracks = 0; 2099 size_t tracksWithEffect = 0; 2100 2101 float masterVolume = mMasterVolume; 2102 bool masterMute = mMasterMute; 2103 2104 if (masterMute) { 2105 masterVolume = 0; 2106 } 2107 // Delegate master volume control to effect in output mix effect chain if needed 2108 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2109 if (chain != 0) { 2110 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2111 chain->setVolume_l(&v, &v); 2112 masterVolume = (float)((v + (1 << 23)) >> 24); 2113 chain.clear(); 2114 } 2115 2116 for (size_t i=0 ; i<count ; i++) { 2117 sp<Track> t = activeTracks[i].promote(); 2118 if (t == 0) continue; 2119 2120 // this const just means the local variable doesn't change 2121 Track* const track = t.get(); 2122 audio_track_cblk_t* cblk = track->cblk(); 2123 2124 // The first time a track is added we wait 2125 // for all its buffers to be filled before processing it 2126 int name = track->name(); 2127 // make sure that we have enough frames to mix one full buffer. 2128 // enforce this condition only once to enable draining the buffer in case the client 2129 // app does not call stop() and relies on underrun to stop: 2130 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2131 // during last round 2132 uint32_t minFrames = 1; 2133 if (!track->isStopped() && !track->isPausing() && 2134 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2135 if (t->sampleRate() == (int)mSampleRate) { 2136 minFrames = mFrameCount; 2137 } else { 2138 // +1 for rounding and +1 for additional sample needed for interpolation 2139 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2140 // add frames already consumed but not yet released by the resampler 2141 // because cblk->framesReady() will include these frames 2142 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2143 // the minimum track buffer size is normally twice the number of frames necessary 2144 // to fill one buffer and the resampler should not leave more than one buffer worth 2145 // of unreleased frames after each pass, but just in case... 2146 ALOG_ASSERT(minFrames <= cblk->frameCount); 2147 } 2148 } 2149 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2150 !track->isPaused() && !track->isTerminated()) 2151 { 2152 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2153 2154 mixedTracks++; 2155 2156 // track->mainBuffer() != mMixBuffer means there is an effect chain 2157 // connected to the track 2158 chain.clear(); 2159 if (track->mainBuffer() != mMixBuffer) { 2160 chain = getEffectChain_l(track->sessionId()); 2161 // Delegate volume control to effect in track effect chain if needed 2162 if (chain != 0) { 2163 tracksWithEffect++; 2164 } else { 2165 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2166 name, track->sessionId()); 2167 } 2168 } 2169 2170 2171 int param = AudioMixer::VOLUME; 2172 if (track->mFillingUpStatus == Track::FS_FILLED) { 2173 // no ramp for the first volume setting 2174 track->mFillingUpStatus = Track::FS_ACTIVE; 2175 if (track->mState == TrackBase::RESUMING) { 2176 track->mState = TrackBase::ACTIVE; 2177 param = AudioMixer::RAMP_VOLUME; 2178 } 2179 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2180 } else if (cblk->server != 0) { 2181 // If the track is stopped before the first frame was mixed, 2182 // do not apply ramp 2183 param = AudioMixer::RAMP_VOLUME; 2184 } 2185 2186 // compute volume for this track 2187 uint32_t vl, vr, va; 2188 if (track->isMuted() || track->isPausing() || 2189 mStreamTypes[track->type()].mute) { 2190 vl = vr = va = 0; 2191 if (track->isPausing()) { 2192 track->setPaused(); 2193 } 2194 } else { 2195 2196 // read original volumes with volume control 2197 float typeVolume = mStreamTypes[track->type()].volume; 2198 float v = masterVolume * typeVolume; 2199 uint32_t vlr = cblk->getVolumeLR(); 2200 vl = vlr & 0xFFFF; 2201 vr = vlr >> 16; 2202 // track volumes come from shared memory, so can't be trusted and must be clamped 2203 if (vl > MAX_GAIN_INT) { 2204 ALOGV("Track left volume out of range: %04X", vl); 2205 vl = MAX_GAIN_INT; 2206 } 2207 if (vr > MAX_GAIN_INT) { 2208 ALOGV("Track right volume out of range: %04X", vr); 2209 vr = MAX_GAIN_INT; 2210 } 2211 // now apply the master volume and stream type volume 2212 vl = (uint32_t)(v * vl) << 12; 2213 vr = (uint32_t)(v * vr) << 12; 2214 // assuming master volume and stream type volume each go up to 1.0, 2215 // vl and vr are now in 8.24 format 2216 2217 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2218 // send level comes from shared memory and so may be corrupt 2219 if (sendLevel >= MAX_GAIN_INT) { 2220 ALOGV("Track send level out of range: %04X", sendLevel); 2221 sendLevel = MAX_GAIN_INT; 2222 } 2223 va = (uint32_t)(v * sendLevel); 2224 } 2225 // Delegate volume control to effect in track effect chain if needed 2226 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2227 // Do not ramp volume if volume is controlled by effect 2228 param = AudioMixer::VOLUME; 2229 track->mHasVolumeController = true; 2230 } else { 2231 // force no volume ramp when volume controller was just disabled or removed 2232 // from effect chain to avoid volume spike 2233 if (track->mHasVolumeController) { 2234 param = AudioMixer::VOLUME; 2235 } 2236 track->mHasVolumeController = false; 2237 } 2238 2239 // Convert volumes from 8.24 to 4.12 format 2240 int16_t left, right, aux; 2241 // This additional clamping is needed in case chain->setVolume_l() overshot 2242 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2243 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2244 left = int16_t(v_clamped); 2245 v_clamped = (vr + (1 << 11)) >> 12; 2246 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2247 right = int16_t(v_clamped); 2248 2249 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2250 aux = int16_t(va); 2251 2252 // XXX: these things DON'T need to be done each time 2253 mAudioMixer->setBufferProvider(name, track); 2254 mAudioMixer->enable(name); 2255 2256 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2257 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2258 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2259 mAudioMixer->setParameter( 2260 name, 2261 AudioMixer::TRACK, 2262 AudioMixer::FORMAT, (void *)track->format()); 2263 mAudioMixer->setParameter( 2264 name, 2265 AudioMixer::TRACK, 2266 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2267 mAudioMixer->setParameter( 2268 name, 2269 AudioMixer::RESAMPLE, 2270 AudioMixer::SAMPLE_RATE, 2271 (void *)(cblk->sampleRate)); 2272 mAudioMixer->setParameter( 2273 name, 2274 AudioMixer::TRACK, 2275 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2276 mAudioMixer->setParameter( 2277 name, 2278 AudioMixer::TRACK, 2279 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2280 2281 // reset retry count 2282 track->mRetryCount = kMaxTrackRetries; 2283 // If one track is ready, set the mixer ready if: 2284 // - the mixer was not ready during previous round OR 2285 // - no other track is not ready 2286 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2287 mixerStatus != MIXER_TRACKS_ENABLED) { 2288 mixerStatus = MIXER_TRACKS_READY; 2289 } 2290 } else { 2291 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2292 if (track->isStopped()) { 2293 track->reset(); 2294 } 2295 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2296 // We have consumed all the buffers of this track. 2297 // Remove it from the list of active tracks. 2298 tracksToRemove->add(track); 2299 } else { 2300 // No buffers for this track. Give it a few chances to 2301 // fill a buffer, then remove it from active list. 2302 if (--(track->mRetryCount) <= 0) { 2303 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2304 tracksToRemove->add(track); 2305 // indicate to client process that the track was disabled because of underrun 2306 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2307 // If one track is not ready, mark the mixer also not ready if: 2308 // - the mixer was ready during previous round OR 2309 // - no other track is ready 2310 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2311 mixerStatus != MIXER_TRACKS_READY) { 2312 mixerStatus = MIXER_TRACKS_ENABLED; 2313 } 2314 } 2315 mAudioMixer->disable(name); 2316 } 2317 } 2318 2319 // remove all the tracks that need to be... 2320 count = tracksToRemove->size(); 2321 if (CC_UNLIKELY(count)) { 2322 for (size_t i=0 ; i<count ; i++) { 2323 const sp<Track>& track = tracksToRemove->itemAt(i); 2324 mActiveTracks.remove(track); 2325 if (track->mainBuffer() != mMixBuffer) { 2326 chain = getEffectChain_l(track->sessionId()); 2327 if (chain != 0) { 2328 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2329 chain->decActiveTrackCnt(); 2330 } 2331 } 2332 if (track->isTerminated()) { 2333 removeTrack_l(track); 2334 } 2335 } 2336 } 2337 2338 // mix buffer must be cleared if all tracks are connected to an 2339 // effect chain as in this case the mixer will not write to 2340 // mix buffer and track effects will accumulate into it 2341 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2342 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2343 } 2344 2345 mPrevMixerStatus = mixerStatus; 2346 return mixerStatus; 2347} 2348 2349void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2350{ 2351 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2352 this, streamType, mTracks.size()); 2353 Mutex::Autolock _l(mLock); 2354 2355 size_t size = mTracks.size(); 2356 for (size_t i = 0; i < size; i++) { 2357 sp<Track> t = mTracks[i]; 2358 if (t->type() == streamType) { 2359 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2360 t->mCblk->cv.signal(); 2361 } 2362 } 2363} 2364 2365void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2366{ 2367 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2368 this, streamType, valid); 2369 Mutex::Autolock _l(mLock); 2370 2371 mStreamTypes[streamType].valid = valid; 2372} 2373 2374// getTrackName_l() must be called with ThreadBase::mLock held 2375int AudioFlinger::MixerThread::getTrackName_l() 2376{ 2377 return mAudioMixer->getTrackName(); 2378} 2379 2380// deleteTrackName_l() must be called with ThreadBase::mLock held 2381void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2382{ 2383 ALOGV("remove track (%d) and delete from mixer", name); 2384 mAudioMixer->deleteTrackName(name); 2385} 2386 2387// checkForNewParameters_l() must be called with ThreadBase::mLock held 2388bool AudioFlinger::MixerThread::checkForNewParameters_l() 2389{ 2390 bool reconfig = false; 2391 2392 while (!mNewParameters.isEmpty()) { 2393 status_t status = NO_ERROR; 2394 String8 keyValuePair = mNewParameters[0]; 2395 AudioParameter param = AudioParameter(keyValuePair); 2396 int value; 2397 2398 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2399 reconfig = true; 2400 } 2401 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2402 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2403 status = BAD_VALUE; 2404 } else { 2405 reconfig = true; 2406 } 2407 } 2408 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2409 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2410 status = BAD_VALUE; 2411 } else { 2412 reconfig = true; 2413 } 2414 } 2415 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2416 // do not accept frame count changes if tracks are open as the track buffer 2417 // size depends on frame count and correct behavior would not be guaranteed 2418 // if frame count is changed after track creation 2419 if (!mTracks.isEmpty()) { 2420 status = INVALID_OPERATION; 2421 } else { 2422 reconfig = true; 2423 } 2424 } 2425 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2426 // when changing the audio output device, call addBatteryData to notify 2427 // the change 2428 if ((int)mDevice != value) { 2429 uint32_t params = 0; 2430 // check whether speaker is on 2431 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2432 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2433 } 2434 2435 int deviceWithoutSpeaker 2436 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2437 // check if any other device (except speaker) is on 2438 if (value & deviceWithoutSpeaker ) { 2439 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2440 } 2441 2442 if (params != 0) { 2443 addBatteryData(params); 2444 } 2445 } 2446 2447 // forward device change to effects that have requested to be 2448 // aware of attached audio device. 2449 mDevice = (uint32_t)value; 2450 for (size_t i = 0; i < mEffectChains.size(); i++) { 2451 mEffectChains[i]->setDevice_l(mDevice); 2452 } 2453 } 2454 2455 if (status == NO_ERROR) { 2456 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2457 keyValuePair.string()); 2458 if (!mStandby && status == INVALID_OPERATION) { 2459 mOutput->stream->common.standby(&mOutput->stream->common); 2460 mStandby = true; 2461 mBytesWritten = 0; 2462 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2463 keyValuePair.string()); 2464 } 2465 if (status == NO_ERROR && reconfig) { 2466 delete mAudioMixer; 2467 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2468 mAudioMixer = NULL; 2469 readOutputParameters(); 2470 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2471 for (size_t i = 0; i < mTracks.size() ; i++) { 2472 int name = getTrackName_l(); 2473 if (name < 0) break; 2474 mTracks[i]->mName = name; 2475 // limit track sample rate to 2 x new output sample rate 2476 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2477 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2478 } 2479 } 2480 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2481 } 2482 } 2483 2484 mNewParameters.removeAt(0); 2485 2486 mParamStatus = status; 2487 mParamCond.signal(); 2488 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2489 // already timed out waiting for the status and will never signal the condition. 2490 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2491 } 2492 return reconfig; 2493} 2494 2495status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2496{ 2497 const size_t SIZE = 256; 2498 char buffer[SIZE]; 2499 String8 result; 2500 2501 PlaybackThread::dumpInternals(fd, args); 2502 2503 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2504 result.append(buffer); 2505 write(fd, result.string(), result.size()); 2506 return NO_ERROR; 2507} 2508 2509uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2510{ 2511 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2512} 2513 2514uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2515{ 2516 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2517} 2518 2519// ---------------------------------------------------------------------------- 2520AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2521 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2522{ 2523} 2524 2525AudioFlinger::DirectOutputThread::~DirectOutputThread() 2526{ 2527} 2528 2529static inline 2530int32_t mul(int16_t in, int16_t v) 2531{ 2532#if defined(__arm__) && !defined(__thumb__) 2533 int32_t out; 2534 asm( "smulbb %[out], %[in], %[v] \n" 2535 : [out]"=r"(out) 2536 : [in]"%r"(in), [v]"r"(v) 2537 : ); 2538 return out; 2539#else 2540 return in * int32_t(v); 2541#endif 2542} 2543 2544void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2545{ 2546 // Do not apply volume on compressed audio 2547 if (!audio_is_linear_pcm(mFormat)) { 2548 return; 2549 } 2550 2551 // convert to signed 16 bit before volume calculation 2552 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2553 size_t count = mFrameCount * mChannelCount; 2554 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2555 int16_t *dst = mMixBuffer + count-1; 2556 while(count--) { 2557 *dst-- = (int16_t)(*src--^0x80) << 8; 2558 } 2559 } 2560 2561 size_t frameCount = mFrameCount; 2562 int16_t *out = mMixBuffer; 2563 if (ramp) { 2564 if (mChannelCount == 1) { 2565 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2566 int32_t vlInc = d / (int32_t)frameCount; 2567 int32_t vl = ((int32_t)mLeftVolShort << 16); 2568 do { 2569 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2570 out++; 2571 vl += vlInc; 2572 } while (--frameCount); 2573 2574 } else { 2575 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2576 int32_t vlInc = d / (int32_t)frameCount; 2577 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2578 int32_t vrInc = d / (int32_t)frameCount; 2579 int32_t vl = ((int32_t)mLeftVolShort << 16); 2580 int32_t vr = ((int32_t)mRightVolShort << 16); 2581 do { 2582 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2583 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2584 out += 2; 2585 vl += vlInc; 2586 vr += vrInc; 2587 } while (--frameCount); 2588 } 2589 } else { 2590 if (mChannelCount == 1) { 2591 do { 2592 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2593 out++; 2594 } while (--frameCount); 2595 } else { 2596 do { 2597 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2598 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2599 out += 2; 2600 } while (--frameCount); 2601 } 2602 } 2603 2604 // convert back to unsigned 8 bit after volume calculation 2605 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2606 size_t count = mFrameCount * mChannelCount; 2607 int16_t *src = mMixBuffer; 2608 uint8_t *dst = (uint8_t *)mMixBuffer; 2609 while(count--) { 2610 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2611 } 2612 } 2613 2614 mLeftVolShort = leftVol; 2615 mRightVolShort = rightVol; 2616} 2617 2618bool AudioFlinger::DirectOutputThread::threadLoop() 2619{ 2620 mixer_state mixerStatus = MIXER_IDLE; 2621 sp<Track> trackToRemove; 2622 sp<Track> activeTrack; 2623 nsecs_t standbyTime = systemTime(); 2624 int8_t *curBuf; 2625 size_t mixBufferSize = mFrameCount*mFrameSize; 2626 uint32_t activeSleepTime = activeSleepTimeUs(); 2627 uint32_t idleSleepTime = idleSleepTimeUs(); 2628 uint32_t sleepTime = idleSleepTime; 2629 // use shorter standby delay as on normal output to release 2630 // hardware resources as soon as possible 2631 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2632 2633 acquireWakeLock(); 2634 2635 while (!exitPending()) 2636 { 2637 bool rampVolume; 2638 uint16_t leftVol; 2639 uint16_t rightVol; 2640 Vector< sp<EffectChain> > effectChains; 2641 2642 processConfigEvents(); 2643 2644 mixerStatus = MIXER_IDLE; 2645 2646 { // scope for the mLock 2647 2648 Mutex::Autolock _l(mLock); 2649 2650 if (checkForNewParameters_l()) { 2651 mixBufferSize = mFrameCount*mFrameSize; 2652 activeSleepTime = activeSleepTimeUs(); 2653 idleSleepTime = idleSleepTimeUs(); 2654 standbyDelay = microseconds(activeSleepTime*2); 2655 } 2656 2657 // put audio hardware into standby after short delay 2658 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2659 mSuspended)) { 2660 // wait until we have something to do... 2661 if (!mStandby) { 2662 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2663 mOutput->stream->common.standby(&mOutput->stream->common); 2664 mStandby = true; 2665 mBytesWritten = 0; 2666 } 2667 2668 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2669 // we're about to wait, flush the binder command buffer 2670 IPCThreadState::self()->flushCommands(); 2671 2672 if (exitPending()) break; 2673 2674 releaseWakeLock_l(); 2675 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2676 mWaitWorkCV.wait(mLock); 2677 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2678 acquireWakeLock_l(); 2679 2680 if (!mMasterMute) { 2681 char value[PROPERTY_VALUE_MAX]; 2682 property_get("ro.audio.silent", value, "0"); 2683 if (atoi(value)) { 2684 ALOGD("Silence is golden"); 2685 setMasterMute(true); 2686 } 2687 } 2688 2689 standbyTime = systemTime() + standbyDelay; 2690 sleepTime = idleSleepTime; 2691 continue; 2692 } 2693 } 2694 2695 effectChains = mEffectChains; 2696 2697 // find out which tracks need to be processed 2698 if (mActiveTracks.size() != 0) { 2699 sp<Track> t = mActiveTracks[0].promote(); 2700 if (t == 0) continue; 2701 2702 Track* const track = t.get(); 2703 audio_track_cblk_t* cblk = track->cblk(); 2704 2705 // The first time a track is added we wait 2706 // for all its buffers to be filled before processing it 2707 if (cblk->framesReady() && track->isReady() && 2708 !track->isPaused() && !track->isTerminated()) 2709 { 2710 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2711 2712 if (track->mFillingUpStatus == Track::FS_FILLED) { 2713 track->mFillingUpStatus = Track::FS_ACTIVE; 2714 mLeftVolFloat = mRightVolFloat = 0; 2715 mLeftVolShort = mRightVolShort = 0; 2716 if (track->mState == TrackBase::RESUMING) { 2717 track->mState = TrackBase::ACTIVE; 2718 rampVolume = true; 2719 } 2720 } else if (cblk->server != 0) { 2721 // If the track is stopped before the first frame was mixed, 2722 // do not apply ramp 2723 rampVolume = true; 2724 } 2725 // compute volume for this track 2726 float left, right; 2727 if (track->isMuted() || mMasterMute || track->isPausing() || 2728 mStreamTypes[track->type()].mute) { 2729 left = right = 0; 2730 if (track->isPausing()) { 2731 track->setPaused(); 2732 } 2733 } else { 2734 float typeVolume = mStreamTypes[track->type()].volume; 2735 float v = mMasterVolume * typeVolume; 2736 uint32_t vlr = cblk->getVolumeLR(); 2737 float v_clamped = v * (vlr & 0xFFFF); 2738 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2739 left = v_clamped/MAX_GAIN; 2740 v_clamped = v * (vlr >> 16); 2741 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2742 right = v_clamped/MAX_GAIN; 2743 } 2744 2745 if (left != mLeftVolFloat || right != mRightVolFloat) { 2746 mLeftVolFloat = left; 2747 mRightVolFloat = right; 2748 2749 // If audio HAL implements volume control, 2750 // force software volume to nominal value 2751 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2752 left = 1.0f; 2753 right = 1.0f; 2754 } 2755 2756 // Convert volumes from float to 8.24 2757 uint32_t vl = (uint32_t)(left * (1 << 24)); 2758 uint32_t vr = (uint32_t)(right * (1 << 24)); 2759 2760 // Delegate volume control to effect in track effect chain if needed 2761 // only one effect chain can be present on DirectOutputThread, so if 2762 // there is one, the track is connected to it 2763 if (!effectChains.isEmpty()) { 2764 // Do not ramp volume if volume is controlled by effect 2765 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2766 rampVolume = false; 2767 } 2768 } 2769 2770 // Convert volumes from 8.24 to 4.12 format 2771 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2772 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2773 leftVol = (uint16_t)v_clamped; 2774 v_clamped = (vr + (1 << 11)) >> 12; 2775 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2776 rightVol = (uint16_t)v_clamped; 2777 } else { 2778 leftVol = mLeftVolShort; 2779 rightVol = mRightVolShort; 2780 rampVolume = false; 2781 } 2782 2783 // reset retry count 2784 track->mRetryCount = kMaxTrackRetriesDirect; 2785 activeTrack = t; 2786 mixerStatus = MIXER_TRACKS_READY; 2787 } else { 2788 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2789 if (track->isStopped()) { 2790 track->reset(); 2791 } 2792 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2793 // We have consumed all the buffers of this track. 2794 // Remove it from the list of active tracks. 2795 trackToRemove = track; 2796 } else { 2797 // No buffers for this track. Give it a few chances to 2798 // fill a buffer, then remove it from active list. 2799 if (--(track->mRetryCount) <= 0) { 2800 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2801 trackToRemove = track; 2802 } else { 2803 mixerStatus = MIXER_TRACKS_ENABLED; 2804 } 2805 } 2806 } 2807 } 2808 2809 // remove all the tracks that need to be... 2810 if (CC_UNLIKELY(trackToRemove != 0)) { 2811 mActiveTracks.remove(trackToRemove); 2812 if (!effectChains.isEmpty()) { 2813 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2814 trackToRemove->sessionId()); 2815 effectChains[0]->decActiveTrackCnt(); 2816 } 2817 if (trackToRemove->isTerminated()) { 2818 removeTrack_l(trackToRemove); 2819 } 2820 } 2821 2822 lockEffectChains_l(effectChains); 2823 } 2824 2825 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2826 AudioBufferProvider::Buffer buffer; 2827 size_t frameCount = mFrameCount; 2828 curBuf = (int8_t *)mMixBuffer; 2829 // output audio to hardware 2830 while (frameCount) { 2831 buffer.frameCount = frameCount; 2832 activeTrack->getNextBuffer(&buffer); 2833 if (CC_UNLIKELY(buffer.raw == NULL)) { 2834 memset(curBuf, 0, frameCount * mFrameSize); 2835 break; 2836 } 2837 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2838 frameCount -= buffer.frameCount; 2839 curBuf += buffer.frameCount * mFrameSize; 2840 activeTrack->releaseBuffer(&buffer); 2841 } 2842 sleepTime = 0; 2843 standbyTime = systemTime() + standbyDelay; 2844 } else { 2845 if (sleepTime == 0) { 2846 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2847 sleepTime = activeSleepTime; 2848 } else { 2849 sleepTime = idleSleepTime; 2850 } 2851 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2852 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2853 sleepTime = 0; 2854 } 2855 } 2856 2857 if (mSuspended) { 2858 sleepTime = suspendSleepTimeUs(); 2859 } 2860 // sleepTime == 0 means we must write to audio hardware 2861 if (sleepTime == 0) { 2862 if (mixerStatus == MIXER_TRACKS_READY) { 2863 applyVolume(leftVol, rightVol, rampVolume); 2864 } 2865 for (size_t i = 0; i < effectChains.size(); i ++) { 2866 effectChains[i]->process_l(); 2867 } 2868 unlockEffectChains(effectChains); 2869 2870 mLastWriteTime = systemTime(); 2871 mInWrite = true; 2872 mBytesWritten += mixBufferSize; 2873 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2874 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2875 mNumWrites++; 2876 mInWrite = false; 2877 mStandby = false; 2878 } else { 2879 unlockEffectChains(effectChains); 2880 usleep(sleepTime); 2881 } 2882 2883 // finally let go of removed track, without the lock held 2884 // since we can't guarantee the destructors won't acquire that 2885 // same lock. 2886 trackToRemove.clear(); 2887 activeTrack.clear(); 2888 2889 // Effect chains will be actually deleted here if they were removed from 2890 // mEffectChains list during mixing or effects processing 2891 effectChains.clear(); 2892 } 2893 2894 if (!mStandby) { 2895 mOutput->stream->common.standby(&mOutput->stream->common); 2896 } 2897 2898 releaseWakeLock(); 2899 2900 ALOGV("DirectOutputThread %p exiting", this); 2901 return false; 2902} 2903 2904// getTrackName_l() must be called with ThreadBase::mLock held 2905int AudioFlinger::DirectOutputThread::getTrackName_l() 2906{ 2907 return 0; 2908} 2909 2910// deleteTrackName_l() must be called with ThreadBase::mLock held 2911void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2912{ 2913} 2914 2915// checkForNewParameters_l() must be called with ThreadBase::mLock held 2916bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2917{ 2918 bool reconfig = false; 2919 2920 while (!mNewParameters.isEmpty()) { 2921 status_t status = NO_ERROR; 2922 String8 keyValuePair = mNewParameters[0]; 2923 AudioParameter param = AudioParameter(keyValuePair); 2924 int value; 2925 2926 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2927 // do not accept frame count changes if tracks are open as the track buffer 2928 // size depends on frame count and correct behavior would not be garantied 2929 // if frame count is changed after track creation 2930 if (!mTracks.isEmpty()) { 2931 status = INVALID_OPERATION; 2932 } else { 2933 reconfig = true; 2934 } 2935 } 2936 if (status == NO_ERROR) { 2937 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2938 keyValuePair.string()); 2939 if (!mStandby && status == INVALID_OPERATION) { 2940 mOutput->stream->common.standby(&mOutput->stream->common); 2941 mStandby = true; 2942 mBytesWritten = 0; 2943 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2944 keyValuePair.string()); 2945 } 2946 if (status == NO_ERROR && reconfig) { 2947 readOutputParameters(); 2948 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2949 } 2950 } 2951 2952 mNewParameters.removeAt(0); 2953 2954 mParamStatus = status; 2955 mParamCond.signal(); 2956 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2957 // already timed out waiting for the status and will never signal the condition. 2958 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2959 } 2960 return reconfig; 2961} 2962 2963uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2964{ 2965 uint32_t time; 2966 if (audio_is_linear_pcm(mFormat)) { 2967 time = PlaybackThread::activeSleepTimeUs(); 2968 } else { 2969 time = 10000; 2970 } 2971 return time; 2972} 2973 2974uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2975{ 2976 uint32_t time; 2977 if (audio_is_linear_pcm(mFormat)) { 2978 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2979 } else { 2980 time = 10000; 2981 } 2982 return time; 2983} 2984 2985uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2986{ 2987 uint32_t time; 2988 if (audio_is_linear_pcm(mFormat)) { 2989 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2990 } else { 2991 time = 10000; 2992 } 2993 return time; 2994} 2995 2996 2997// ---------------------------------------------------------------------------- 2998 2999AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3000 AudioFlinger::MixerThread* mainThread, int id) 3001 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3002 mWaitTimeMs(UINT_MAX) 3003{ 3004 addOutputTrack(mainThread); 3005} 3006 3007AudioFlinger::DuplicatingThread::~DuplicatingThread() 3008{ 3009 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3010 mOutputTracks[i]->destroy(); 3011 } 3012 mOutputTracks.clear(); 3013} 3014 3015bool AudioFlinger::DuplicatingThread::threadLoop() 3016{ 3017 Vector< sp<Track> > tracksToRemove; 3018 mixer_state mixerStatus = MIXER_IDLE; 3019 nsecs_t standbyTime = systemTime(); 3020 size_t mixBufferSize = mFrameCount*mFrameSize; 3021 SortedVector< sp<OutputTrack> > outputTracks; 3022 uint32_t writeFrames = 0; 3023 uint32_t activeSleepTime = activeSleepTimeUs(); 3024 uint32_t idleSleepTime = idleSleepTimeUs(); 3025 uint32_t sleepTime = idleSleepTime; 3026 Vector< sp<EffectChain> > effectChains; 3027 3028 acquireWakeLock(); 3029 3030 while (!exitPending()) 3031 { 3032 processConfigEvents(); 3033 3034 mixerStatus = MIXER_IDLE; 3035 { // scope for the mLock 3036 3037 Mutex::Autolock _l(mLock); 3038 3039 if (checkForNewParameters_l()) { 3040 mixBufferSize = mFrameCount*mFrameSize; 3041 updateWaitTime(); 3042 activeSleepTime = activeSleepTimeUs(); 3043 idleSleepTime = idleSleepTimeUs(); 3044 } 3045 3046 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3047 3048 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3049 outputTracks.add(mOutputTracks[i]); 3050 } 3051 3052 // put audio hardware into standby after short delay 3053 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3054 mSuspended)) { 3055 if (!mStandby) { 3056 for (size_t i = 0; i < outputTracks.size(); i++) { 3057 outputTracks[i]->stop(); 3058 } 3059 mStandby = true; 3060 mBytesWritten = 0; 3061 } 3062 3063 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3064 // we're about to wait, flush the binder command buffer 3065 IPCThreadState::self()->flushCommands(); 3066 outputTracks.clear(); 3067 3068 if (exitPending()) break; 3069 3070 releaseWakeLock_l(); 3071 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3072 mWaitWorkCV.wait(mLock); 3073 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3074 acquireWakeLock_l(); 3075 3076 mPrevMixerStatus = MIXER_IDLE; 3077 if (!mMasterMute) { 3078 char value[PROPERTY_VALUE_MAX]; 3079 property_get("ro.audio.silent", value, "0"); 3080 if (atoi(value)) { 3081 ALOGD("Silence is golden"); 3082 setMasterMute(true); 3083 } 3084 } 3085 3086 standbyTime = systemTime() + kStandbyTimeInNsecs; 3087 sleepTime = idleSleepTime; 3088 continue; 3089 } 3090 } 3091 3092 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3093 3094 // prevent any changes in effect chain list and in each effect chain 3095 // during mixing and effect process as the audio buffers could be deleted 3096 // or modified if an effect is created or deleted 3097 lockEffectChains_l(effectChains); 3098 } 3099 3100 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3101 // mix buffers... 3102 if (outputsReady(outputTracks)) { 3103 mAudioMixer->process(); 3104 } else { 3105 memset(mMixBuffer, 0, mixBufferSize); 3106 } 3107 sleepTime = 0; 3108 writeFrames = mFrameCount; 3109 } else { 3110 if (sleepTime == 0) { 3111 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3112 sleepTime = activeSleepTime; 3113 } else { 3114 sleepTime = idleSleepTime; 3115 } 3116 } else if (mBytesWritten != 0) { 3117 // flush remaining overflow buffers in output tracks 3118 for (size_t i = 0; i < outputTracks.size(); i++) { 3119 if (outputTracks[i]->isActive()) { 3120 sleepTime = 0; 3121 writeFrames = 0; 3122 memset(mMixBuffer, 0, mixBufferSize); 3123 break; 3124 } 3125 } 3126 } 3127 } 3128 3129 if (mSuspended) { 3130 sleepTime = suspendSleepTimeUs(); 3131 } 3132 // sleepTime == 0 means we must write to audio hardware 3133 if (sleepTime == 0) { 3134 for (size_t i = 0; i < effectChains.size(); i ++) { 3135 effectChains[i]->process_l(); 3136 } 3137 // enable changes in effect chain 3138 unlockEffectChains(effectChains); 3139 3140 standbyTime = systemTime() + kStandbyTimeInNsecs; 3141 for (size_t i = 0; i < outputTracks.size(); i++) { 3142 outputTracks[i]->write(mMixBuffer, writeFrames); 3143 } 3144 mStandby = false; 3145 mBytesWritten += mixBufferSize; 3146 } else { 3147 // enable changes in effect chain 3148 unlockEffectChains(effectChains); 3149 usleep(sleepTime); 3150 } 3151 3152 // finally let go of all our tracks, without the lock held 3153 // since we can't guarantee the destructors won't acquire that 3154 // same lock. 3155 tracksToRemove.clear(); 3156 outputTracks.clear(); 3157 3158 // Effect chains will be actually deleted here if they were removed from 3159 // mEffectChains list during mixing or effects processing 3160 effectChains.clear(); 3161 } 3162 3163 releaseWakeLock(); 3164 3165 return false; 3166} 3167 3168void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3169{ 3170 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3171 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3172 this, 3173 mSampleRate, 3174 mFormat, 3175 mChannelMask, 3176 frameCount); 3177 if (outputTrack->cblk() != NULL) { 3178 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3179 mOutputTracks.add(outputTrack); 3180 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3181 updateWaitTime(); 3182 } 3183} 3184 3185void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3186{ 3187 Mutex::Autolock _l(mLock); 3188 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3189 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3190 mOutputTracks[i]->destroy(); 3191 mOutputTracks.removeAt(i); 3192 updateWaitTime(); 3193 return; 3194 } 3195 } 3196 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3197} 3198 3199void AudioFlinger::DuplicatingThread::updateWaitTime() 3200{ 3201 mWaitTimeMs = UINT_MAX; 3202 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3203 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3204 if (strong != NULL) { 3205 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3206 if (waitTimeMs < mWaitTimeMs) { 3207 mWaitTimeMs = waitTimeMs; 3208 } 3209 } 3210 } 3211} 3212 3213 3214bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3215{ 3216 for (size_t i = 0; i < outputTracks.size(); i++) { 3217 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3218 if (thread == 0) { 3219 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3220 return false; 3221 } 3222 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3223 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3224 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3225 return false; 3226 } 3227 } 3228 return true; 3229} 3230 3231uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3232{ 3233 return (mWaitTimeMs * 1000) / 2; 3234} 3235 3236// ---------------------------------------------------------------------------- 3237 3238// TrackBase constructor must be called with AudioFlinger::mLock held 3239AudioFlinger::ThreadBase::TrackBase::TrackBase( 3240 const wp<ThreadBase>& thread, 3241 const sp<Client>& client, 3242 uint32_t sampleRate, 3243 audio_format_t format, 3244 uint32_t channelMask, 3245 int frameCount, 3246 uint32_t flags, 3247 const sp<IMemory>& sharedBuffer, 3248 int sessionId) 3249 : RefBase(), 3250 mThread(thread), 3251 mClient(client), 3252 mCblk(0), 3253 mFrameCount(0), 3254 mState(IDLE), 3255 mClientTid(-1), 3256 mFormat(format), 3257 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3258 mSessionId(sessionId) 3259{ 3260 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3261 3262 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3263 size_t size = sizeof(audio_track_cblk_t); 3264 uint8_t channelCount = popcount(channelMask); 3265 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3266 if (sharedBuffer == 0) { 3267 size += bufferSize; 3268 } 3269 3270 if (client != NULL) { 3271 mCblkMemory = client->heap()->allocate(size); 3272 if (mCblkMemory != 0) { 3273 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3274 if (mCblk) { // construct the shared structure in-place. 3275 new(mCblk) audio_track_cblk_t(); 3276 // clear all buffers 3277 mCblk->frameCount = frameCount; 3278 mCblk->sampleRate = sampleRate; 3279 mChannelCount = channelCount; 3280 mChannelMask = channelMask; 3281 if (sharedBuffer == 0) { 3282 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3283 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3284 // Force underrun condition to avoid false underrun callback until first data is 3285 // written to buffer (other flags are cleared) 3286 mCblk->flags = CBLK_UNDERRUN_ON; 3287 } else { 3288 mBuffer = sharedBuffer->pointer(); 3289 } 3290 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3291 } 3292 } else { 3293 ALOGE("not enough memory for AudioTrack size=%u", size); 3294 client->heap()->dump("AudioTrack"); 3295 return; 3296 } 3297 } else { 3298 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3299 // construct the shared structure in-place. 3300 new(mCblk) audio_track_cblk_t(); 3301 // clear all buffers 3302 mCblk->frameCount = frameCount; 3303 mCblk->sampleRate = sampleRate; 3304 mChannelCount = channelCount; 3305 mChannelMask = channelMask; 3306 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3307 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3308 // Force underrun condition to avoid false underrun callback until first data is 3309 // written to buffer (other flags are cleared) 3310 mCblk->flags = CBLK_UNDERRUN_ON; 3311 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3312 } 3313} 3314 3315AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3316{ 3317 if (mCblk) { 3318 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3319 if (mClient == NULL) { 3320 delete mCblk; 3321 } 3322 } 3323 mCblkMemory.clear(); // and free the shared memory 3324 if (mClient != NULL) { 3325 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3326 mClient.clear(); 3327 } 3328} 3329 3330void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3331{ 3332 buffer->raw = NULL; 3333 mFrameCount = buffer->frameCount; 3334 step(); 3335 buffer->frameCount = 0; 3336} 3337 3338bool AudioFlinger::ThreadBase::TrackBase::step() { 3339 bool result; 3340 audio_track_cblk_t* cblk = this->cblk(); 3341 3342 result = cblk->stepServer(mFrameCount); 3343 if (!result) { 3344 ALOGV("stepServer failed acquiring cblk mutex"); 3345 mFlags |= STEPSERVER_FAILED; 3346 } 3347 return result; 3348} 3349 3350void AudioFlinger::ThreadBase::TrackBase::reset() { 3351 audio_track_cblk_t* cblk = this->cblk(); 3352 3353 cblk->user = 0; 3354 cblk->server = 0; 3355 cblk->userBase = 0; 3356 cblk->serverBase = 0; 3357 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3358 ALOGV("TrackBase::reset"); 3359} 3360 3361sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3362{ 3363 return mCblkMemory; 3364} 3365 3366int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3367 return (int)mCblk->sampleRate; 3368} 3369 3370int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3371 return (const int)mChannelCount; 3372} 3373 3374uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3375 return mChannelMask; 3376} 3377 3378void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3379 audio_track_cblk_t* cblk = this->cblk(); 3380 size_t frameSize = cblk->frameSize; 3381 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3382 int8_t *bufferEnd = bufferStart + frames * frameSize; 3383 3384 // Check validity of returned pointer in case the track control block would have been corrupted. 3385 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3386 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3387 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3388 server %d, serverBase %d, user %d, userBase %d", 3389 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3390 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3391 return 0; 3392 } 3393 3394 return bufferStart; 3395} 3396 3397// ---------------------------------------------------------------------------- 3398 3399// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3400AudioFlinger::PlaybackThread::Track::Track( 3401 const wp<ThreadBase>& thread, 3402 const sp<Client>& client, 3403 audio_stream_type_t streamType, 3404 uint32_t sampleRate, 3405 audio_format_t format, 3406 uint32_t channelMask, 3407 int frameCount, 3408 const sp<IMemory>& sharedBuffer, 3409 int sessionId) 3410 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3411 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3412 mAuxEffectId(0), mHasVolumeController(false) 3413{ 3414 if (mCblk != NULL) { 3415 sp<ThreadBase> baseThread = thread.promote(); 3416 if (baseThread != 0) { 3417 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3418 mName = playbackThread->getTrackName_l(); 3419 mMainBuffer = playbackThread->mixBuffer(); 3420 } 3421 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3422 if (mName < 0) { 3423 ALOGE("no more track names available"); 3424 } 3425 mStreamType = streamType; 3426 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3427 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3428 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3429 } 3430} 3431 3432AudioFlinger::PlaybackThread::Track::~Track() 3433{ 3434 ALOGV("PlaybackThread::Track destructor"); 3435 sp<ThreadBase> thread = mThread.promote(); 3436 if (thread != 0) { 3437 Mutex::Autolock _l(thread->mLock); 3438 mState = TERMINATED; 3439 } 3440} 3441 3442void AudioFlinger::PlaybackThread::Track::destroy() 3443{ 3444 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3445 // by removing it from mTracks vector, so there is a risk that this Tracks's 3446 // desctructor is called. As the destructor needs to lock mLock, 3447 // we must acquire a strong reference on this Track before locking mLock 3448 // here so that the destructor is called only when exiting this function. 3449 // On the other hand, as long as Track::destroy() is only called by 3450 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3451 // this Track with its member mTrack. 3452 sp<Track> keep(this); 3453 { // scope for mLock 3454 sp<ThreadBase> thread = mThread.promote(); 3455 if (thread != 0) { 3456 if (!isOutputTrack()) { 3457 if (mState == ACTIVE || mState == RESUMING) { 3458 AudioSystem::stopOutput(thread->id(), 3459 (audio_stream_type_t)mStreamType, 3460 mSessionId); 3461 3462 // to track the speaker usage 3463 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3464 } 3465 AudioSystem::releaseOutput(thread->id()); 3466 } 3467 Mutex::Autolock _l(thread->mLock); 3468 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3469 playbackThread->destroyTrack_l(this); 3470 } 3471 } 3472} 3473 3474void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3475{ 3476 uint32_t vlr = mCblk->getVolumeLR(); 3477 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", 3478 mName - AudioMixer::TRACK0, 3479 (mClient == NULL) ? getpid() : mClient->pid(), 3480 mStreamType, 3481 mFormat, 3482 mChannelMask, 3483 mSessionId, 3484 mFrameCount, 3485 mState, 3486 mMute, 3487 mFillingUpStatus, 3488 mCblk->sampleRate, 3489 vlr & 0xFFFF, 3490 vlr >> 16, 3491 mCblk->server, 3492 mCblk->user, 3493 (int)mMainBuffer, 3494 (int)mAuxBuffer); 3495} 3496 3497status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3498{ 3499 audio_track_cblk_t* cblk = this->cblk(); 3500 uint32_t framesReady; 3501 uint32_t framesReq = buffer->frameCount; 3502 3503 // Check if last stepServer failed, try to step now 3504 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3505 if (!step()) goto getNextBuffer_exit; 3506 ALOGV("stepServer recovered"); 3507 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3508 } 3509 3510 framesReady = cblk->framesReady(); 3511 3512 if (CC_LIKELY(framesReady)) { 3513 uint32_t s = cblk->server; 3514 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3515 3516 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3517 if (framesReq > framesReady) { 3518 framesReq = framesReady; 3519 } 3520 if (s + framesReq > bufferEnd) { 3521 framesReq = bufferEnd - s; 3522 } 3523 3524 buffer->raw = getBuffer(s, framesReq); 3525 if (buffer->raw == NULL) goto getNextBuffer_exit; 3526 3527 buffer->frameCount = framesReq; 3528 return NO_ERROR; 3529 } 3530 3531getNextBuffer_exit: 3532 buffer->raw = NULL; 3533 buffer->frameCount = 0; 3534 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3535 return NOT_ENOUGH_DATA; 3536} 3537 3538bool AudioFlinger::PlaybackThread::Track::isReady() const { 3539 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3540 3541 if (mCblk->framesReady() >= mCblk->frameCount || 3542 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3543 mFillingUpStatus = FS_FILLED; 3544 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3545 return true; 3546 } 3547 return false; 3548} 3549 3550status_t AudioFlinger::PlaybackThread::Track::start() 3551{ 3552 status_t status = NO_ERROR; 3553 ALOGV("start(%d), calling thread %d session %d", 3554 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3555 sp<ThreadBase> thread = mThread.promote(); 3556 if (thread != 0) { 3557 Mutex::Autolock _l(thread->mLock); 3558 track_state state = mState; 3559 // here the track could be either new, or restarted 3560 // in both cases "unstop" the track 3561 if (mState == PAUSED) { 3562 mState = TrackBase::RESUMING; 3563 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3564 } else { 3565 mState = TrackBase::ACTIVE; 3566 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3567 } 3568 3569 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3570 thread->mLock.unlock(); 3571 status = AudioSystem::startOutput(thread->id(), 3572 (audio_stream_type_t)mStreamType, 3573 mSessionId); 3574 thread->mLock.lock(); 3575 3576 // to track the speaker usage 3577 if (status == NO_ERROR) { 3578 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3579 } 3580 } 3581 if (status == NO_ERROR) { 3582 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3583 playbackThread->addTrack_l(this); 3584 } else { 3585 mState = state; 3586 } 3587 } else { 3588 status = BAD_VALUE; 3589 } 3590 return status; 3591} 3592 3593void AudioFlinger::PlaybackThread::Track::stop() 3594{ 3595 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3596 sp<ThreadBase> thread = mThread.promote(); 3597 if (thread != 0) { 3598 Mutex::Autolock _l(thread->mLock); 3599 track_state state = mState; 3600 if (mState > STOPPED) { 3601 mState = STOPPED; 3602 // If the track is not active (PAUSED and buffers full), flush buffers 3603 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3604 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3605 reset(); 3606 } 3607 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3608 } 3609 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3610 thread->mLock.unlock(); 3611 AudioSystem::stopOutput(thread->id(), 3612 (audio_stream_type_t)mStreamType, 3613 mSessionId); 3614 thread->mLock.lock(); 3615 3616 // to track the speaker usage 3617 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3618 } 3619 } 3620} 3621 3622void AudioFlinger::PlaybackThread::Track::pause() 3623{ 3624 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3625 sp<ThreadBase> thread = mThread.promote(); 3626 if (thread != 0) { 3627 Mutex::Autolock _l(thread->mLock); 3628 if (mState == ACTIVE || mState == RESUMING) { 3629 mState = PAUSING; 3630 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3631 if (!isOutputTrack()) { 3632 thread->mLock.unlock(); 3633 AudioSystem::stopOutput(thread->id(), 3634 (audio_stream_type_t)mStreamType, 3635 mSessionId); 3636 thread->mLock.lock(); 3637 3638 // to track the speaker usage 3639 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3640 } 3641 } 3642 } 3643} 3644 3645void AudioFlinger::PlaybackThread::Track::flush() 3646{ 3647 ALOGV("flush(%d)", mName); 3648 sp<ThreadBase> thread = mThread.promote(); 3649 if (thread != 0) { 3650 Mutex::Autolock _l(thread->mLock); 3651 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3652 return; 3653 } 3654 // No point remaining in PAUSED state after a flush => go to 3655 // STOPPED state 3656 mState = STOPPED; 3657 3658 // do not reset the track if it is still in the process of being stopped or paused. 3659 // this will be done by prepareTracks_l() when the track is stopped. 3660 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3661 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3662 reset(); 3663 } 3664 } 3665} 3666 3667void AudioFlinger::PlaybackThread::Track::reset() 3668{ 3669 // Do not reset twice to avoid discarding data written just after a flush and before 3670 // the audioflinger thread detects the track is stopped. 3671 if (!mResetDone) { 3672 TrackBase::reset(); 3673 // Force underrun condition to avoid false underrun callback until first data is 3674 // written to buffer 3675 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3676 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3677 mFillingUpStatus = FS_FILLING; 3678 mResetDone = true; 3679 } 3680} 3681 3682void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3683{ 3684 mMute = muted; 3685} 3686 3687status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3688{ 3689 status_t status = DEAD_OBJECT; 3690 sp<ThreadBase> thread = mThread.promote(); 3691 if (thread != 0) { 3692 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3693 status = playbackThread->attachAuxEffect(this, EffectId); 3694 } 3695 return status; 3696} 3697 3698void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3699{ 3700 mAuxEffectId = EffectId; 3701 mAuxBuffer = buffer; 3702} 3703 3704// ---------------------------------------------------------------------------- 3705 3706// RecordTrack constructor must be called with AudioFlinger::mLock held 3707AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3708 const wp<ThreadBase>& thread, 3709 const sp<Client>& client, 3710 uint32_t sampleRate, 3711 audio_format_t format, 3712 uint32_t channelMask, 3713 int frameCount, 3714 uint32_t flags, 3715 int sessionId) 3716 : TrackBase(thread, client, sampleRate, format, 3717 channelMask, frameCount, flags, 0, sessionId), 3718 mOverflow(false) 3719{ 3720 if (mCblk != NULL) { 3721 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3722 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3723 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3724 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3725 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3726 } else { 3727 mCblk->frameSize = sizeof(int8_t); 3728 } 3729 } 3730} 3731 3732AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3733{ 3734 sp<ThreadBase> thread = mThread.promote(); 3735 if (thread != 0) { 3736 AudioSystem::releaseInput(thread->id()); 3737 } 3738} 3739 3740status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3741{ 3742 audio_track_cblk_t* cblk = this->cblk(); 3743 uint32_t framesAvail; 3744 uint32_t framesReq = buffer->frameCount; 3745 3746 // Check if last stepServer failed, try to step now 3747 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3748 if (!step()) goto getNextBuffer_exit; 3749 ALOGV("stepServer recovered"); 3750 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3751 } 3752 3753 framesAvail = cblk->framesAvailable_l(); 3754 3755 if (CC_LIKELY(framesAvail)) { 3756 uint32_t s = cblk->server; 3757 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3758 3759 if (framesReq > framesAvail) { 3760 framesReq = framesAvail; 3761 } 3762 if (s + framesReq > bufferEnd) { 3763 framesReq = bufferEnd - s; 3764 } 3765 3766 buffer->raw = getBuffer(s, framesReq); 3767 if (buffer->raw == NULL) goto getNextBuffer_exit; 3768 3769 buffer->frameCount = framesReq; 3770 return NO_ERROR; 3771 } 3772 3773getNextBuffer_exit: 3774 buffer->raw = NULL; 3775 buffer->frameCount = 0; 3776 return NOT_ENOUGH_DATA; 3777} 3778 3779status_t AudioFlinger::RecordThread::RecordTrack::start() 3780{ 3781 sp<ThreadBase> thread = mThread.promote(); 3782 if (thread != 0) { 3783 RecordThread *recordThread = (RecordThread *)thread.get(); 3784 return recordThread->start(this); 3785 } else { 3786 return BAD_VALUE; 3787 } 3788} 3789 3790void AudioFlinger::RecordThread::RecordTrack::stop() 3791{ 3792 sp<ThreadBase> thread = mThread.promote(); 3793 if (thread != 0) { 3794 RecordThread *recordThread = (RecordThread *)thread.get(); 3795 recordThread->stop(this); 3796 TrackBase::reset(); 3797 // Force overerrun condition to avoid false overrun callback until first data is 3798 // read from buffer 3799 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3800 } 3801} 3802 3803void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3804{ 3805 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3806 (mClient == NULL) ? getpid() : mClient->pid(), 3807 mFormat, 3808 mChannelMask, 3809 mSessionId, 3810 mFrameCount, 3811 mState, 3812 mCblk->sampleRate, 3813 mCblk->server, 3814 mCblk->user); 3815} 3816 3817 3818// ---------------------------------------------------------------------------- 3819 3820AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3821 const wp<ThreadBase>& thread, 3822 DuplicatingThread *sourceThread, 3823 uint32_t sampleRate, 3824 audio_format_t format, 3825 uint32_t channelMask, 3826 int frameCount) 3827 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3828 mActive(false), mSourceThread(sourceThread) 3829{ 3830 3831 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3832 if (mCblk != NULL) { 3833 mCblk->flags |= CBLK_DIRECTION_OUT; 3834 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3835 mOutBuffer.frameCount = 0; 3836 playbackThread->mTracks.add(this); 3837 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3838 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3839 mCblk, mBuffer, mCblk->buffers, 3840 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3841 } else { 3842 ALOGW("Error creating output track on thread %p", playbackThread); 3843 } 3844} 3845 3846AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3847{ 3848 clearBufferQueue(); 3849} 3850 3851status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3852{ 3853 status_t status = Track::start(); 3854 if (status != NO_ERROR) { 3855 return status; 3856 } 3857 3858 mActive = true; 3859 mRetryCount = 127; 3860 return status; 3861} 3862 3863void AudioFlinger::PlaybackThread::OutputTrack::stop() 3864{ 3865 Track::stop(); 3866 clearBufferQueue(); 3867 mOutBuffer.frameCount = 0; 3868 mActive = false; 3869} 3870 3871bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3872{ 3873 Buffer *pInBuffer; 3874 Buffer inBuffer; 3875 uint32_t channelCount = mChannelCount; 3876 bool outputBufferFull = false; 3877 inBuffer.frameCount = frames; 3878 inBuffer.i16 = data; 3879 3880 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3881 3882 if (!mActive && frames != 0) { 3883 start(); 3884 sp<ThreadBase> thread = mThread.promote(); 3885 if (thread != 0) { 3886 MixerThread *mixerThread = (MixerThread *)thread.get(); 3887 if (mCblk->frameCount > frames){ 3888 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3889 uint32_t startFrames = (mCblk->frameCount - frames); 3890 pInBuffer = new Buffer; 3891 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3892 pInBuffer->frameCount = startFrames; 3893 pInBuffer->i16 = pInBuffer->mBuffer; 3894 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3895 mBufferQueue.add(pInBuffer); 3896 } else { 3897 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3898 } 3899 } 3900 } 3901 } 3902 3903 while (waitTimeLeftMs) { 3904 // First write pending buffers, then new data 3905 if (mBufferQueue.size()) { 3906 pInBuffer = mBufferQueue.itemAt(0); 3907 } else { 3908 pInBuffer = &inBuffer; 3909 } 3910 3911 if (pInBuffer->frameCount == 0) { 3912 break; 3913 } 3914 3915 if (mOutBuffer.frameCount == 0) { 3916 mOutBuffer.frameCount = pInBuffer->frameCount; 3917 nsecs_t startTime = systemTime(); 3918 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3919 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3920 outputBufferFull = true; 3921 break; 3922 } 3923 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3924 if (waitTimeLeftMs >= waitTimeMs) { 3925 waitTimeLeftMs -= waitTimeMs; 3926 } else { 3927 waitTimeLeftMs = 0; 3928 } 3929 } 3930 3931 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3932 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3933 mCblk->stepUser(outFrames); 3934 pInBuffer->frameCount -= outFrames; 3935 pInBuffer->i16 += outFrames * channelCount; 3936 mOutBuffer.frameCount -= outFrames; 3937 mOutBuffer.i16 += outFrames * channelCount; 3938 3939 if (pInBuffer->frameCount == 0) { 3940 if (mBufferQueue.size()) { 3941 mBufferQueue.removeAt(0); 3942 delete [] pInBuffer->mBuffer; 3943 delete pInBuffer; 3944 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3945 } else { 3946 break; 3947 } 3948 } 3949 } 3950 3951 // If we could not write all frames, allocate a buffer and queue it for next time. 3952 if (inBuffer.frameCount) { 3953 sp<ThreadBase> thread = mThread.promote(); 3954 if (thread != 0 && !thread->standby()) { 3955 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3956 pInBuffer = new Buffer; 3957 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3958 pInBuffer->frameCount = inBuffer.frameCount; 3959 pInBuffer->i16 = pInBuffer->mBuffer; 3960 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3961 mBufferQueue.add(pInBuffer); 3962 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3963 } else { 3964 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3965 } 3966 } 3967 } 3968 3969 // Calling write() with a 0 length buffer, means that no more data will be written: 3970 // If no more buffers are pending, fill output track buffer to make sure it is started 3971 // by output mixer. 3972 if (frames == 0 && mBufferQueue.size() == 0) { 3973 if (mCblk->user < mCblk->frameCount) { 3974 frames = mCblk->frameCount - mCblk->user; 3975 pInBuffer = new Buffer; 3976 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3977 pInBuffer->frameCount = frames; 3978 pInBuffer->i16 = pInBuffer->mBuffer; 3979 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3980 mBufferQueue.add(pInBuffer); 3981 } else if (mActive) { 3982 stop(); 3983 } 3984 } 3985 3986 return outputBufferFull; 3987} 3988 3989status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3990{ 3991 int active; 3992 status_t result; 3993 audio_track_cblk_t* cblk = mCblk; 3994 uint32_t framesReq = buffer->frameCount; 3995 3996// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3997 buffer->frameCount = 0; 3998 3999 uint32_t framesAvail = cblk->framesAvailable(); 4000 4001 4002 if (framesAvail == 0) { 4003 Mutex::Autolock _l(cblk->lock); 4004 goto start_loop_here; 4005 while (framesAvail == 0) { 4006 active = mActive; 4007 if (CC_UNLIKELY(!active)) { 4008 ALOGV("Not active and NO_MORE_BUFFERS"); 4009 return NO_MORE_BUFFERS; 4010 } 4011 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4012 if (result != NO_ERROR) { 4013 return NO_MORE_BUFFERS; 4014 } 4015 // read the server count again 4016 start_loop_here: 4017 framesAvail = cblk->framesAvailable_l(); 4018 } 4019 } 4020 4021// if (framesAvail < framesReq) { 4022// return NO_MORE_BUFFERS; 4023// } 4024 4025 if (framesReq > framesAvail) { 4026 framesReq = framesAvail; 4027 } 4028 4029 uint32_t u = cblk->user; 4030 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4031 4032 if (u + framesReq > bufferEnd) { 4033 framesReq = bufferEnd - u; 4034 } 4035 4036 buffer->frameCount = framesReq; 4037 buffer->raw = (void *)cblk->buffer(u); 4038 return NO_ERROR; 4039} 4040 4041 4042void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4043{ 4044 size_t size = mBufferQueue.size(); 4045 Buffer *pBuffer; 4046 4047 for (size_t i = 0; i < size; i++) { 4048 pBuffer = mBufferQueue.itemAt(i); 4049 delete [] pBuffer->mBuffer; 4050 delete pBuffer; 4051 } 4052 mBufferQueue.clear(); 4053} 4054 4055// ---------------------------------------------------------------------------- 4056 4057AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4058 : RefBase(), 4059 mAudioFlinger(audioFlinger), 4060 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4061 mPid(pid) 4062{ 4063 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4064} 4065 4066// Client destructor must be called with AudioFlinger::mLock held 4067AudioFlinger::Client::~Client() 4068{ 4069 mAudioFlinger->removeClient_l(mPid); 4070} 4071 4072sp<MemoryDealer> AudioFlinger::Client::heap() const 4073{ 4074 return mMemoryDealer; 4075} 4076 4077// ---------------------------------------------------------------------------- 4078 4079AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4080 const sp<IAudioFlingerClient>& client, 4081 pid_t pid) 4082 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4083{ 4084} 4085 4086AudioFlinger::NotificationClient::~NotificationClient() 4087{ 4088 mClient.clear(); 4089} 4090 4091void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4092{ 4093 sp<NotificationClient> keep(this); 4094 { 4095 mAudioFlinger->removeNotificationClient(mPid); 4096 } 4097} 4098 4099// ---------------------------------------------------------------------------- 4100 4101AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4102 : BnAudioTrack(), 4103 mTrack(track) 4104{ 4105} 4106 4107AudioFlinger::TrackHandle::~TrackHandle() { 4108 // just stop the track on deletion, associated resources 4109 // will be freed from the main thread once all pending buffers have 4110 // been played. Unless it's not in the active track list, in which 4111 // case we free everything now... 4112 mTrack->destroy(); 4113} 4114 4115sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4116 return mTrack->getCblk(); 4117} 4118 4119status_t AudioFlinger::TrackHandle::start() { 4120 return mTrack->start(); 4121} 4122 4123void AudioFlinger::TrackHandle::stop() { 4124 mTrack->stop(); 4125} 4126 4127void AudioFlinger::TrackHandle::flush() { 4128 mTrack->flush(); 4129} 4130 4131void AudioFlinger::TrackHandle::mute(bool e) { 4132 mTrack->mute(e); 4133} 4134 4135void AudioFlinger::TrackHandle::pause() { 4136 mTrack->pause(); 4137} 4138 4139status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4140{ 4141 return mTrack->attachAuxEffect(EffectId); 4142} 4143 4144status_t AudioFlinger::TrackHandle::onTransact( 4145 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4146{ 4147 return BnAudioTrack::onTransact(code, data, reply, flags); 4148} 4149 4150// ---------------------------------------------------------------------------- 4151 4152sp<IAudioRecord> AudioFlinger::openRecord( 4153 pid_t pid, 4154 int input, 4155 uint32_t sampleRate, 4156 audio_format_t format, 4157 uint32_t channelMask, 4158 int frameCount, 4159 uint32_t flags, 4160 int *sessionId, 4161 status_t *status) 4162{ 4163 sp<RecordThread::RecordTrack> recordTrack; 4164 sp<RecordHandle> recordHandle; 4165 sp<Client> client; 4166 wp<Client> wclient; 4167 status_t lStatus; 4168 RecordThread *thread; 4169 size_t inFrameCount; 4170 int lSessionId; 4171 4172 // check calling permissions 4173 if (!recordingAllowed()) { 4174 lStatus = PERMISSION_DENIED; 4175 goto Exit; 4176 } 4177 4178 // add client to list 4179 { // scope for mLock 4180 Mutex::Autolock _l(mLock); 4181 thread = checkRecordThread_l(input); 4182 if (thread == NULL) { 4183 lStatus = BAD_VALUE; 4184 goto Exit; 4185 } 4186 4187 wclient = mClients.valueFor(pid); 4188 if (wclient != NULL) { 4189 client = wclient.promote(); 4190 } else { 4191 client = new Client(this, pid); 4192 mClients.add(pid, client); 4193 } 4194 4195 // If no audio session id is provided, create one here 4196 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4197 lSessionId = *sessionId; 4198 } else { 4199 lSessionId = nextUniqueId(); 4200 if (sessionId != NULL) { 4201 *sessionId = lSessionId; 4202 } 4203 } 4204 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4205 recordTrack = thread->createRecordTrack_l(client, 4206 sampleRate, 4207 format, 4208 channelMask, 4209 frameCount, 4210 flags, 4211 lSessionId, 4212 &lStatus); 4213 } 4214 if (lStatus != NO_ERROR) { 4215 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4216 // destructor is called by the TrackBase destructor with mLock held 4217 client.clear(); 4218 recordTrack.clear(); 4219 goto Exit; 4220 } 4221 4222 // return to handle to client 4223 recordHandle = new RecordHandle(recordTrack); 4224 lStatus = NO_ERROR; 4225 4226Exit: 4227 if (status) { 4228 *status = lStatus; 4229 } 4230 return recordHandle; 4231} 4232 4233// ---------------------------------------------------------------------------- 4234 4235AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4236 : BnAudioRecord(), 4237 mRecordTrack(recordTrack) 4238{ 4239} 4240 4241AudioFlinger::RecordHandle::~RecordHandle() { 4242 stop(); 4243} 4244 4245sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4246 return mRecordTrack->getCblk(); 4247} 4248 4249status_t AudioFlinger::RecordHandle::start() { 4250 ALOGV("RecordHandle::start()"); 4251 return mRecordTrack->start(); 4252} 4253 4254void AudioFlinger::RecordHandle::stop() { 4255 ALOGV("RecordHandle::stop()"); 4256 mRecordTrack->stop(); 4257} 4258 4259status_t AudioFlinger::RecordHandle::onTransact( 4260 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4261{ 4262 return BnAudioRecord::onTransact(code, data, reply, flags); 4263} 4264 4265// ---------------------------------------------------------------------------- 4266 4267AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4268 AudioStreamIn *input, 4269 uint32_t sampleRate, 4270 uint32_t channels, 4271 int id, 4272 uint32_t device) : 4273 ThreadBase(audioFlinger, id, device, RECORD), 4274 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4275{ 4276 snprintf(mName, kNameLength, "AudioIn_%d", id); 4277 4278 mReqChannelCount = popcount(channels); 4279 mReqSampleRate = sampleRate; 4280 readInputParameters(); 4281} 4282 4283 4284AudioFlinger::RecordThread::~RecordThread() 4285{ 4286 delete[] mRsmpInBuffer; 4287 delete mResampler; 4288 delete[] mRsmpOutBuffer; 4289} 4290 4291void AudioFlinger::RecordThread::onFirstRef() 4292{ 4293 run(mName, PRIORITY_URGENT_AUDIO); 4294} 4295 4296status_t AudioFlinger::RecordThread::readyToRun() 4297{ 4298 status_t status = initCheck(); 4299 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4300 return status; 4301} 4302 4303bool AudioFlinger::RecordThread::threadLoop() 4304{ 4305 AudioBufferProvider::Buffer buffer; 4306 sp<RecordTrack> activeTrack; 4307 Vector< sp<EffectChain> > effectChains; 4308 4309 nsecs_t lastWarning = 0; 4310 4311 acquireWakeLock(); 4312 4313 // start recording 4314 while (!exitPending()) { 4315 4316 processConfigEvents(); 4317 4318 { // scope for mLock 4319 Mutex::Autolock _l(mLock); 4320 checkForNewParameters_l(); 4321 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4322 if (!mStandby) { 4323 mInput->stream->common.standby(&mInput->stream->common); 4324 mStandby = true; 4325 } 4326 4327 if (exitPending()) break; 4328 4329 releaseWakeLock_l(); 4330 ALOGV("RecordThread: loop stopping"); 4331 // go to sleep 4332 mWaitWorkCV.wait(mLock); 4333 ALOGV("RecordThread: loop starting"); 4334 acquireWakeLock_l(); 4335 continue; 4336 } 4337 if (mActiveTrack != 0) { 4338 if (mActiveTrack->mState == TrackBase::PAUSING) { 4339 if (!mStandby) { 4340 mInput->stream->common.standby(&mInput->stream->common); 4341 mStandby = true; 4342 } 4343 mActiveTrack.clear(); 4344 mStartStopCond.broadcast(); 4345 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4346 if (mReqChannelCount != mActiveTrack->channelCount()) { 4347 mActiveTrack.clear(); 4348 mStartStopCond.broadcast(); 4349 } else if (mBytesRead != 0) { 4350 // record start succeeds only if first read from audio input 4351 // succeeds 4352 if (mBytesRead > 0) { 4353 mActiveTrack->mState = TrackBase::ACTIVE; 4354 } else { 4355 mActiveTrack.clear(); 4356 } 4357 mStartStopCond.broadcast(); 4358 } 4359 mStandby = false; 4360 } 4361 } 4362 lockEffectChains_l(effectChains); 4363 } 4364 4365 if (mActiveTrack != 0) { 4366 if (mActiveTrack->mState != TrackBase::ACTIVE && 4367 mActiveTrack->mState != TrackBase::RESUMING) { 4368 unlockEffectChains(effectChains); 4369 usleep(kRecordThreadSleepUs); 4370 continue; 4371 } 4372 for (size_t i = 0; i < effectChains.size(); i ++) { 4373 effectChains[i]->process_l(); 4374 } 4375 4376 buffer.frameCount = mFrameCount; 4377 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4378 size_t framesOut = buffer.frameCount; 4379 if (mResampler == NULL) { 4380 // no resampling 4381 while (framesOut) { 4382 size_t framesIn = mFrameCount - mRsmpInIndex; 4383 if (framesIn) { 4384 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4385 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4386 if (framesIn > framesOut) 4387 framesIn = framesOut; 4388 mRsmpInIndex += framesIn; 4389 framesOut -= framesIn; 4390 if ((int)mChannelCount == mReqChannelCount || 4391 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4392 memcpy(dst, src, framesIn * mFrameSize); 4393 } else { 4394 int16_t *src16 = (int16_t *)src; 4395 int16_t *dst16 = (int16_t *)dst; 4396 if (mChannelCount == 1) { 4397 while (framesIn--) { 4398 *dst16++ = *src16; 4399 *dst16++ = *src16++; 4400 } 4401 } else { 4402 while (framesIn--) { 4403 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4404 src16 += 2; 4405 } 4406 } 4407 } 4408 } 4409 if (framesOut && mFrameCount == mRsmpInIndex) { 4410 if (framesOut == mFrameCount && 4411 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4412 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4413 framesOut = 0; 4414 } else { 4415 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4416 mRsmpInIndex = 0; 4417 } 4418 if (mBytesRead < 0) { 4419 ALOGE("Error reading audio input"); 4420 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4421 // Force input into standby so that it tries to 4422 // recover at next read attempt 4423 mInput->stream->common.standby(&mInput->stream->common); 4424 usleep(kRecordThreadSleepUs); 4425 } 4426 mRsmpInIndex = mFrameCount; 4427 framesOut = 0; 4428 buffer.frameCount = 0; 4429 } 4430 } 4431 } 4432 } else { 4433 // resampling 4434 4435 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4436 // alter output frame count as if we were expecting stereo samples 4437 if (mChannelCount == 1 && mReqChannelCount == 1) { 4438 framesOut >>= 1; 4439 } 4440 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4441 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4442 // are 32 bit aligned which should be always true. 4443 if (mChannelCount == 2 && mReqChannelCount == 1) { 4444 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4445 // the resampler always outputs stereo samples: do post stereo to mono conversion 4446 int16_t *src = (int16_t *)mRsmpOutBuffer; 4447 int16_t *dst = buffer.i16; 4448 while (framesOut--) { 4449 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4450 src += 2; 4451 } 4452 } else { 4453 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4454 } 4455 4456 } 4457 mActiveTrack->releaseBuffer(&buffer); 4458 mActiveTrack->overflow(); 4459 } 4460 // client isn't retrieving buffers fast enough 4461 else { 4462 if (!mActiveTrack->setOverflow()) { 4463 nsecs_t now = systemTime(); 4464 if ((now - lastWarning) > kWarningThrottleNs) { 4465 ALOGW("RecordThread: buffer overflow"); 4466 lastWarning = now; 4467 } 4468 } 4469 // Release the processor for a while before asking for a new buffer. 4470 // This will give the application more chance to read from the buffer and 4471 // clear the overflow. 4472 usleep(kRecordThreadSleepUs); 4473 } 4474 } 4475 // enable changes in effect chain 4476 unlockEffectChains(effectChains); 4477 effectChains.clear(); 4478 } 4479 4480 if (!mStandby) { 4481 mInput->stream->common.standby(&mInput->stream->common); 4482 } 4483 mActiveTrack.clear(); 4484 4485 mStartStopCond.broadcast(); 4486 4487 releaseWakeLock(); 4488 4489 ALOGV("RecordThread %p exiting", this); 4490 return false; 4491} 4492 4493 4494sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4495 const sp<AudioFlinger::Client>& client, 4496 uint32_t sampleRate, 4497 audio_format_t format, 4498 int channelMask, 4499 int frameCount, 4500 uint32_t flags, 4501 int sessionId, 4502 status_t *status) 4503{ 4504 sp<RecordTrack> track; 4505 status_t lStatus; 4506 4507 lStatus = initCheck(); 4508 if (lStatus != NO_ERROR) { 4509 ALOGE("Audio driver not initialized."); 4510 goto Exit; 4511 } 4512 4513 { // scope for mLock 4514 Mutex::Autolock _l(mLock); 4515 4516 track = new RecordTrack(this, client, sampleRate, 4517 format, channelMask, frameCount, flags, sessionId); 4518 4519 if (track->getCblk() == NULL) { 4520 lStatus = NO_MEMORY; 4521 goto Exit; 4522 } 4523 4524 mTrack = track.get(); 4525 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4526 bool suspend = audio_is_bluetooth_sco_device( 4527 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4528 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4529 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4530 } 4531 lStatus = NO_ERROR; 4532 4533Exit: 4534 if (status) { 4535 *status = lStatus; 4536 } 4537 return track; 4538} 4539 4540status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4541{ 4542 ALOGV("RecordThread::start"); 4543 sp <ThreadBase> strongMe = this; 4544 status_t status = NO_ERROR; 4545 { 4546 AutoMutex lock(mLock); 4547 if (mActiveTrack != 0) { 4548 if (recordTrack != mActiveTrack.get()) { 4549 status = -EBUSY; 4550 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4551 mActiveTrack->mState = TrackBase::ACTIVE; 4552 } 4553 return status; 4554 } 4555 4556 recordTrack->mState = TrackBase::IDLE; 4557 mActiveTrack = recordTrack; 4558 mLock.unlock(); 4559 status_t status = AudioSystem::startInput(mId); 4560 mLock.lock(); 4561 if (status != NO_ERROR) { 4562 mActiveTrack.clear(); 4563 return status; 4564 } 4565 mRsmpInIndex = mFrameCount; 4566 mBytesRead = 0; 4567 if (mResampler != NULL) { 4568 mResampler->reset(); 4569 } 4570 mActiveTrack->mState = TrackBase::RESUMING; 4571 // signal thread to start 4572 ALOGV("Signal record thread"); 4573 mWaitWorkCV.signal(); 4574 // do not wait for mStartStopCond if exiting 4575 if (mExiting) { 4576 mActiveTrack.clear(); 4577 status = INVALID_OPERATION; 4578 goto startError; 4579 } 4580 mStartStopCond.wait(mLock); 4581 if (mActiveTrack == 0) { 4582 ALOGV("Record failed to start"); 4583 status = BAD_VALUE; 4584 goto startError; 4585 } 4586 ALOGV("Record started OK"); 4587 return status; 4588 } 4589startError: 4590 AudioSystem::stopInput(mId); 4591 return status; 4592} 4593 4594void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4595 ALOGV("RecordThread::stop"); 4596 sp <ThreadBase> strongMe = this; 4597 { 4598 AutoMutex lock(mLock); 4599 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4600 mActiveTrack->mState = TrackBase::PAUSING; 4601 // do not wait for mStartStopCond if exiting 4602 if (mExiting) { 4603 return; 4604 } 4605 mStartStopCond.wait(mLock); 4606 // if we have been restarted, recordTrack == mActiveTrack.get() here 4607 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4608 mLock.unlock(); 4609 AudioSystem::stopInput(mId); 4610 mLock.lock(); 4611 ALOGV("Record stopped OK"); 4612 } 4613 } 4614 } 4615} 4616 4617status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4618{ 4619 const size_t SIZE = 256; 4620 char buffer[SIZE]; 4621 String8 result; 4622 pid_t pid = 0; 4623 4624 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4625 result.append(buffer); 4626 4627 if (mActiveTrack != 0) { 4628 result.append("Active Track:\n"); 4629 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4630 mActiveTrack->dump(buffer, SIZE); 4631 result.append(buffer); 4632 4633 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4634 result.append(buffer); 4635 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4636 result.append(buffer); 4637 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4638 result.append(buffer); 4639 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4640 result.append(buffer); 4641 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4642 result.append(buffer); 4643 4644 4645 } else { 4646 result.append("No record client\n"); 4647 } 4648 write(fd, result.string(), result.size()); 4649 4650 dumpBase(fd, args); 4651 dumpEffectChains(fd, args); 4652 4653 return NO_ERROR; 4654} 4655 4656status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4657{ 4658 size_t framesReq = buffer->frameCount; 4659 size_t framesReady = mFrameCount - mRsmpInIndex; 4660 int channelCount; 4661 4662 if (framesReady == 0) { 4663 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4664 if (mBytesRead < 0) { 4665 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4666 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4667 // Force input into standby so that it tries to 4668 // recover at next read attempt 4669 mInput->stream->common.standby(&mInput->stream->common); 4670 usleep(kRecordThreadSleepUs); 4671 } 4672 buffer->raw = NULL; 4673 buffer->frameCount = 0; 4674 return NOT_ENOUGH_DATA; 4675 } 4676 mRsmpInIndex = 0; 4677 framesReady = mFrameCount; 4678 } 4679 4680 if (framesReq > framesReady) { 4681 framesReq = framesReady; 4682 } 4683 4684 if (mChannelCount == 1 && mReqChannelCount == 2) { 4685 channelCount = 1; 4686 } else { 4687 channelCount = 2; 4688 } 4689 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4690 buffer->frameCount = framesReq; 4691 return NO_ERROR; 4692} 4693 4694void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4695{ 4696 mRsmpInIndex += buffer->frameCount; 4697 buffer->frameCount = 0; 4698} 4699 4700bool AudioFlinger::RecordThread::checkForNewParameters_l() 4701{ 4702 bool reconfig = false; 4703 4704 while (!mNewParameters.isEmpty()) { 4705 status_t status = NO_ERROR; 4706 String8 keyValuePair = mNewParameters[0]; 4707 AudioParameter param = AudioParameter(keyValuePair); 4708 int value; 4709 audio_format_t reqFormat = mFormat; 4710 int reqSamplingRate = mReqSampleRate; 4711 int reqChannelCount = mReqChannelCount; 4712 4713 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4714 reqSamplingRate = value; 4715 reconfig = true; 4716 } 4717 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4718 reqFormat = (audio_format_t) value; 4719 reconfig = true; 4720 } 4721 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4722 reqChannelCount = popcount(value); 4723 reconfig = true; 4724 } 4725 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4726 // do not accept frame count changes if tracks are open as the track buffer 4727 // size depends on frame count and correct behavior would not be garantied 4728 // if frame count is changed after track creation 4729 if (mActiveTrack != 0) { 4730 status = INVALID_OPERATION; 4731 } else { 4732 reconfig = true; 4733 } 4734 } 4735 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4736 // forward device change to effects that have requested to be 4737 // aware of attached audio device. 4738 for (size_t i = 0; i < mEffectChains.size(); i++) { 4739 mEffectChains[i]->setDevice_l(value); 4740 } 4741 // store input device and output device but do not forward output device to audio HAL. 4742 // Note that status is ignored by the caller for output device 4743 // (see AudioFlinger::setParameters() 4744 if (value & AUDIO_DEVICE_OUT_ALL) { 4745 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4746 status = BAD_VALUE; 4747 } else { 4748 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4749 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4750 if (mTrack != NULL) { 4751 bool suspend = audio_is_bluetooth_sco_device( 4752 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4753 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4754 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4755 } 4756 } 4757 mDevice |= (uint32_t)value; 4758 } 4759 if (status == NO_ERROR) { 4760 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4761 if (status == INVALID_OPERATION) { 4762 mInput->stream->common.standby(&mInput->stream->common); 4763 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4764 } 4765 if (reconfig) { 4766 if (status == BAD_VALUE && 4767 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4768 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4769 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4770 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4771 (reqChannelCount < 3)) { 4772 status = NO_ERROR; 4773 } 4774 if (status == NO_ERROR) { 4775 readInputParameters(); 4776 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4777 } 4778 } 4779 } 4780 4781 mNewParameters.removeAt(0); 4782 4783 mParamStatus = status; 4784 mParamCond.signal(); 4785 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4786 // already timed out waiting for the status and will never signal the condition. 4787 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4788 } 4789 return reconfig; 4790} 4791 4792String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4793{ 4794 char *s; 4795 String8 out_s8 = String8(); 4796 4797 Mutex::Autolock _l(mLock); 4798 if (initCheck() != NO_ERROR) { 4799 return out_s8; 4800 } 4801 4802 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4803 out_s8 = String8(s); 4804 free(s); 4805 return out_s8; 4806} 4807 4808void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4809 AudioSystem::OutputDescriptor desc; 4810 void *param2 = 0; 4811 4812 switch (event) { 4813 case AudioSystem::INPUT_OPENED: 4814 case AudioSystem::INPUT_CONFIG_CHANGED: 4815 desc.channels = mChannelMask; 4816 desc.samplingRate = mSampleRate; 4817 desc.format = mFormat; 4818 desc.frameCount = mFrameCount; 4819 desc.latency = 0; 4820 param2 = &desc; 4821 break; 4822 4823 case AudioSystem::INPUT_CLOSED: 4824 default: 4825 break; 4826 } 4827 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4828} 4829 4830void AudioFlinger::RecordThread::readInputParameters() 4831{ 4832 delete mRsmpInBuffer; 4833 // mRsmpInBuffer is always assigned a new[] below 4834 delete mRsmpOutBuffer; 4835 mRsmpOutBuffer = NULL; 4836 delete mResampler; 4837 mResampler = NULL; 4838 4839 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4840 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4841 mChannelCount = (uint16_t)popcount(mChannelMask); 4842 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4843 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4844 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4845 mFrameCount = mInputBytes / mFrameSize; 4846 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4847 4848 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4849 { 4850 int channelCount; 4851 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4852 // stereo to mono post process as the resampler always outputs stereo. 4853 if (mChannelCount == 1 && mReqChannelCount == 2) { 4854 channelCount = 1; 4855 } else { 4856 channelCount = 2; 4857 } 4858 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4859 mResampler->setSampleRate(mSampleRate); 4860 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4861 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4862 4863 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4864 if (mChannelCount == 1 && mReqChannelCount == 1) { 4865 mFrameCount >>= 1; 4866 } 4867 4868 } 4869 mRsmpInIndex = mFrameCount; 4870} 4871 4872unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4873{ 4874 Mutex::Autolock _l(mLock); 4875 if (initCheck() != NO_ERROR) { 4876 return 0; 4877 } 4878 4879 return mInput->stream->get_input_frames_lost(mInput->stream); 4880} 4881 4882uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4883{ 4884 Mutex::Autolock _l(mLock); 4885 uint32_t result = 0; 4886 if (getEffectChain_l(sessionId) != 0) { 4887 result = EFFECT_SESSION; 4888 } 4889 4890 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4891 result |= TRACK_SESSION; 4892 } 4893 4894 return result; 4895} 4896 4897AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4898{ 4899 Mutex::Autolock _l(mLock); 4900 return mTrack; 4901} 4902 4903AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4904{ 4905 Mutex::Autolock _l(mLock); 4906 return mInput; 4907} 4908 4909AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4910{ 4911 Mutex::Autolock _l(mLock); 4912 AudioStreamIn *input = mInput; 4913 mInput = NULL; 4914 return input; 4915} 4916 4917// this method must always be called either with ThreadBase mLock held or inside the thread loop 4918audio_stream_t* AudioFlinger::RecordThread::stream() 4919{ 4920 if (mInput == NULL) { 4921 return NULL; 4922 } 4923 return &mInput->stream->common; 4924} 4925 4926 4927// ---------------------------------------------------------------------------- 4928 4929int AudioFlinger::openOutput(uint32_t *pDevices, 4930 uint32_t *pSamplingRate, 4931 audio_format_t *pFormat, 4932 uint32_t *pChannels, 4933 uint32_t *pLatencyMs, 4934 uint32_t flags) 4935{ 4936 status_t status; 4937 PlaybackThread *thread = NULL; 4938 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4939 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4940 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4941 uint32_t channels = pChannels ? *pChannels : 0; 4942 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4943 audio_stream_out_t *outStream; 4944 audio_hw_device_t *outHwDev; 4945 4946 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4947 pDevices ? *pDevices : 0, 4948 samplingRate, 4949 format, 4950 channels, 4951 flags); 4952 4953 if (pDevices == NULL || *pDevices == 0) { 4954 return 0; 4955 } 4956 4957 Mutex::Autolock _l(mLock); 4958 4959 outHwDev = findSuitableHwDev_l(*pDevices); 4960 if (outHwDev == NULL) 4961 return 0; 4962 4963 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4964 &channels, &samplingRate, &outStream); 4965 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4966 outStream, 4967 samplingRate, 4968 format, 4969 channels, 4970 status); 4971 4972 mHardwareStatus = AUDIO_HW_IDLE; 4973 if (outStream != NULL) { 4974 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4975 int id = nextUniqueId(); 4976 4977 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4978 (format != AUDIO_FORMAT_PCM_16_BIT) || 4979 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4980 thread = new DirectOutputThread(this, output, id, *pDevices); 4981 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4982 } else { 4983 thread = new MixerThread(this, output, id, *pDevices); 4984 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4985 } 4986 mPlaybackThreads.add(id, thread); 4987 4988 if (pSamplingRate) *pSamplingRate = samplingRate; 4989 if (pFormat) *pFormat = format; 4990 if (pChannels) *pChannels = channels; 4991 if (pLatencyMs) *pLatencyMs = thread->latency(); 4992 4993 // notify client processes of the new output creation 4994 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4995 return id; 4996 } 4997 4998 return 0; 4999} 5000 5001int AudioFlinger::openDuplicateOutput(int output1, int output2) 5002{ 5003 Mutex::Autolock _l(mLock); 5004 MixerThread *thread1 = checkMixerThread_l(output1); 5005 MixerThread *thread2 = checkMixerThread_l(output2); 5006 5007 if (thread1 == NULL || thread2 == NULL) { 5008 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5009 return 0; 5010 } 5011 5012 int id = nextUniqueId(); 5013 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5014 thread->addOutputTrack(thread2); 5015 mPlaybackThreads.add(id, thread); 5016 // notify client processes of the new output creation 5017 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5018 return id; 5019} 5020 5021status_t AudioFlinger::closeOutput(int output) 5022{ 5023 // keep strong reference on the playback thread so that 5024 // it is not destroyed while exit() is executed 5025 sp <PlaybackThread> thread; 5026 { 5027 Mutex::Autolock _l(mLock); 5028 thread = checkPlaybackThread_l(output); 5029 if (thread == NULL) { 5030 return BAD_VALUE; 5031 } 5032 5033 ALOGV("closeOutput() %d", output); 5034 5035 if (thread->type() == ThreadBase::MIXER) { 5036 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5037 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5038 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5039 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5040 } 5041 } 5042 } 5043 void *param2 = 0; 5044 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5045 mPlaybackThreads.removeItem(output); 5046 } 5047 thread->exit(); 5048 5049 if (thread->type() != ThreadBase::DUPLICATING) { 5050 AudioStreamOut *out = thread->clearOutput(); 5051 assert(out != NULL); 5052 // from now on thread->mOutput is NULL 5053 out->hwDev->close_output_stream(out->hwDev, out->stream); 5054 delete out; 5055 } 5056 return NO_ERROR; 5057} 5058 5059status_t AudioFlinger::suspendOutput(int output) 5060{ 5061 Mutex::Autolock _l(mLock); 5062 PlaybackThread *thread = checkPlaybackThread_l(output); 5063 5064 if (thread == NULL) { 5065 return BAD_VALUE; 5066 } 5067 5068 ALOGV("suspendOutput() %d", output); 5069 thread->suspend(); 5070 5071 return NO_ERROR; 5072} 5073 5074status_t AudioFlinger::restoreOutput(int output) 5075{ 5076 Mutex::Autolock _l(mLock); 5077 PlaybackThread *thread = checkPlaybackThread_l(output); 5078 5079 if (thread == NULL) { 5080 return BAD_VALUE; 5081 } 5082 5083 ALOGV("restoreOutput() %d", output); 5084 5085 thread->restore(); 5086 5087 return NO_ERROR; 5088} 5089 5090int AudioFlinger::openInput(uint32_t *pDevices, 5091 uint32_t *pSamplingRate, 5092 audio_format_t *pFormat, 5093 uint32_t *pChannels, 5094 uint32_t acoustics) 5095{ 5096 status_t status; 5097 RecordThread *thread = NULL; 5098 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5099 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5100 uint32_t channels = pChannels ? *pChannels : 0; 5101 uint32_t reqSamplingRate = samplingRate; 5102 audio_format_t reqFormat = format; 5103 uint32_t reqChannels = channels; 5104 audio_stream_in_t *inStream; 5105 audio_hw_device_t *inHwDev; 5106 5107 if (pDevices == NULL || *pDevices == 0) { 5108 return 0; 5109 } 5110 5111 Mutex::Autolock _l(mLock); 5112 5113 inHwDev = findSuitableHwDev_l(*pDevices); 5114 if (inHwDev == NULL) 5115 return 0; 5116 5117 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5118 &channels, &samplingRate, 5119 (audio_in_acoustics_t)acoustics, 5120 &inStream); 5121 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5122 inStream, 5123 samplingRate, 5124 format, 5125 channels, 5126 acoustics, 5127 status); 5128 5129 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5130 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5131 // or stereo to mono conversions on 16 bit PCM inputs. 5132 if (inStream == NULL && status == BAD_VALUE && 5133 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5134 (samplingRate <= 2 * reqSamplingRate) && 5135 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5136 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5137 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5138 &channels, &samplingRate, 5139 (audio_in_acoustics_t)acoustics, 5140 &inStream); 5141 } 5142 5143 if (inStream != NULL) { 5144 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5145 5146 int id = nextUniqueId(); 5147 // Start record thread 5148 // RecorThread require both input and output device indication to forward to audio 5149 // pre processing modules 5150 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5151 thread = new RecordThread(this, 5152 input, 5153 reqSamplingRate, 5154 reqChannels, 5155 id, 5156 device); 5157 mRecordThreads.add(id, thread); 5158 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5159 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5160 if (pFormat) *pFormat = format; 5161 if (pChannels) *pChannels = reqChannels; 5162 5163 input->stream->common.standby(&input->stream->common); 5164 5165 // notify client processes of the new input creation 5166 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5167 return id; 5168 } 5169 5170 return 0; 5171} 5172 5173status_t AudioFlinger::closeInput(int input) 5174{ 5175 // keep strong reference on the record thread so that 5176 // it is not destroyed while exit() is executed 5177 sp <RecordThread> thread; 5178 { 5179 Mutex::Autolock _l(mLock); 5180 thread = checkRecordThread_l(input); 5181 if (thread == NULL) { 5182 return BAD_VALUE; 5183 } 5184 5185 ALOGV("closeInput() %d", input); 5186 void *param2 = 0; 5187 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5188 mRecordThreads.removeItem(input); 5189 } 5190 thread->exit(); 5191 5192 AudioStreamIn *in = thread->clearInput(); 5193 assert(in != NULL); 5194 // from now on thread->mInput is NULL 5195 in->hwDev->close_input_stream(in->hwDev, in->stream); 5196 delete in; 5197 5198 return NO_ERROR; 5199} 5200 5201status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5202{ 5203 Mutex::Autolock _l(mLock); 5204 MixerThread *dstThread = checkMixerThread_l(output); 5205 if (dstThread == NULL) { 5206 ALOGW("setStreamOutput() bad output id %d", output); 5207 return BAD_VALUE; 5208 } 5209 5210 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5211 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5212 5213 dstThread->setStreamValid(stream, true); 5214 5215 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5216 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5217 if (thread != dstThread && 5218 thread->type() != ThreadBase::DIRECT) { 5219 MixerThread *srcThread = (MixerThread *)thread; 5220 srcThread->setStreamValid(stream, false); 5221 srcThread->invalidateTracks(stream); 5222 } 5223 } 5224 5225 return NO_ERROR; 5226} 5227 5228 5229int AudioFlinger::newAudioSessionId() 5230{ 5231 return nextUniqueId(); 5232} 5233 5234void AudioFlinger::acquireAudioSessionId(int audioSession) 5235{ 5236 Mutex::Autolock _l(mLock); 5237 int caller = IPCThreadState::self()->getCallingPid(); 5238 ALOGV("acquiring %d from %d", audioSession, caller); 5239 int num = mAudioSessionRefs.size(); 5240 for (int i = 0; i< num; i++) { 5241 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5242 if (ref->sessionid == audioSession && ref->pid == caller) { 5243 ref->cnt++; 5244 ALOGV(" incremented refcount to %d", ref->cnt); 5245 return; 5246 } 5247 } 5248 AudioSessionRef *ref = new AudioSessionRef(); 5249 ref->sessionid = audioSession; 5250 ref->pid = caller; 5251 ref->cnt = 1; 5252 mAudioSessionRefs.push(ref); 5253 ALOGV(" added new entry for %d", ref->sessionid); 5254} 5255 5256void AudioFlinger::releaseAudioSessionId(int audioSession) 5257{ 5258 Mutex::Autolock _l(mLock); 5259 int caller = IPCThreadState::self()->getCallingPid(); 5260 ALOGV("releasing %d from %d", audioSession, caller); 5261 int num = mAudioSessionRefs.size(); 5262 for (int i = 0; i< num; i++) { 5263 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5264 if (ref->sessionid == audioSession && ref->pid == caller) { 5265 ref->cnt--; 5266 ALOGV(" decremented refcount to %d", ref->cnt); 5267 if (ref->cnt == 0) { 5268 mAudioSessionRefs.removeAt(i); 5269 delete ref; 5270 purgeStaleEffects_l(); 5271 } 5272 return; 5273 } 5274 } 5275 ALOGW("session id %d not found for pid %d", audioSession, caller); 5276} 5277 5278void AudioFlinger::purgeStaleEffects_l() { 5279 5280 ALOGV("purging stale effects"); 5281 5282 Vector< sp<EffectChain> > chains; 5283 5284 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5285 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5286 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5287 sp<EffectChain> ec = t->mEffectChains[j]; 5288 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5289 chains.push(ec); 5290 } 5291 } 5292 } 5293 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5294 sp<RecordThread> t = mRecordThreads.valueAt(i); 5295 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5296 sp<EffectChain> ec = t->mEffectChains[j]; 5297 chains.push(ec); 5298 } 5299 } 5300 5301 for (size_t i = 0; i < chains.size(); i++) { 5302 sp<EffectChain> ec = chains[i]; 5303 int sessionid = ec->sessionId(); 5304 sp<ThreadBase> t = ec->mThread.promote(); 5305 if (t == 0) { 5306 continue; 5307 } 5308 size_t numsessionrefs = mAudioSessionRefs.size(); 5309 bool found = false; 5310 for (size_t k = 0; k < numsessionrefs; k++) { 5311 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5312 if (ref->sessionid == sessionid) { 5313 ALOGV(" session %d still exists for %d with %d refs", 5314 sessionid, ref->pid, ref->cnt); 5315 found = true; 5316 break; 5317 } 5318 } 5319 if (!found) { 5320 // remove all effects from the chain 5321 while (ec->mEffects.size()) { 5322 sp<EffectModule> effect = ec->mEffects[0]; 5323 effect->unPin(); 5324 Mutex::Autolock _l (t->mLock); 5325 t->removeEffect_l(effect); 5326 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5327 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5328 if (handle != 0) { 5329 handle->mEffect.clear(); 5330 if (handle->mHasControl && handle->mEnabled) { 5331 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5332 } 5333 } 5334 } 5335 AudioSystem::unregisterEffect(effect->id()); 5336 } 5337 } 5338 } 5339 return; 5340} 5341 5342// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5343AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5344{ 5345 PlaybackThread *thread = NULL; 5346 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5347 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5348 } 5349 return thread; 5350} 5351 5352// checkMixerThread_l() must be called with AudioFlinger::mLock held 5353AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5354{ 5355 PlaybackThread *thread = checkPlaybackThread_l(output); 5356 if (thread != NULL) { 5357 if (thread->type() == ThreadBase::DIRECT) { 5358 thread = NULL; 5359 } 5360 } 5361 return (MixerThread *)thread; 5362} 5363 5364// checkRecordThread_l() must be called with AudioFlinger::mLock held 5365AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5366{ 5367 RecordThread *thread = NULL; 5368 if (mRecordThreads.indexOfKey(input) >= 0) { 5369 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5370 } 5371 return thread; 5372} 5373 5374uint32_t AudioFlinger::nextUniqueId() 5375{ 5376 return android_atomic_inc(&mNextUniqueId); 5377} 5378 5379AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5380{ 5381 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5382 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5383 AudioStreamOut *output = thread->getOutput(); 5384 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5385 return thread; 5386 } 5387 } 5388 return NULL; 5389} 5390 5391uint32_t AudioFlinger::primaryOutputDevice_l() 5392{ 5393 PlaybackThread *thread = primaryPlaybackThread_l(); 5394 5395 if (thread == NULL) { 5396 return 0; 5397 } 5398 5399 return thread->device(); 5400} 5401 5402 5403// ---------------------------------------------------------------------------- 5404// Effect management 5405// ---------------------------------------------------------------------------- 5406 5407 5408status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5409{ 5410 Mutex::Autolock _l(mLock); 5411 return EffectQueryNumberEffects(numEffects); 5412} 5413 5414status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5415{ 5416 Mutex::Autolock _l(mLock); 5417 return EffectQueryEffect(index, descriptor); 5418} 5419 5420status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5421{ 5422 Mutex::Autolock _l(mLock); 5423 return EffectGetDescriptor(pUuid, descriptor); 5424} 5425 5426 5427sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5428 effect_descriptor_t *pDesc, 5429 const sp<IEffectClient>& effectClient, 5430 int32_t priority, 5431 int io, 5432 int sessionId, 5433 status_t *status, 5434 int *id, 5435 int *enabled) 5436{ 5437 status_t lStatus = NO_ERROR; 5438 sp<EffectHandle> handle; 5439 effect_descriptor_t desc; 5440 sp<Client> client; 5441 wp<Client> wclient; 5442 5443 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5444 pid, effectClient.get(), priority, sessionId, io); 5445 5446 if (pDesc == NULL) { 5447 lStatus = BAD_VALUE; 5448 goto Exit; 5449 } 5450 5451 // check audio settings permission for global effects 5452 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5453 lStatus = PERMISSION_DENIED; 5454 goto Exit; 5455 } 5456 5457 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5458 // that can only be created by audio policy manager (running in same process) 5459 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5460 lStatus = PERMISSION_DENIED; 5461 goto Exit; 5462 } 5463 5464 if (io == 0) { 5465 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5466 // output must be specified by AudioPolicyManager when using session 5467 // AUDIO_SESSION_OUTPUT_STAGE 5468 lStatus = BAD_VALUE; 5469 goto Exit; 5470 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5471 // if the output returned by getOutputForEffect() is removed before we lock the 5472 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5473 // and we will exit safely 5474 io = AudioSystem::getOutputForEffect(&desc); 5475 } 5476 } 5477 5478 { 5479 Mutex::Autolock _l(mLock); 5480 5481 5482 if (!EffectIsNullUuid(&pDesc->uuid)) { 5483 // if uuid is specified, request effect descriptor 5484 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5485 if (lStatus < 0) { 5486 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5487 goto Exit; 5488 } 5489 } else { 5490 // if uuid is not specified, look for an available implementation 5491 // of the required type in effect factory 5492 if (EffectIsNullUuid(&pDesc->type)) { 5493 ALOGW("createEffect() no effect type"); 5494 lStatus = BAD_VALUE; 5495 goto Exit; 5496 } 5497 uint32_t numEffects = 0; 5498 effect_descriptor_t d; 5499 d.flags = 0; // prevent compiler warning 5500 bool found = false; 5501 5502 lStatus = EffectQueryNumberEffects(&numEffects); 5503 if (lStatus < 0) { 5504 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5505 goto Exit; 5506 } 5507 for (uint32_t i = 0; i < numEffects; i++) { 5508 lStatus = EffectQueryEffect(i, &desc); 5509 if (lStatus < 0) { 5510 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5511 continue; 5512 } 5513 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5514 // If matching type found save effect descriptor. If the session is 5515 // 0 and the effect is not auxiliary, continue enumeration in case 5516 // an auxiliary version of this effect type is available 5517 found = true; 5518 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5519 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5520 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5521 break; 5522 } 5523 } 5524 } 5525 if (!found) { 5526 lStatus = BAD_VALUE; 5527 ALOGW("createEffect() effect not found"); 5528 goto Exit; 5529 } 5530 // For same effect type, chose auxiliary version over insert version if 5531 // connect to output mix (Compliance to OpenSL ES) 5532 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5533 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5534 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5535 } 5536 } 5537 5538 // Do not allow auxiliary effects on a session different from 0 (output mix) 5539 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5540 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5541 lStatus = INVALID_OPERATION; 5542 goto Exit; 5543 } 5544 5545 // check recording permission for visualizer 5546 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5547 !recordingAllowed()) { 5548 lStatus = PERMISSION_DENIED; 5549 goto Exit; 5550 } 5551 5552 // return effect descriptor 5553 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5554 5555 // If output is not specified try to find a matching audio session ID in one of the 5556 // output threads. 5557 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5558 // because of code checking output when entering the function. 5559 // Note: io is never 0 when creating an effect on an input 5560 if (io == 0) { 5561 // look for the thread where the specified audio session is present 5562 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5563 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5564 io = mPlaybackThreads.keyAt(i); 5565 break; 5566 } 5567 } 5568 if (io == 0) { 5569 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5570 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5571 io = mRecordThreads.keyAt(i); 5572 break; 5573 } 5574 } 5575 } 5576 // If no output thread contains the requested session ID, default to 5577 // first output. The effect chain will be moved to the correct output 5578 // thread when a track with the same session ID is created 5579 if (io == 0 && mPlaybackThreads.size()) { 5580 io = mPlaybackThreads.keyAt(0); 5581 } 5582 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5583 } 5584 ThreadBase *thread = checkRecordThread_l(io); 5585 if (thread == NULL) { 5586 thread = checkPlaybackThread_l(io); 5587 if (thread == NULL) { 5588 ALOGE("createEffect() unknown output thread"); 5589 lStatus = BAD_VALUE; 5590 goto Exit; 5591 } 5592 } 5593 5594 wclient = mClients.valueFor(pid); 5595 5596 if (wclient != NULL) { 5597 client = wclient.promote(); 5598 } else { 5599 client = new Client(this, pid); 5600 mClients.add(pid, client); 5601 } 5602 5603 // create effect on selected output thread 5604 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5605 &desc, enabled, &lStatus); 5606 if (handle != 0 && id != NULL) { 5607 *id = handle->id(); 5608 } 5609 } 5610 5611Exit: 5612 if(status) { 5613 *status = lStatus; 5614 } 5615 return handle; 5616} 5617 5618status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5619{ 5620 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5621 sessionId, srcOutput, dstOutput); 5622 Mutex::Autolock _l(mLock); 5623 if (srcOutput == dstOutput) { 5624 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5625 return NO_ERROR; 5626 } 5627 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5628 if (srcThread == NULL) { 5629 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5630 return BAD_VALUE; 5631 } 5632 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5633 if (dstThread == NULL) { 5634 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5635 return BAD_VALUE; 5636 } 5637 5638 Mutex::Autolock _dl(dstThread->mLock); 5639 Mutex::Autolock _sl(srcThread->mLock); 5640 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5641 5642 return NO_ERROR; 5643} 5644 5645// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5646status_t AudioFlinger::moveEffectChain_l(int sessionId, 5647 AudioFlinger::PlaybackThread *srcThread, 5648 AudioFlinger::PlaybackThread *dstThread, 5649 bool reRegister) 5650{ 5651 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5652 sessionId, srcThread, dstThread); 5653 5654 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5655 if (chain == 0) { 5656 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5657 sessionId, srcThread); 5658 return INVALID_OPERATION; 5659 } 5660 5661 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5662 // so that a new chain is created with correct parameters when first effect is added. This is 5663 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5664 // removed. 5665 srcThread->removeEffectChain_l(chain); 5666 5667 // transfer all effects one by one so that new effect chain is created on new thread with 5668 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5669 int dstOutput = dstThread->id(); 5670 sp<EffectChain> dstChain; 5671 uint32_t strategy = 0; // prevent compiler warning 5672 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5673 while (effect != 0) { 5674 srcThread->removeEffect_l(effect); 5675 dstThread->addEffect_l(effect); 5676 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5677 if (effect->state() == EffectModule::ACTIVE || 5678 effect->state() == EffectModule::STOPPING) { 5679 effect->start(); 5680 } 5681 // if the move request is not received from audio policy manager, the effect must be 5682 // re-registered with the new strategy and output 5683 if (dstChain == 0) { 5684 dstChain = effect->chain().promote(); 5685 if (dstChain == 0) { 5686 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5687 srcThread->addEffect_l(effect); 5688 return NO_INIT; 5689 } 5690 strategy = dstChain->strategy(); 5691 } 5692 if (reRegister) { 5693 AudioSystem::unregisterEffect(effect->id()); 5694 AudioSystem::registerEffect(&effect->desc(), 5695 dstOutput, 5696 strategy, 5697 sessionId, 5698 effect->id()); 5699 } 5700 effect = chain->getEffectFromId_l(0); 5701 } 5702 5703 return NO_ERROR; 5704} 5705 5706 5707// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5708sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5709 const sp<AudioFlinger::Client>& client, 5710 const sp<IEffectClient>& effectClient, 5711 int32_t priority, 5712 int sessionId, 5713 effect_descriptor_t *desc, 5714 int *enabled, 5715 status_t *status 5716 ) 5717{ 5718 sp<EffectModule> effect; 5719 sp<EffectHandle> handle; 5720 status_t lStatus; 5721 sp<EffectChain> chain; 5722 bool chainCreated = false; 5723 bool effectCreated = false; 5724 bool effectRegistered = false; 5725 5726 lStatus = initCheck(); 5727 if (lStatus != NO_ERROR) { 5728 ALOGW("createEffect_l() Audio driver not initialized."); 5729 goto Exit; 5730 } 5731 5732 // Do not allow effects with session ID 0 on direct output or duplicating threads 5733 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5734 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5735 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5736 desc->name, sessionId); 5737 lStatus = BAD_VALUE; 5738 goto Exit; 5739 } 5740 // Only Pre processor effects are allowed on input threads and only on input threads 5741 if ((mType == RECORD && 5742 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5743 (mType != RECORD && 5744 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5745 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5746 desc->name, desc->flags, mType); 5747 lStatus = BAD_VALUE; 5748 goto Exit; 5749 } 5750 5751 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5752 5753 { // scope for mLock 5754 Mutex::Autolock _l(mLock); 5755 5756 // check for existing effect chain with the requested audio session 5757 chain = getEffectChain_l(sessionId); 5758 if (chain == 0) { 5759 // create a new chain for this session 5760 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5761 chain = new EffectChain(this, sessionId); 5762 addEffectChain_l(chain); 5763 chain->setStrategy(getStrategyForSession_l(sessionId)); 5764 chainCreated = true; 5765 } else { 5766 effect = chain->getEffectFromDesc_l(desc); 5767 } 5768 5769 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5770 5771 if (effect == 0) { 5772 int id = mAudioFlinger->nextUniqueId(); 5773 // Check CPU and memory usage 5774 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5775 if (lStatus != NO_ERROR) { 5776 goto Exit; 5777 } 5778 effectRegistered = true; 5779 // create a new effect module if none present in the chain 5780 effect = new EffectModule(this, chain, desc, id, sessionId); 5781 lStatus = effect->status(); 5782 if (lStatus != NO_ERROR) { 5783 goto Exit; 5784 } 5785 lStatus = chain->addEffect_l(effect); 5786 if (lStatus != NO_ERROR) { 5787 goto Exit; 5788 } 5789 effectCreated = true; 5790 5791 effect->setDevice(mDevice); 5792 effect->setMode(mAudioFlinger->getMode()); 5793 } 5794 // create effect handle and connect it to effect module 5795 handle = new EffectHandle(effect, client, effectClient, priority); 5796 lStatus = effect->addHandle(handle); 5797 if (enabled) { 5798 *enabled = (int)effect->isEnabled(); 5799 } 5800 } 5801 5802Exit: 5803 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5804 Mutex::Autolock _l(mLock); 5805 if (effectCreated) { 5806 chain->removeEffect_l(effect); 5807 } 5808 if (effectRegistered) { 5809 AudioSystem::unregisterEffect(effect->id()); 5810 } 5811 if (chainCreated) { 5812 removeEffectChain_l(chain); 5813 } 5814 handle.clear(); 5815 } 5816 5817 if(status) { 5818 *status = lStatus; 5819 } 5820 return handle; 5821} 5822 5823sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5824{ 5825 sp<EffectModule> effect; 5826 5827 sp<EffectChain> chain = getEffectChain_l(sessionId); 5828 if (chain != 0) { 5829 effect = chain->getEffectFromId_l(effectId); 5830 } 5831 return effect; 5832} 5833 5834// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5835// PlaybackThread::mLock held 5836status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5837{ 5838 // check for existing effect chain with the requested audio session 5839 int sessionId = effect->sessionId(); 5840 sp<EffectChain> chain = getEffectChain_l(sessionId); 5841 bool chainCreated = false; 5842 5843 if (chain == 0) { 5844 // create a new chain for this session 5845 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5846 chain = new EffectChain(this, sessionId); 5847 addEffectChain_l(chain); 5848 chain->setStrategy(getStrategyForSession_l(sessionId)); 5849 chainCreated = true; 5850 } 5851 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5852 5853 if (chain->getEffectFromId_l(effect->id()) != 0) { 5854 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5855 this, effect->desc().name, chain.get()); 5856 return BAD_VALUE; 5857 } 5858 5859 status_t status = chain->addEffect_l(effect); 5860 if (status != NO_ERROR) { 5861 if (chainCreated) { 5862 removeEffectChain_l(chain); 5863 } 5864 return status; 5865 } 5866 5867 effect->setDevice(mDevice); 5868 effect->setMode(mAudioFlinger->getMode()); 5869 return NO_ERROR; 5870} 5871 5872void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5873 5874 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5875 effect_descriptor_t desc = effect->desc(); 5876 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5877 detachAuxEffect_l(effect->id()); 5878 } 5879 5880 sp<EffectChain> chain = effect->chain().promote(); 5881 if (chain != 0) { 5882 // remove effect chain if removing last effect 5883 if (chain->removeEffect_l(effect) == 0) { 5884 removeEffectChain_l(chain); 5885 } 5886 } else { 5887 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5888 } 5889} 5890 5891void AudioFlinger::ThreadBase::lockEffectChains_l( 5892 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5893{ 5894 effectChains = mEffectChains; 5895 for (size_t i = 0; i < mEffectChains.size(); i++) { 5896 mEffectChains[i]->lock(); 5897 } 5898} 5899 5900void AudioFlinger::ThreadBase::unlockEffectChains( 5901 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5902{ 5903 for (size_t i = 0; i < effectChains.size(); i++) { 5904 effectChains[i]->unlock(); 5905 } 5906} 5907 5908sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5909{ 5910 Mutex::Autolock _l(mLock); 5911 return getEffectChain_l(sessionId); 5912} 5913 5914sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5915{ 5916 sp<EffectChain> chain; 5917 5918 size_t size = mEffectChains.size(); 5919 for (size_t i = 0; i < size; i++) { 5920 if (mEffectChains[i]->sessionId() == sessionId) { 5921 chain = mEffectChains[i]; 5922 break; 5923 } 5924 } 5925 return chain; 5926} 5927 5928void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5929{ 5930 Mutex::Autolock _l(mLock); 5931 size_t size = mEffectChains.size(); 5932 for (size_t i = 0; i < size; i++) { 5933 mEffectChains[i]->setMode_l(mode); 5934 } 5935} 5936 5937void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5938 const wp<EffectHandle>& handle, 5939 bool unpiniflast) { 5940 5941 Mutex::Autolock _l(mLock); 5942 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5943 // delete the effect module if removing last handle on it 5944 if (effect->removeHandle(handle) == 0) { 5945 if (!effect->isPinned() || unpiniflast) { 5946 removeEffect_l(effect); 5947 AudioSystem::unregisterEffect(effect->id()); 5948 } 5949 } 5950} 5951 5952status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5953{ 5954 int session = chain->sessionId(); 5955 int16_t *buffer = mMixBuffer; 5956 bool ownsBuffer = false; 5957 5958 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5959 if (session > 0) { 5960 // Only one effect chain can be present in direct output thread and it uses 5961 // the mix buffer as input 5962 if (mType != DIRECT) { 5963 size_t numSamples = mFrameCount * mChannelCount; 5964 buffer = new int16_t[numSamples]; 5965 memset(buffer, 0, numSamples * sizeof(int16_t)); 5966 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5967 ownsBuffer = true; 5968 } 5969 5970 // Attach all tracks with same session ID to this chain. 5971 for (size_t i = 0; i < mTracks.size(); ++i) { 5972 sp<Track> track = mTracks[i]; 5973 if (session == track->sessionId()) { 5974 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5975 track->setMainBuffer(buffer); 5976 chain->incTrackCnt(); 5977 } 5978 } 5979 5980 // indicate all active tracks in the chain 5981 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5982 sp<Track> track = mActiveTracks[i].promote(); 5983 if (track == 0) continue; 5984 if (session == track->sessionId()) { 5985 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5986 chain->incActiveTrackCnt(); 5987 } 5988 } 5989 } 5990 5991 chain->setInBuffer(buffer, ownsBuffer); 5992 chain->setOutBuffer(mMixBuffer); 5993 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5994 // chains list in order to be processed last as it contains output stage effects 5995 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5996 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5997 // after track specific effects and before output stage 5998 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5999 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6000 // Effect chain for other sessions are inserted at beginning of effect 6001 // chains list to be processed before output mix effects. Relative order between other 6002 // sessions is not important 6003 size_t size = mEffectChains.size(); 6004 size_t i = 0; 6005 for (i = 0; i < size; i++) { 6006 if (mEffectChains[i]->sessionId() < session) break; 6007 } 6008 mEffectChains.insertAt(chain, i); 6009 checkSuspendOnAddEffectChain_l(chain); 6010 6011 return NO_ERROR; 6012} 6013 6014size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6015{ 6016 int session = chain->sessionId(); 6017 6018 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6019 6020 for (size_t i = 0; i < mEffectChains.size(); i++) { 6021 if (chain == mEffectChains[i]) { 6022 mEffectChains.removeAt(i); 6023 // detach all active tracks from the chain 6024 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6025 sp<Track> track = mActiveTracks[i].promote(); 6026 if (track == 0) continue; 6027 if (session == track->sessionId()) { 6028 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6029 chain.get(), session); 6030 chain->decActiveTrackCnt(); 6031 } 6032 } 6033 6034 // detach all tracks with same session ID from this chain 6035 for (size_t i = 0; i < mTracks.size(); ++i) { 6036 sp<Track> track = mTracks[i]; 6037 if (session == track->sessionId()) { 6038 track->setMainBuffer(mMixBuffer); 6039 chain->decTrackCnt(); 6040 } 6041 } 6042 break; 6043 } 6044 } 6045 return mEffectChains.size(); 6046} 6047 6048status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6049 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6050{ 6051 Mutex::Autolock _l(mLock); 6052 return attachAuxEffect_l(track, EffectId); 6053} 6054 6055status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6056 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6057{ 6058 status_t status = NO_ERROR; 6059 6060 if (EffectId == 0) { 6061 track->setAuxBuffer(0, NULL); 6062 } else { 6063 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6064 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6065 if (effect != 0) { 6066 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6067 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6068 } else { 6069 status = INVALID_OPERATION; 6070 } 6071 } else { 6072 status = BAD_VALUE; 6073 } 6074 } 6075 return status; 6076} 6077 6078void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6079{ 6080 for (size_t i = 0; i < mTracks.size(); ++i) { 6081 sp<Track> track = mTracks[i]; 6082 if (track->auxEffectId() == effectId) { 6083 attachAuxEffect_l(track, 0); 6084 } 6085 } 6086} 6087 6088status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6089{ 6090 // only one chain per input thread 6091 if (mEffectChains.size() != 0) { 6092 return INVALID_OPERATION; 6093 } 6094 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6095 6096 chain->setInBuffer(NULL); 6097 chain->setOutBuffer(NULL); 6098 6099 checkSuspendOnAddEffectChain_l(chain); 6100 6101 mEffectChains.add(chain); 6102 6103 return NO_ERROR; 6104} 6105 6106size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6107{ 6108 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6109 ALOGW_IF(mEffectChains.size() != 1, 6110 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6111 chain.get(), mEffectChains.size(), this); 6112 if (mEffectChains.size() == 1) { 6113 mEffectChains.removeAt(0); 6114 } 6115 return 0; 6116} 6117 6118// ---------------------------------------------------------------------------- 6119// EffectModule implementation 6120// ---------------------------------------------------------------------------- 6121 6122#undef LOG_TAG 6123#define LOG_TAG "AudioFlinger::EffectModule" 6124 6125AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6126 const wp<AudioFlinger::EffectChain>& chain, 6127 effect_descriptor_t *desc, 6128 int id, 6129 int sessionId) 6130 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6131 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6132{ 6133 ALOGV("Constructor %p", this); 6134 int lStatus; 6135 sp<ThreadBase> thread = mThread.promote(); 6136 if (thread == 0) { 6137 return; 6138 } 6139 6140 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6141 6142 // create effect engine from effect factory 6143 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6144 6145 if (mStatus != NO_ERROR) { 6146 return; 6147 } 6148 lStatus = init(); 6149 if (lStatus < 0) { 6150 mStatus = lStatus; 6151 goto Error; 6152 } 6153 6154 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6155 mPinned = true; 6156 } 6157 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6158 return; 6159Error: 6160 EffectRelease(mEffectInterface); 6161 mEffectInterface = NULL; 6162 ALOGV("Constructor Error %d", mStatus); 6163} 6164 6165AudioFlinger::EffectModule::~EffectModule() 6166{ 6167 ALOGV("Destructor %p", this); 6168 if (mEffectInterface != NULL) { 6169 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6170 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6171 sp<ThreadBase> thread = mThread.promote(); 6172 if (thread != 0) { 6173 audio_stream_t *stream = thread->stream(); 6174 if (stream != NULL) { 6175 stream->remove_audio_effect(stream, mEffectInterface); 6176 } 6177 } 6178 } 6179 // release effect engine 6180 EffectRelease(mEffectInterface); 6181 } 6182} 6183 6184status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6185{ 6186 status_t status; 6187 6188 Mutex::Autolock _l(mLock); 6189 // First handle in mHandles has highest priority and controls the effect module 6190 int priority = handle->priority(); 6191 size_t size = mHandles.size(); 6192 sp<EffectHandle> h; 6193 size_t i; 6194 for (i = 0; i < size; i++) { 6195 h = mHandles[i].promote(); 6196 if (h == 0) continue; 6197 if (h->priority() <= priority) break; 6198 } 6199 // if inserted in first place, move effect control from previous owner to this handle 6200 if (i == 0) { 6201 bool enabled = false; 6202 if (h != 0) { 6203 enabled = h->enabled(); 6204 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6205 } 6206 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6207 status = NO_ERROR; 6208 } else { 6209 status = ALREADY_EXISTS; 6210 } 6211 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6212 mHandles.insertAt(handle, i); 6213 return status; 6214} 6215 6216size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6217{ 6218 Mutex::Autolock _l(mLock); 6219 size_t size = mHandles.size(); 6220 size_t i; 6221 for (i = 0; i < size; i++) { 6222 if (mHandles[i] == handle) break; 6223 } 6224 if (i == size) { 6225 return size; 6226 } 6227 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6228 6229 bool enabled = false; 6230 EffectHandle *hdl = handle.unsafe_get(); 6231 if (hdl) { 6232 ALOGV("removeHandle() unsafe_get OK"); 6233 enabled = hdl->enabled(); 6234 } 6235 mHandles.removeAt(i); 6236 size = mHandles.size(); 6237 // if removed from first place, move effect control from this handle to next in line 6238 if (i == 0 && size != 0) { 6239 sp<EffectHandle> h = mHandles[0].promote(); 6240 if (h != 0) { 6241 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6242 } 6243 } 6244 6245 // Prevent calls to process() and other functions on effect interface from now on. 6246 // The effect engine will be released by the destructor when the last strong reference on 6247 // this object is released which can happen after next process is called. 6248 if (size == 0 && !mPinned) { 6249 mState = DESTROYED; 6250 } 6251 6252 return size; 6253} 6254 6255sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6256{ 6257 Mutex::Autolock _l(mLock); 6258 sp<EffectHandle> handle; 6259 if (mHandles.size() != 0) { 6260 handle = mHandles[0].promote(); 6261 } 6262 return handle; 6263} 6264 6265void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6266{ 6267 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6268 // keep a strong reference on this EffectModule to avoid calling the 6269 // destructor before we exit 6270 sp<EffectModule> keep(this); 6271 { 6272 sp<ThreadBase> thread = mThread.promote(); 6273 if (thread != 0) { 6274 thread->disconnectEffect(keep, handle, unpiniflast); 6275 } 6276 } 6277} 6278 6279void AudioFlinger::EffectModule::updateState() { 6280 Mutex::Autolock _l(mLock); 6281 6282 switch (mState) { 6283 case RESTART: 6284 reset_l(); 6285 // FALL THROUGH 6286 6287 case STARTING: 6288 // clear auxiliary effect input buffer for next accumulation 6289 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6290 memset(mConfig.inputCfg.buffer.raw, 6291 0, 6292 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6293 } 6294 start_l(); 6295 mState = ACTIVE; 6296 break; 6297 case STOPPING: 6298 stop_l(); 6299 mDisableWaitCnt = mMaxDisableWaitCnt; 6300 mState = STOPPED; 6301 break; 6302 case STOPPED: 6303 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6304 // turn off sequence. 6305 if (--mDisableWaitCnt == 0) { 6306 reset_l(); 6307 mState = IDLE; 6308 } 6309 break; 6310 default: //IDLE , ACTIVE, DESTROYED 6311 break; 6312 } 6313} 6314 6315void AudioFlinger::EffectModule::process() 6316{ 6317 Mutex::Autolock _l(mLock); 6318 6319 if (mState == DESTROYED || mEffectInterface == NULL || 6320 mConfig.inputCfg.buffer.raw == NULL || 6321 mConfig.outputCfg.buffer.raw == NULL) { 6322 return; 6323 } 6324 6325 if (isProcessEnabled()) { 6326 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6327 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6328 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6329 mConfig.inputCfg.buffer.s32, 6330 mConfig.inputCfg.buffer.frameCount/2); 6331 } 6332 6333 // do the actual processing in the effect engine 6334 int ret = (*mEffectInterface)->process(mEffectInterface, 6335 &mConfig.inputCfg.buffer, 6336 &mConfig.outputCfg.buffer); 6337 6338 // force transition to IDLE state when engine is ready 6339 if (mState == STOPPED && ret == -ENODATA) { 6340 mDisableWaitCnt = 1; 6341 } 6342 6343 // clear auxiliary effect input buffer for next accumulation 6344 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6345 memset(mConfig.inputCfg.buffer.raw, 0, 6346 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6347 } 6348 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6349 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6350 // If an insert effect is idle and input buffer is different from output buffer, 6351 // accumulate input onto output 6352 sp<EffectChain> chain = mChain.promote(); 6353 if (chain != 0 && chain->activeTrackCnt() != 0) { 6354 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6355 int16_t *in = mConfig.inputCfg.buffer.s16; 6356 int16_t *out = mConfig.outputCfg.buffer.s16; 6357 for (size_t i = 0; i < frameCnt; i++) { 6358 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6359 } 6360 } 6361 } 6362} 6363 6364void AudioFlinger::EffectModule::reset_l() 6365{ 6366 if (mEffectInterface == NULL) { 6367 return; 6368 } 6369 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6370} 6371 6372status_t AudioFlinger::EffectModule::configure() 6373{ 6374 uint32_t channels; 6375 if (mEffectInterface == NULL) { 6376 return NO_INIT; 6377 } 6378 6379 sp<ThreadBase> thread = mThread.promote(); 6380 if (thread == 0) { 6381 return DEAD_OBJECT; 6382 } 6383 6384 // TODO: handle configuration of effects replacing track process 6385 if (thread->channelCount() == 1) { 6386 channels = AUDIO_CHANNEL_OUT_MONO; 6387 } else { 6388 channels = AUDIO_CHANNEL_OUT_STEREO; 6389 } 6390 6391 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6392 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6393 } else { 6394 mConfig.inputCfg.channels = channels; 6395 } 6396 mConfig.outputCfg.channels = channels; 6397 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6398 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6399 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6400 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6401 mConfig.inputCfg.bufferProvider.cookie = NULL; 6402 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6403 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6404 mConfig.outputCfg.bufferProvider.cookie = NULL; 6405 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6406 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6407 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6408 // Insert effect: 6409 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6410 // always overwrites output buffer: input buffer == output buffer 6411 // - in other sessions: 6412 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6413 // other effect: overwrites output buffer: input buffer == output buffer 6414 // Auxiliary effect: 6415 // accumulates in output buffer: input buffer != output buffer 6416 // Therefore: accumulate <=> input buffer != output buffer 6417 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6418 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6419 } else { 6420 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6421 } 6422 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6423 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6424 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6425 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6426 6427 ALOGV("configure() %p thread %p buffer %p framecount %d", 6428 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6429 6430 status_t cmdStatus; 6431 uint32_t size = sizeof(int); 6432 status_t status = (*mEffectInterface)->command(mEffectInterface, 6433 EFFECT_CMD_SET_CONFIG, 6434 sizeof(effect_config_t), 6435 &mConfig, 6436 &size, 6437 &cmdStatus); 6438 if (status == 0) { 6439 status = cmdStatus; 6440 } 6441 6442 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6443 (1000 * mConfig.outputCfg.buffer.frameCount); 6444 6445 return status; 6446} 6447 6448status_t AudioFlinger::EffectModule::init() 6449{ 6450 Mutex::Autolock _l(mLock); 6451 if (mEffectInterface == NULL) { 6452 return NO_INIT; 6453 } 6454 status_t cmdStatus; 6455 uint32_t size = sizeof(status_t); 6456 status_t status = (*mEffectInterface)->command(mEffectInterface, 6457 EFFECT_CMD_INIT, 6458 0, 6459 NULL, 6460 &size, 6461 &cmdStatus); 6462 if (status == 0) { 6463 status = cmdStatus; 6464 } 6465 return status; 6466} 6467 6468status_t AudioFlinger::EffectModule::start() 6469{ 6470 Mutex::Autolock _l(mLock); 6471 return start_l(); 6472} 6473 6474status_t AudioFlinger::EffectModule::start_l() 6475{ 6476 if (mEffectInterface == NULL) { 6477 return NO_INIT; 6478 } 6479 status_t cmdStatus; 6480 uint32_t size = sizeof(status_t); 6481 status_t status = (*mEffectInterface)->command(mEffectInterface, 6482 EFFECT_CMD_ENABLE, 6483 0, 6484 NULL, 6485 &size, 6486 &cmdStatus); 6487 if (status == 0) { 6488 status = cmdStatus; 6489 } 6490 if (status == 0 && 6491 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6492 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6493 sp<ThreadBase> thread = mThread.promote(); 6494 if (thread != 0) { 6495 audio_stream_t *stream = thread->stream(); 6496 if (stream != NULL) { 6497 stream->add_audio_effect(stream, mEffectInterface); 6498 } 6499 } 6500 } 6501 return status; 6502} 6503 6504status_t AudioFlinger::EffectModule::stop() 6505{ 6506 Mutex::Autolock _l(mLock); 6507 return stop_l(); 6508} 6509 6510status_t AudioFlinger::EffectModule::stop_l() 6511{ 6512 if (mEffectInterface == NULL) { 6513 return NO_INIT; 6514 } 6515 status_t cmdStatus; 6516 uint32_t size = sizeof(status_t); 6517 status_t status = (*mEffectInterface)->command(mEffectInterface, 6518 EFFECT_CMD_DISABLE, 6519 0, 6520 NULL, 6521 &size, 6522 &cmdStatus); 6523 if (status == 0) { 6524 status = cmdStatus; 6525 } 6526 if (status == 0 && 6527 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6528 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6529 sp<ThreadBase> thread = mThread.promote(); 6530 if (thread != 0) { 6531 audio_stream_t *stream = thread->stream(); 6532 if (stream != NULL) { 6533 stream->remove_audio_effect(stream, mEffectInterface); 6534 } 6535 } 6536 } 6537 return status; 6538} 6539 6540status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6541 uint32_t cmdSize, 6542 void *pCmdData, 6543 uint32_t *replySize, 6544 void *pReplyData) 6545{ 6546 Mutex::Autolock _l(mLock); 6547// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6548 6549 if (mState == DESTROYED || mEffectInterface == NULL) { 6550 return NO_INIT; 6551 } 6552 status_t status = (*mEffectInterface)->command(mEffectInterface, 6553 cmdCode, 6554 cmdSize, 6555 pCmdData, 6556 replySize, 6557 pReplyData); 6558 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6559 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6560 for (size_t i = 1; i < mHandles.size(); i++) { 6561 sp<EffectHandle> h = mHandles[i].promote(); 6562 if (h != 0) { 6563 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6564 } 6565 } 6566 } 6567 return status; 6568} 6569 6570status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6571{ 6572 6573 Mutex::Autolock _l(mLock); 6574 ALOGV("setEnabled %p enabled %d", this, enabled); 6575 6576 if (enabled != isEnabled()) { 6577 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6578 if (enabled && status != NO_ERROR) { 6579 return status; 6580 } 6581 6582 switch (mState) { 6583 // going from disabled to enabled 6584 case IDLE: 6585 mState = STARTING; 6586 break; 6587 case STOPPED: 6588 mState = RESTART; 6589 break; 6590 case STOPPING: 6591 mState = ACTIVE; 6592 break; 6593 6594 // going from enabled to disabled 6595 case RESTART: 6596 mState = STOPPED; 6597 break; 6598 case STARTING: 6599 mState = IDLE; 6600 break; 6601 case ACTIVE: 6602 mState = STOPPING; 6603 break; 6604 case DESTROYED: 6605 return NO_ERROR; // simply ignore as we are being destroyed 6606 } 6607 for (size_t i = 1; i < mHandles.size(); i++) { 6608 sp<EffectHandle> h = mHandles[i].promote(); 6609 if (h != 0) { 6610 h->setEnabled(enabled); 6611 } 6612 } 6613 } 6614 return NO_ERROR; 6615} 6616 6617bool AudioFlinger::EffectModule::isEnabled() 6618{ 6619 switch (mState) { 6620 case RESTART: 6621 case STARTING: 6622 case ACTIVE: 6623 return true; 6624 case IDLE: 6625 case STOPPING: 6626 case STOPPED: 6627 case DESTROYED: 6628 default: 6629 return false; 6630 } 6631} 6632 6633bool AudioFlinger::EffectModule::isProcessEnabled() 6634{ 6635 switch (mState) { 6636 case RESTART: 6637 case ACTIVE: 6638 case STOPPING: 6639 case STOPPED: 6640 return true; 6641 case IDLE: 6642 case STARTING: 6643 case DESTROYED: 6644 default: 6645 return false; 6646 } 6647} 6648 6649status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6650{ 6651 Mutex::Autolock _l(mLock); 6652 status_t status = NO_ERROR; 6653 6654 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6655 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6656 if (isProcessEnabled() && 6657 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6658 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6659 status_t cmdStatus; 6660 uint32_t volume[2]; 6661 uint32_t *pVolume = NULL; 6662 uint32_t size = sizeof(volume); 6663 volume[0] = *left; 6664 volume[1] = *right; 6665 if (controller) { 6666 pVolume = volume; 6667 } 6668 status = (*mEffectInterface)->command(mEffectInterface, 6669 EFFECT_CMD_SET_VOLUME, 6670 size, 6671 volume, 6672 &size, 6673 pVolume); 6674 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6675 *left = volume[0]; 6676 *right = volume[1]; 6677 } 6678 } 6679 return status; 6680} 6681 6682status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6683{ 6684 Mutex::Autolock _l(mLock); 6685 status_t status = NO_ERROR; 6686 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6687 // audio pre processing modules on RecordThread can receive both output and 6688 // input device indication in the same call 6689 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6690 if (dev) { 6691 status_t cmdStatus; 6692 uint32_t size = sizeof(status_t); 6693 6694 status = (*mEffectInterface)->command(mEffectInterface, 6695 EFFECT_CMD_SET_DEVICE, 6696 sizeof(uint32_t), 6697 &dev, 6698 &size, 6699 &cmdStatus); 6700 if (status == NO_ERROR) { 6701 status = cmdStatus; 6702 } 6703 } 6704 dev = device & AUDIO_DEVICE_IN_ALL; 6705 if (dev) { 6706 status_t cmdStatus; 6707 uint32_t size = sizeof(status_t); 6708 6709 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6710 EFFECT_CMD_SET_INPUT_DEVICE, 6711 sizeof(uint32_t), 6712 &dev, 6713 &size, 6714 &cmdStatus); 6715 if (status2 == NO_ERROR) { 6716 status2 = cmdStatus; 6717 } 6718 if (status == NO_ERROR) { 6719 status = status2; 6720 } 6721 } 6722 } 6723 return status; 6724} 6725 6726status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6727{ 6728 Mutex::Autolock _l(mLock); 6729 status_t status = NO_ERROR; 6730 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6731 status_t cmdStatus; 6732 uint32_t size = sizeof(status_t); 6733 status = (*mEffectInterface)->command(mEffectInterface, 6734 EFFECT_CMD_SET_AUDIO_MODE, 6735 sizeof(audio_mode_t), 6736 &mode, 6737 &size, 6738 &cmdStatus); 6739 if (status == NO_ERROR) { 6740 status = cmdStatus; 6741 } 6742 } 6743 return status; 6744} 6745 6746void AudioFlinger::EffectModule::setSuspended(bool suspended) 6747{ 6748 Mutex::Autolock _l(mLock); 6749 mSuspended = suspended; 6750} 6751 6752bool AudioFlinger::EffectModule::suspended() const 6753{ 6754 Mutex::Autolock _l(mLock); 6755 return mSuspended; 6756} 6757 6758status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6759{ 6760 const size_t SIZE = 256; 6761 char buffer[SIZE]; 6762 String8 result; 6763 6764 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6765 result.append(buffer); 6766 6767 bool locked = tryLock(mLock); 6768 // failed to lock - AudioFlinger is probably deadlocked 6769 if (!locked) { 6770 result.append("\t\tCould not lock Fx mutex:\n"); 6771 } 6772 6773 result.append("\t\tSession Status State Engine:\n"); 6774 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6775 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6776 result.append(buffer); 6777 6778 result.append("\t\tDescriptor:\n"); 6779 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6780 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6781 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6782 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6783 result.append(buffer); 6784 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6785 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6786 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6787 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6788 result.append(buffer); 6789 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6790 mDescriptor.apiVersion, 6791 mDescriptor.flags); 6792 result.append(buffer); 6793 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6794 mDescriptor.name); 6795 result.append(buffer); 6796 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6797 mDescriptor.implementor); 6798 result.append(buffer); 6799 6800 result.append("\t\t- Input configuration:\n"); 6801 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6802 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6803 (uint32_t)mConfig.inputCfg.buffer.raw, 6804 mConfig.inputCfg.buffer.frameCount, 6805 mConfig.inputCfg.samplingRate, 6806 mConfig.inputCfg.channels, 6807 mConfig.inputCfg.format); 6808 result.append(buffer); 6809 6810 result.append("\t\t- Output configuration:\n"); 6811 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6812 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6813 (uint32_t)mConfig.outputCfg.buffer.raw, 6814 mConfig.outputCfg.buffer.frameCount, 6815 mConfig.outputCfg.samplingRate, 6816 mConfig.outputCfg.channels, 6817 mConfig.outputCfg.format); 6818 result.append(buffer); 6819 6820 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6821 result.append(buffer); 6822 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6823 for (size_t i = 0; i < mHandles.size(); ++i) { 6824 sp<EffectHandle> handle = mHandles[i].promote(); 6825 if (handle != 0) { 6826 handle->dump(buffer, SIZE); 6827 result.append(buffer); 6828 } 6829 } 6830 6831 result.append("\n"); 6832 6833 write(fd, result.string(), result.length()); 6834 6835 if (locked) { 6836 mLock.unlock(); 6837 } 6838 6839 return NO_ERROR; 6840} 6841 6842// ---------------------------------------------------------------------------- 6843// EffectHandle implementation 6844// ---------------------------------------------------------------------------- 6845 6846#undef LOG_TAG 6847#define LOG_TAG "AudioFlinger::EffectHandle" 6848 6849AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6850 const sp<AudioFlinger::Client>& client, 6851 const sp<IEffectClient>& effectClient, 6852 int32_t priority) 6853 : BnEffect(), 6854 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6855 mPriority(priority), mHasControl(false), mEnabled(false) 6856{ 6857 ALOGV("constructor %p", this); 6858 6859 if (client == 0) { 6860 return; 6861 } 6862 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6863 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6864 if (mCblkMemory != 0) { 6865 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6866 6867 if (mCblk) { 6868 new(mCblk) effect_param_cblk_t(); 6869 mBuffer = (uint8_t *)mCblk + bufOffset; 6870 } 6871 } else { 6872 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6873 return; 6874 } 6875} 6876 6877AudioFlinger::EffectHandle::~EffectHandle() 6878{ 6879 ALOGV("Destructor %p", this); 6880 disconnect(false); 6881 ALOGV("Destructor DONE %p", this); 6882} 6883 6884status_t AudioFlinger::EffectHandle::enable() 6885{ 6886 ALOGV("enable %p", this); 6887 if (!mHasControl) return INVALID_OPERATION; 6888 if (mEffect == 0) return DEAD_OBJECT; 6889 6890 if (mEnabled) { 6891 return NO_ERROR; 6892 } 6893 6894 mEnabled = true; 6895 6896 sp<ThreadBase> thread = mEffect->thread().promote(); 6897 if (thread != 0) { 6898 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6899 } 6900 6901 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6902 if (mEffect->suspended()) { 6903 return NO_ERROR; 6904 } 6905 6906 status_t status = mEffect->setEnabled(true); 6907 if (status != NO_ERROR) { 6908 if (thread != 0) { 6909 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6910 } 6911 mEnabled = false; 6912 } 6913 return status; 6914} 6915 6916status_t AudioFlinger::EffectHandle::disable() 6917{ 6918 ALOGV("disable %p", this); 6919 if (!mHasControl) return INVALID_OPERATION; 6920 if (mEffect == 0) return DEAD_OBJECT; 6921 6922 if (!mEnabled) { 6923 return NO_ERROR; 6924 } 6925 mEnabled = false; 6926 6927 if (mEffect->suspended()) { 6928 return NO_ERROR; 6929 } 6930 6931 status_t status = mEffect->setEnabled(false); 6932 6933 sp<ThreadBase> thread = mEffect->thread().promote(); 6934 if (thread != 0) { 6935 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6936 } 6937 6938 return status; 6939} 6940 6941void AudioFlinger::EffectHandle::disconnect() 6942{ 6943 disconnect(true); 6944} 6945 6946void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6947{ 6948 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6949 if (mEffect == 0) { 6950 return; 6951 } 6952 mEffect->disconnect(this, unpiniflast); 6953 6954 if (mHasControl && mEnabled) { 6955 sp<ThreadBase> thread = mEffect->thread().promote(); 6956 if (thread != 0) { 6957 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6958 } 6959 } 6960 6961 // release sp on module => module destructor can be called now 6962 mEffect.clear(); 6963 if (mClient != 0) { 6964 if (mCblk) { 6965 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6966 } 6967 mCblkMemory.clear(); // and free the shared memory 6968 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6969 mClient.clear(); 6970 } 6971} 6972 6973status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6974 uint32_t cmdSize, 6975 void *pCmdData, 6976 uint32_t *replySize, 6977 void *pReplyData) 6978{ 6979// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6980// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6981 6982 // only get parameter command is permitted for applications not controlling the effect 6983 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6984 return INVALID_OPERATION; 6985 } 6986 if (mEffect == 0) return DEAD_OBJECT; 6987 if (mClient == 0) return INVALID_OPERATION; 6988 6989 // handle commands that are not forwarded transparently to effect engine 6990 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6991 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6992 // no risk to block the whole media server process or mixer threads is we are stuck here 6993 Mutex::Autolock _l(mCblk->lock); 6994 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6995 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6996 mCblk->serverIndex = 0; 6997 mCblk->clientIndex = 0; 6998 return BAD_VALUE; 6999 } 7000 status_t status = NO_ERROR; 7001 while (mCblk->serverIndex < mCblk->clientIndex) { 7002 int reply; 7003 uint32_t rsize = sizeof(int); 7004 int *p = (int *)(mBuffer + mCblk->serverIndex); 7005 int size = *p++; 7006 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7007 ALOGW("command(): invalid parameter block size"); 7008 break; 7009 } 7010 effect_param_t *param = (effect_param_t *)p; 7011 if (param->psize == 0 || param->vsize == 0) { 7012 ALOGW("command(): null parameter or value size"); 7013 mCblk->serverIndex += size; 7014 continue; 7015 } 7016 uint32_t psize = sizeof(effect_param_t) + 7017 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7018 param->vsize; 7019 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7020 psize, 7021 p, 7022 &rsize, 7023 &reply); 7024 // stop at first error encountered 7025 if (ret != NO_ERROR) { 7026 status = ret; 7027 *(int *)pReplyData = reply; 7028 break; 7029 } else if (reply != NO_ERROR) { 7030 *(int *)pReplyData = reply; 7031 break; 7032 } 7033 mCblk->serverIndex += size; 7034 } 7035 mCblk->serverIndex = 0; 7036 mCblk->clientIndex = 0; 7037 return status; 7038 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7039 *(int *)pReplyData = NO_ERROR; 7040 return enable(); 7041 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7042 *(int *)pReplyData = NO_ERROR; 7043 return disable(); 7044 } 7045 7046 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7047} 7048 7049sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7050 return mCblkMemory; 7051} 7052 7053void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7054{ 7055 ALOGV("setControl %p control %d", this, hasControl); 7056 7057 mHasControl = hasControl; 7058 mEnabled = enabled; 7059 7060 if (signal && mEffectClient != 0) { 7061 mEffectClient->controlStatusChanged(hasControl); 7062 } 7063} 7064 7065void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7066 uint32_t cmdSize, 7067 void *pCmdData, 7068 uint32_t replySize, 7069 void *pReplyData) 7070{ 7071 if (mEffectClient != 0) { 7072 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7073 } 7074} 7075 7076 7077 7078void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7079{ 7080 if (mEffectClient != 0) { 7081 mEffectClient->enableStatusChanged(enabled); 7082 } 7083} 7084 7085status_t AudioFlinger::EffectHandle::onTransact( 7086 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7087{ 7088 return BnEffect::onTransact(code, data, reply, flags); 7089} 7090 7091 7092void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7093{ 7094 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7095 7096 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7097 (mClient == NULL) ? getpid() : mClient->pid(), 7098 mPriority, 7099 mHasControl, 7100 !locked, 7101 mCblk ? mCblk->clientIndex : 0, 7102 mCblk ? mCblk->serverIndex : 0 7103 ); 7104 7105 if (locked) { 7106 mCblk->lock.unlock(); 7107 } 7108} 7109 7110#undef LOG_TAG 7111#define LOG_TAG "AudioFlinger::EffectChain" 7112 7113AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7114 int sessionId) 7115 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7116 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7117 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7118{ 7119 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7120 sp<ThreadBase> thread = mThread.promote(); 7121 if (thread == 0) { 7122 return; 7123 } 7124 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7125 thread->frameCount(); 7126} 7127 7128AudioFlinger::EffectChain::~EffectChain() 7129{ 7130 if (mOwnInBuffer) { 7131 delete mInBuffer; 7132 } 7133 7134} 7135 7136// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7137sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7138{ 7139 sp<EffectModule> effect; 7140 size_t size = mEffects.size(); 7141 7142 for (size_t i = 0; i < size; i++) { 7143 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7144 effect = mEffects[i]; 7145 break; 7146 } 7147 } 7148 return effect; 7149} 7150 7151// getEffectFromId_l() must be called with ThreadBase::mLock held 7152sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7153{ 7154 sp<EffectModule> effect; 7155 size_t size = mEffects.size(); 7156 7157 for (size_t i = 0; i < size; i++) { 7158 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7159 if (id == 0 || mEffects[i]->id() == id) { 7160 effect = mEffects[i]; 7161 break; 7162 } 7163 } 7164 return effect; 7165} 7166 7167// getEffectFromType_l() must be called with ThreadBase::mLock held 7168sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7169 const effect_uuid_t *type) 7170{ 7171 sp<EffectModule> effect; 7172 size_t size = mEffects.size(); 7173 7174 for (size_t i = 0; i < size; i++) { 7175 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7176 effect = mEffects[i]; 7177 break; 7178 } 7179 } 7180 return effect; 7181} 7182 7183// Must be called with EffectChain::mLock locked 7184void AudioFlinger::EffectChain::process_l() 7185{ 7186 sp<ThreadBase> thread = mThread.promote(); 7187 if (thread == 0) { 7188 ALOGW("process_l(): cannot promote mixer thread"); 7189 return; 7190 } 7191 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7192 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7193 // always process effects unless no more tracks are on the session and the effect tail 7194 // has been rendered 7195 bool doProcess = true; 7196 if (!isGlobalSession) { 7197 bool tracksOnSession = (trackCnt() != 0); 7198 7199 if (!tracksOnSession && mTailBufferCount == 0) { 7200 doProcess = false; 7201 } 7202 7203 if (activeTrackCnt() == 0) { 7204 // if no track is active and the effect tail has not been rendered, 7205 // the input buffer must be cleared here as the mixer process will not do it 7206 if (tracksOnSession || mTailBufferCount > 0) { 7207 size_t numSamples = thread->frameCount() * thread->channelCount(); 7208 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7209 if (mTailBufferCount > 0) { 7210 mTailBufferCount--; 7211 } 7212 } 7213 } 7214 } 7215 7216 size_t size = mEffects.size(); 7217 if (doProcess) { 7218 for (size_t i = 0; i < size; i++) { 7219 mEffects[i]->process(); 7220 } 7221 } 7222 for (size_t i = 0; i < size; i++) { 7223 mEffects[i]->updateState(); 7224 } 7225} 7226 7227// addEffect_l() must be called with PlaybackThread::mLock held 7228status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7229{ 7230 effect_descriptor_t desc = effect->desc(); 7231 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7232 7233 Mutex::Autolock _l(mLock); 7234 effect->setChain(this); 7235 sp<ThreadBase> thread = mThread.promote(); 7236 if (thread == 0) { 7237 return NO_INIT; 7238 } 7239 effect->setThread(thread); 7240 7241 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7242 // Auxiliary effects are inserted at the beginning of mEffects vector as 7243 // they are processed first and accumulated in chain input buffer 7244 mEffects.insertAt(effect, 0); 7245 7246 // the input buffer for auxiliary effect contains mono samples in 7247 // 32 bit format. This is to avoid saturation in AudoMixer 7248 // accumulation stage. Saturation is done in EffectModule::process() before 7249 // calling the process in effect engine 7250 size_t numSamples = thread->frameCount(); 7251 int32_t *buffer = new int32_t[numSamples]; 7252 memset(buffer, 0, numSamples * sizeof(int32_t)); 7253 effect->setInBuffer((int16_t *)buffer); 7254 // auxiliary effects output samples to chain input buffer for further processing 7255 // by insert effects 7256 effect->setOutBuffer(mInBuffer); 7257 } else { 7258 // Insert effects are inserted at the end of mEffects vector as they are processed 7259 // after track and auxiliary effects. 7260 // Insert effect order as a function of indicated preference: 7261 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7262 // another effect is present 7263 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7264 // last effect claiming first position 7265 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7266 // first effect claiming last position 7267 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7268 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7269 // already present 7270 7271 int size = (int)mEffects.size(); 7272 int idx_insert = size; 7273 int idx_insert_first = -1; 7274 int idx_insert_last = -1; 7275 7276 for (int i = 0; i < size; i++) { 7277 effect_descriptor_t d = mEffects[i]->desc(); 7278 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7279 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7280 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7281 // check invalid effect chaining combinations 7282 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7283 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7284 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7285 return INVALID_OPERATION; 7286 } 7287 // remember position of first insert effect and by default 7288 // select this as insert position for new effect 7289 if (idx_insert == size) { 7290 idx_insert = i; 7291 } 7292 // remember position of last insert effect claiming 7293 // first position 7294 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7295 idx_insert_first = i; 7296 } 7297 // remember position of first insert effect claiming 7298 // last position 7299 if (iPref == EFFECT_FLAG_INSERT_LAST && 7300 idx_insert_last == -1) { 7301 idx_insert_last = i; 7302 } 7303 } 7304 } 7305 7306 // modify idx_insert from first position if needed 7307 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7308 if (idx_insert_last != -1) { 7309 idx_insert = idx_insert_last; 7310 } else { 7311 idx_insert = size; 7312 } 7313 } else { 7314 if (idx_insert_first != -1) { 7315 idx_insert = idx_insert_first + 1; 7316 } 7317 } 7318 7319 // always read samples from chain input buffer 7320 effect->setInBuffer(mInBuffer); 7321 7322 // if last effect in the chain, output samples to chain 7323 // output buffer, otherwise to chain input buffer 7324 if (idx_insert == size) { 7325 if (idx_insert != 0) { 7326 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7327 mEffects[idx_insert-1]->configure(); 7328 } 7329 effect->setOutBuffer(mOutBuffer); 7330 } else { 7331 effect->setOutBuffer(mInBuffer); 7332 } 7333 mEffects.insertAt(effect, idx_insert); 7334 7335 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7336 } 7337 effect->configure(); 7338 return NO_ERROR; 7339} 7340 7341// removeEffect_l() must be called with PlaybackThread::mLock held 7342size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7343{ 7344 Mutex::Autolock _l(mLock); 7345 int size = (int)mEffects.size(); 7346 int i; 7347 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7348 7349 for (i = 0; i < size; i++) { 7350 if (effect == mEffects[i]) { 7351 // calling stop here will remove pre-processing effect from the audio HAL. 7352 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7353 // the middle of a read from audio HAL 7354 if (mEffects[i]->state() == EffectModule::ACTIVE || 7355 mEffects[i]->state() == EffectModule::STOPPING) { 7356 mEffects[i]->stop(); 7357 } 7358 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7359 delete[] effect->inBuffer(); 7360 } else { 7361 if (i == size - 1 && i != 0) { 7362 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7363 mEffects[i - 1]->configure(); 7364 } 7365 } 7366 mEffects.removeAt(i); 7367 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7368 break; 7369 } 7370 } 7371 7372 return mEffects.size(); 7373} 7374 7375// setDevice_l() must be called with PlaybackThread::mLock held 7376void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7377{ 7378 size_t size = mEffects.size(); 7379 for (size_t i = 0; i < size; i++) { 7380 mEffects[i]->setDevice(device); 7381 } 7382} 7383 7384// setMode_l() must be called with PlaybackThread::mLock held 7385void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7386{ 7387 size_t size = mEffects.size(); 7388 for (size_t i = 0; i < size; i++) { 7389 mEffects[i]->setMode(mode); 7390 } 7391} 7392 7393// setVolume_l() must be called with PlaybackThread::mLock held 7394bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7395{ 7396 uint32_t newLeft = *left; 7397 uint32_t newRight = *right; 7398 bool hasControl = false; 7399 int ctrlIdx = -1; 7400 size_t size = mEffects.size(); 7401 7402 // first update volume controller 7403 for (size_t i = size; i > 0; i--) { 7404 if (mEffects[i - 1]->isProcessEnabled() && 7405 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7406 ctrlIdx = i - 1; 7407 hasControl = true; 7408 break; 7409 } 7410 } 7411 7412 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7413 if (hasControl) { 7414 *left = mNewLeftVolume; 7415 *right = mNewRightVolume; 7416 } 7417 return hasControl; 7418 } 7419 7420 mVolumeCtrlIdx = ctrlIdx; 7421 mLeftVolume = newLeft; 7422 mRightVolume = newRight; 7423 7424 // second get volume update from volume controller 7425 if (ctrlIdx >= 0) { 7426 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7427 mNewLeftVolume = newLeft; 7428 mNewRightVolume = newRight; 7429 } 7430 // then indicate volume to all other effects in chain. 7431 // Pass altered volume to effects before volume controller 7432 // and requested volume to effects after controller 7433 uint32_t lVol = newLeft; 7434 uint32_t rVol = newRight; 7435 7436 for (size_t i = 0; i < size; i++) { 7437 if ((int)i == ctrlIdx) continue; 7438 // this also works for ctrlIdx == -1 when there is no volume controller 7439 if ((int)i > ctrlIdx) { 7440 lVol = *left; 7441 rVol = *right; 7442 } 7443 mEffects[i]->setVolume(&lVol, &rVol, false); 7444 } 7445 *left = newLeft; 7446 *right = newRight; 7447 7448 return hasControl; 7449} 7450 7451status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7452{ 7453 const size_t SIZE = 256; 7454 char buffer[SIZE]; 7455 String8 result; 7456 7457 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7458 result.append(buffer); 7459 7460 bool locked = tryLock(mLock); 7461 // failed to lock - AudioFlinger is probably deadlocked 7462 if (!locked) { 7463 result.append("\tCould not lock mutex:\n"); 7464 } 7465 7466 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7467 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7468 mEffects.size(), 7469 (uint32_t)mInBuffer, 7470 (uint32_t)mOutBuffer, 7471 mActiveTrackCnt); 7472 result.append(buffer); 7473 write(fd, result.string(), result.size()); 7474 7475 for (size_t i = 0; i < mEffects.size(); ++i) { 7476 sp<EffectModule> effect = mEffects[i]; 7477 if (effect != 0) { 7478 effect->dump(fd, args); 7479 } 7480 } 7481 7482 if (locked) { 7483 mLock.unlock(); 7484 } 7485 7486 return NO_ERROR; 7487} 7488 7489// must be called with ThreadBase::mLock held 7490void AudioFlinger::EffectChain::setEffectSuspended_l( 7491 const effect_uuid_t *type, bool suspend) 7492{ 7493 sp<SuspendedEffectDesc> desc; 7494 // use effect type UUID timelow as key as there is no real risk of identical 7495 // timeLow fields among effect type UUIDs. 7496 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7497 if (suspend) { 7498 if (index >= 0) { 7499 desc = mSuspendedEffects.valueAt(index); 7500 } else { 7501 desc = new SuspendedEffectDesc(); 7502 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7503 mSuspendedEffects.add(type->timeLow, desc); 7504 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7505 } 7506 if (desc->mRefCount++ == 0) { 7507 sp<EffectModule> effect = getEffectIfEnabled(type); 7508 if (effect != 0) { 7509 desc->mEffect = effect; 7510 effect->setSuspended(true); 7511 effect->setEnabled(false); 7512 } 7513 } 7514 } else { 7515 if (index < 0) { 7516 return; 7517 } 7518 desc = mSuspendedEffects.valueAt(index); 7519 if (desc->mRefCount <= 0) { 7520 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7521 desc->mRefCount = 1; 7522 } 7523 if (--desc->mRefCount == 0) { 7524 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7525 if (desc->mEffect != 0) { 7526 sp<EffectModule> effect = desc->mEffect.promote(); 7527 if (effect != 0) { 7528 effect->setSuspended(false); 7529 sp<EffectHandle> handle = effect->controlHandle(); 7530 if (handle != 0) { 7531 effect->setEnabled(handle->enabled()); 7532 } 7533 } 7534 desc->mEffect.clear(); 7535 } 7536 mSuspendedEffects.removeItemsAt(index); 7537 } 7538 } 7539} 7540 7541// must be called with ThreadBase::mLock held 7542void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7543{ 7544 sp<SuspendedEffectDesc> desc; 7545 7546 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7547 if (suspend) { 7548 if (index >= 0) { 7549 desc = mSuspendedEffects.valueAt(index); 7550 } else { 7551 desc = new SuspendedEffectDesc(); 7552 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7553 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7554 } 7555 if (desc->mRefCount++ == 0) { 7556 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7557 for (size_t i = 0; i < effects.size(); i++) { 7558 setEffectSuspended_l(&effects[i]->desc().type, true); 7559 } 7560 } 7561 } else { 7562 if (index < 0) { 7563 return; 7564 } 7565 desc = mSuspendedEffects.valueAt(index); 7566 if (desc->mRefCount <= 0) { 7567 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7568 desc->mRefCount = 1; 7569 } 7570 if (--desc->mRefCount == 0) { 7571 Vector<const effect_uuid_t *> types; 7572 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7573 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7574 continue; 7575 } 7576 types.add(&mSuspendedEffects.valueAt(i)->mType); 7577 } 7578 for (size_t i = 0; i < types.size(); i++) { 7579 setEffectSuspended_l(types[i], false); 7580 } 7581 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7582 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7583 } 7584 } 7585} 7586 7587 7588// The volume effect is used for automated tests only 7589#ifndef OPENSL_ES_H_ 7590static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7591 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7592const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7593#endif //OPENSL_ES_H_ 7594 7595bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7596{ 7597 // auxiliary effects and visualizer are never suspended on output mix 7598 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7599 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7600 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7601 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7602 return false; 7603 } 7604 return true; 7605} 7606 7607Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7608{ 7609 Vector< sp<EffectModule> > effects; 7610 for (size_t i = 0; i < mEffects.size(); i++) { 7611 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7612 continue; 7613 } 7614 effects.add(mEffects[i]); 7615 } 7616 return effects; 7617} 7618 7619sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7620 const effect_uuid_t *type) 7621{ 7622 sp<EffectModule> effect; 7623 effect = getEffectFromType_l(type); 7624 if (effect != 0 && !effect->isEnabled()) { 7625 effect.clear(); 7626 } 7627 return effect; 7628} 7629 7630void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7631 bool enabled) 7632{ 7633 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7634 if (enabled) { 7635 if (index < 0) { 7636 // if the effect is not suspend check if all effects are suspended 7637 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7638 if (index < 0) { 7639 return; 7640 } 7641 if (!isEffectEligibleForSuspend(effect->desc())) { 7642 return; 7643 } 7644 setEffectSuspended_l(&effect->desc().type, enabled); 7645 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7646 if (index < 0) { 7647 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7648 return; 7649 } 7650 } 7651 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7652 effect->desc().type.timeLow); 7653 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7654 // if effect is requested to suspended but was not yet enabled, supend it now. 7655 if (desc->mEffect == 0) { 7656 desc->mEffect = effect; 7657 effect->setEnabled(false); 7658 effect->setSuspended(true); 7659 } 7660 } else { 7661 if (index < 0) { 7662 return; 7663 } 7664 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7665 effect->desc().type.timeLow); 7666 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7667 desc->mEffect.clear(); 7668 effect->setSuspended(false); 7669 } 7670} 7671 7672#undef LOG_TAG 7673#define LOG_TAG "AudioFlinger" 7674 7675// ---------------------------------------------------------------------------- 7676 7677status_t AudioFlinger::onTransact( 7678 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7679{ 7680 return BnAudioFlinger::onTransact(code, data, reply, flags); 7681} 7682 7683}; // namespace android 7684