AudioFlinger.cpp revision 090f01963e215f895020a31e22368cd44e086ce3
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 : Thread(false), 990 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 991 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false), 992 mDevice(device) 993{ 994 mDeathRecipient = new PMDeathRecipient(this); 995} 996 997AudioFlinger::ThreadBase::~ThreadBase() 998{ 999 mParamCond.broadcast(); 1000 // do not lock the mutex in destructor 1001 releaseWakeLock_l(); 1002 if (mPowerManager != 0) { 1003 sp<IBinder> binder = mPowerManager->asBinder(); 1004 binder->unlinkToDeath(mDeathRecipient); 1005 } 1006} 1007 1008void AudioFlinger::ThreadBase::exit() 1009{ 1010 // keep a strong ref on ourself so that we won't get 1011 // destroyed in the middle of requestExitAndWait() 1012 sp <ThreadBase> strongMe = this; 1013 1014 ALOGV("ThreadBase::exit"); 1015 { 1016 AutoMutex lock(mLock); 1017 mExiting = true; 1018 requestExit(); 1019 mWaitWorkCV.signal(); 1020 } 1021 requestExitAndWait(); 1022} 1023 1024uint32_t AudioFlinger::ThreadBase::sampleRate() const 1025{ 1026 return mSampleRate; 1027} 1028 1029int AudioFlinger::ThreadBase::channelCount() const 1030{ 1031 return (int)mChannelCount; 1032} 1033 1034audio_format_t AudioFlinger::ThreadBase::format() const 1035{ 1036 return mFormat; 1037} 1038 1039size_t AudioFlinger::ThreadBase::frameCount() const 1040{ 1041 return mFrameCount; 1042} 1043 1044status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1045{ 1046 status_t status; 1047 1048 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1049 Mutex::Autolock _l(mLock); 1050 1051 mNewParameters.add(keyValuePairs); 1052 mWaitWorkCV.signal(); 1053 // wait condition with timeout in case the thread loop has exited 1054 // before the request could be processed 1055 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1056 status = mParamStatus; 1057 mWaitWorkCV.signal(); 1058 } else { 1059 status = TIMED_OUT; 1060 } 1061 return status; 1062} 1063 1064void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1065{ 1066 Mutex::Autolock _l(mLock); 1067 sendConfigEvent_l(event, param); 1068} 1069 1070// sendConfigEvent_l() must be called with ThreadBase::mLock held 1071void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1072{ 1073 ConfigEvent configEvent; 1074 configEvent.mEvent = event; 1075 configEvent.mParam = param; 1076 mConfigEvents.add(configEvent); 1077 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1078 mWaitWorkCV.signal(); 1079} 1080 1081void AudioFlinger::ThreadBase::processConfigEvents() 1082{ 1083 mLock.lock(); 1084 while(!mConfigEvents.isEmpty()) { 1085 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1086 ConfigEvent configEvent = mConfigEvents[0]; 1087 mConfigEvents.removeAt(0); 1088 // release mLock before locking AudioFlinger mLock: lock order is always 1089 // AudioFlinger then ThreadBase to avoid cross deadlock 1090 mLock.unlock(); 1091 mAudioFlinger->mLock.lock(); 1092 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1093 mAudioFlinger->mLock.unlock(); 1094 mLock.lock(); 1095 } 1096 mLock.unlock(); 1097} 1098 1099status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1100{ 1101 const size_t SIZE = 256; 1102 char buffer[SIZE]; 1103 String8 result; 1104 1105 bool locked = tryLock(mLock); 1106 if (!locked) { 1107 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1108 write(fd, buffer, strlen(buffer)); 1109 } 1110 1111 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1112 result.append(buffer); 1113 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1114 result.append(buffer); 1115 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1116 result.append(buffer); 1117 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1122 result.append(buffer); 1123 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1124 result.append(buffer); 1125 1126 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1127 result.append(buffer); 1128 result.append(" Index Command"); 1129 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1130 snprintf(buffer, SIZE, "\n %02d ", i); 1131 result.append(buffer); 1132 result.append(mNewParameters[i]); 1133 } 1134 1135 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1136 result.append(buffer); 1137 snprintf(buffer, SIZE, " Index event param\n"); 1138 result.append(buffer); 1139 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1140 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1141 result.append(buffer); 1142 } 1143 result.append("\n"); 1144 1145 write(fd, result.string(), result.size()); 1146 1147 if (locked) { 1148 mLock.unlock(); 1149 } 1150 return NO_ERROR; 1151} 1152 1153status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1154{ 1155 const size_t SIZE = 256; 1156 char buffer[SIZE]; 1157 String8 result; 1158 1159 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1160 write(fd, buffer, strlen(buffer)); 1161 1162 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1163 sp<EffectChain> chain = mEffectChains[i]; 1164 if (chain != 0) { 1165 chain->dump(fd, args); 1166 } 1167 } 1168 return NO_ERROR; 1169} 1170 1171void AudioFlinger::ThreadBase::acquireWakeLock() 1172{ 1173 Mutex::Autolock _l(mLock); 1174 acquireWakeLock_l(); 1175} 1176 1177void AudioFlinger::ThreadBase::acquireWakeLock_l() 1178{ 1179 if (mPowerManager == 0) { 1180 // use checkService() to avoid blocking if power service is not up yet 1181 sp<IBinder> binder = 1182 defaultServiceManager()->checkService(String16("power")); 1183 if (binder == 0) { 1184 ALOGW("Thread %s cannot connect to the power manager service", mName); 1185 } else { 1186 mPowerManager = interface_cast<IPowerManager>(binder); 1187 binder->linkToDeath(mDeathRecipient); 1188 } 1189 } 1190 if (mPowerManager != 0) { 1191 sp<IBinder> binder = new BBinder(); 1192 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1193 binder, 1194 String16(mName)); 1195 if (status == NO_ERROR) { 1196 mWakeLockToken = binder; 1197 } 1198 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1199 } 1200} 1201 1202void AudioFlinger::ThreadBase::releaseWakeLock() 1203{ 1204 Mutex::Autolock _l(mLock); 1205 releaseWakeLock_l(); 1206} 1207 1208void AudioFlinger::ThreadBase::releaseWakeLock_l() 1209{ 1210 if (mWakeLockToken != 0) { 1211 ALOGV("releaseWakeLock_l() %s", mName); 1212 if (mPowerManager != 0) { 1213 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1214 } 1215 mWakeLockToken.clear(); 1216 } 1217} 1218 1219void AudioFlinger::ThreadBase::clearPowerManager() 1220{ 1221 Mutex::Autolock _l(mLock); 1222 releaseWakeLock_l(); 1223 mPowerManager.clear(); 1224} 1225 1226void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1227{ 1228 sp<ThreadBase> thread = mThread.promote(); 1229 if (thread != 0) { 1230 thread->clearPowerManager(); 1231 } 1232 ALOGW("power manager service died !!!"); 1233} 1234 1235void AudioFlinger::ThreadBase::setEffectSuspended( 1236 const effect_uuid_t *type, bool suspend, int sessionId) 1237{ 1238 Mutex::Autolock _l(mLock); 1239 setEffectSuspended_l(type, suspend, sessionId); 1240} 1241 1242void AudioFlinger::ThreadBase::setEffectSuspended_l( 1243 const effect_uuid_t *type, bool suspend, int sessionId) 1244{ 1245 sp<EffectChain> chain = getEffectChain_l(sessionId); 1246 if (chain != 0) { 1247 if (type != NULL) { 1248 chain->setEffectSuspended_l(type, suspend); 1249 } else { 1250 chain->setEffectSuspendedAll_l(suspend); 1251 } 1252 } 1253 1254 updateSuspendedSessions_l(type, suspend, sessionId); 1255} 1256 1257void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1258{ 1259 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1260 if (index < 0) { 1261 return; 1262 } 1263 1264 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1265 mSuspendedSessions.editValueAt(index); 1266 1267 for (size_t i = 0; i < sessionEffects.size(); i++) { 1268 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1269 for (int j = 0; j < desc->mRefCount; j++) { 1270 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1271 chain->setEffectSuspendedAll_l(true); 1272 } else { 1273 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1274 desc->mType.timeLow); 1275 chain->setEffectSuspended_l(&desc->mType, true); 1276 } 1277 } 1278 } 1279} 1280 1281void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1282 bool suspend, 1283 int sessionId) 1284{ 1285 int index = mSuspendedSessions.indexOfKey(sessionId); 1286 1287 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1288 1289 if (suspend) { 1290 if (index >= 0) { 1291 sessionEffects = mSuspendedSessions.editValueAt(index); 1292 } else { 1293 mSuspendedSessions.add(sessionId, sessionEffects); 1294 } 1295 } else { 1296 if (index < 0) { 1297 return; 1298 } 1299 sessionEffects = mSuspendedSessions.editValueAt(index); 1300 } 1301 1302 1303 int key = EffectChain::kKeyForSuspendAll; 1304 if (type != NULL) { 1305 key = type->timeLow; 1306 } 1307 index = sessionEffects.indexOfKey(key); 1308 1309 sp <SuspendedSessionDesc> desc; 1310 if (suspend) { 1311 if (index >= 0) { 1312 desc = sessionEffects.valueAt(index); 1313 } else { 1314 desc = new SuspendedSessionDesc(); 1315 if (type != NULL) { 1316 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1317 } 1318 sessionEffects.add(key, desc); 1319 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1320 } 1321 desc->mRefCount++; 1322 } else { 1323 if (index < 0) { 1324 return; 1325 } 1326 desc = sessionEffects.valueAt(index); 1327 if (--desc->mRefCount == 0) { 1328 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1329 sessionEffects.removeItemsAt(index); 1330 if (sessionEffects.isEmpty()) { 1331 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1332 sessionId); 1333 mSuspendedSessions.removeItem(sessionId); 1334 } 1335 } 1336 } 1337 if (!sessionEffects.isEmpty()) { 1338 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1339 } 1340} 1341 1342void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1343 bool enabled, 1344 int sessionId) 1345{ 1346 Mutex::Autolock _l(mLock); 1347 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1348} 1349 1350void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1351 bool enabled, 1352 int sessionId) 1353{ 1354 if (mType != RECORD) { 1355 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1356 // another session. This gives the priority to well behaved effect control panels 1357 // and applications not using global effects. 1358 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1359 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1360 } 1361 } 1362 1363 sp<EffectChain> chain = getEffectChain_l(sessionId); 1364 if (chain != 0) { 1365 chain->checkSuspendOnEffectEnabled(effect, enabled); 1366 } 1367} 1368 1369// ---------------------------------------------------------------------------- 1370 1371AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1372 AudioStreamOut* output, 1373 int id, 1374 uint32_t device) 1375 : ThreadBase(audioFlinger, id, device), 1376 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1377 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1378{ 1379 snprintf(mName, kNameLength, "AudioOut_%d", id); 1380 1381 readOutputParameters(); 1382 1383 // Assumes constructor is called by AudioFlinger with it's mLock held, 1384 // but it would be safer to explicitly pass these as parameters 1385 mMasterVolume = mAudioFlinger->masterVolume_l(); 1386 mMasterMute = mAudioFlinger->masterMute_l(); 1387 1388 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1389 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1390 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1391 stream = (audio_stream_type_t) (stream + 1)) { 1392 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1393 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1394 // initialized by stream_type_t default constructor 1395 // mStreamTypes[stream].valid = true; 1396 } 1397} 1398 1399AudioFlinger::PlaybackThread::~PlaybackThread() 1400{ 1401 delete [] mMixBuffer; 1402} 1403 1404status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1405{ 1406 dumpInternals(fd, args); 1407 dumpTracks(fd, args); 1408 dumpEffectChains(fd, args); 1409 return NO_ERROR; 1410} 1411 1412status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1413{ 1414 const size_t SIZE = 256; 1415 char buffer[SIZE]; 1416 String8 result; 1417 1418 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1419 result.append(buffer); 1420 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1421 for (size_t i = 0; i < mTracks.size(); ++i) { 1422 sp<Track> track = mTracks[i]; 1423 if (track != 0) { 1424 track->dump(buffer, SIZE); 1425 result.append(buffer); 1426 } 1427 } 1428 1429 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1430 result.append(buffer); 1431 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1432 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1433 wp<Track> wTrack = mActiveTracks[i]; 1434 if (wTrack != 0) { 1435 sp<Track> track = wTrack.promote(); 1436 if (track != 0) { 1437 track->dump(buffer, SIZE); 1438 result.append(buffer); 1439 } 1440 } 1441 } 1442 write(fd, result.string(), result.size()); 1443 return NO_ERROR; 1444} 1445 1446status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1447{ 1448 const size_t SIZE = 256; 1449 char buffer[SIZE]; 1450 String8 result; 1451 1452 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1453 result.append(buffer); 1454 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1455 result.append(buffer); 1456 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1457 result.append(buffer); 1458 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1459 result.append(buffer); 1460 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1461 result.append(buffer); 1462 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1463 result.append(buffer); 1464 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1465 result.append(buffer); 1466 write(fd, result.string(), result.size()); 1467 1468 dumpBase(fd, args); 1469 1470 return NO_ERROR; 1471} 1472 1473// Thread virtuals 1474status_t AudioFlinger::PlaybackThread::readyToRun() 1475{ 1476 status_t status = initCheck(); 1477 if (status == NO_ERROR) { 1478 ALOGI("AudioFlinger's thread %p ready to run", this); 1479 } else { 1480 ALOGE("No working audio driver found."); 1481 } 1482 return status; 1483} 1484 1485void AudioFlinger::PlaybackThread::onFirstRef() 1486{ 1487 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1488} 1489 1490// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1491sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1492 const sp<AudioFlinger::Client>& client, 1493 audio_stream_type_t streamType, 1494 uint32_t sampleRate, 1495 audio_format_t format, 1496 uint32_t channelMask, 1497 int frameCount, 1498 const sp<IMemory>& sharedBuffer, 1499 int sessionId, 1500 status_t *status) 1501{ 1502 sp<Track> track; 1503 status_t lStatus; 1504 1505 if (mType == DIRECT) { 1506 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1507 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1508 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1509 "for output %p with format %d", 1510 sampleRate, format, channelMask, mOutput, mFormat); 1511 lStatus = BAD_VALUE; 1512 goto Exit; 1513 } 1514 } 1515 } else { 1516 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1517 if (sampleRate > mSampleRate*2) { 1518 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1519 lStatus = BAD_VALUE; 1520 goto Exit; 1521 } 1522 } 1523 1524 lStatus = initCheck(); 1525 if (lStatus != NO_ERROR) { 1526 ALOGE("Audio driver not initialized."); 1527 goto Exit; 1528 } 1529 1530 { // scope for mLock 1531 Mutex::Autolock _l(mLock); 1532 1533 // all tracks in same audio session must share the same routing strategy otherwise 1534 // conflicts will happen when tracks are moved from one output to another by audio policy 1535 // manager 1536 uint32_t strategy = 1537 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1538 for (size_t i = 0; i < mTracks.size(); ++i) { 1539 sp<Track> t = mTracks[i]; 1540 if (t != 0) { 1541 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1542 if (sessionId == t->sessionId() && strategy != actual) { 1543 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1544 strategy, actual); 1545 lStatus = BAD_VALUE; 1546 goto Exit; 1547 } 1548 } 1549 } 1550 1551 track = new Track(this, client, streamType, sampleRate, format, 1552 channelMask, frameCount, sharedBuffer, sessionId); 1553 if (track->getCblk() == NULL || track->name() < 0) { 1554 lStatus = NO_MEMORY; 1555 goto Exit; 1556 } 1557 mTracks.add(track); 1558 1559 sp<EffectChain> chain = getEffectChain_l(sessionId); 1560 if (chain != 0) { 1561 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1562 track->setMainBuffer(chain->inBuffer()); 1563 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1564 chain->incTrackCnt(); 1565 } 1566 1567 // invalidate track immediately if the stream type was moved to another thread since 1568 // createTrack() was called by the client process. 1569 if (!mStreamTypes[streamType].valid) { 1570 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1571 this, streamType); 1572 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1573 } 1574 } 1575 lStatus = NO_ERROR; 1576 1577Exit: 1578 if(status) { 1579 *status = lStatus; 1580 } 1581 return track; 1582} 1583 1584uint32_t AudioFlinger::PlaybackThread::latency() const 1585{ 1586 Mutex::Autolock _l(mLock); 1587 if (initCheck() == NO_ERROR) { 1588 return mOutput->stream->get_latency(mOutput->stream); 1589 } else { 1590 return 0; 1591 } 1592} 1593 1594status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1595{ 1596 mMasterVolume = value; 1597 return NO_ERROR; 1598} 1599 1600status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1601{ 1602 mMasterMute = muted; 1603 return NO_ERROR; 1604} 1605 1606float AudioFlinger::PlaybackThread::masterVolume() const 1607{ 1608 return mMasterVolume; 1609} 1610 1611bool AudioFlinger::PlaybackThread::masterMute() const 1612{ 1613 return mMasterMute; 1614} 1615 1616status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1617{ 1618 mStreamTypes[stream].volume = value; 1619 return NO_ERROR; 1620} 1621 1622status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1623{ 1624 mStreamTypes[stream].mute = muted; 1625 return NO_ERROR; 1626} 1627 1628float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1629{ 1630 return mStreamTypes[stream].volume; 1631} 1632 1633bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1634{ 1635 return mStreamTypes[stream].mute; 1636} 1637 1638// addTrack_l() must be called with ThreadBase::mLock held 1639status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1640{ 1641 status_t status = ALREADY_EXISTS; 1642 1643 // set retry count for buffer fill 1644 track->mRetryCount = kMaxTrackStartupRetries; 1645 if (mActiveTracks.indexOf(track) < 0) { 1646 // the track is newly added, make sure it fills up all its 1647 // buffers before playing. This is to ensure the client will 1648 // effectively get the latency it requested. 1649 track->mFillingUpStatus = Track::FS_FILLING; 1650 track->mResetDone = false; 1651 mActiveTracks.add(track); 1652 if (track->mainBuffer() != mMixBuffer) { 1653 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1654 if (chain != 0) { 1655 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1656 chain->incActiveTrackCnt(); 1657 } 1658 } 1659 1660 status = NO_ERROR; 1661 } 1662 1663 ALOGV("mWaitWorkCV.broadcast"); 1664 mWaitWorkCV.broadcast(); 1665 1666 return status; 1667} 1668 1669// destroyTrack_l() must be called with ThreadBase::mLock held 1670void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1671{ 1672 track->mState = TrackBase::TERMINATED; 1673 if (mActiveTracks.indexOf(track) < 0) { 1674 removeTrack_l(track); 1675 } 1676} 1677 1678void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1679{ 1680 mTracks.remove(track); 1681 deleteTrackName_l(track->name()); 1682 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1683 if (chain != 0) { 1684 chain->decTrackCnt(); 1685 } 1686} 1687 1688String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1689{ 1690 String8 out_s8 = String8(""); 1691 char *s; 1692 1693 Mutex::Autolock _l(mLock); 1694 if (initCheck() != NO_ERROR) { 1695 return out_s8; 1696 } 1697 1698 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1699 out_s8 = String8(s); 1700 free(s); 1701 return out_s8; 1702} 1703 1704// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1705void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1706 AudioSystem::OutputDescriptor desc; 1707 void *param2 = 0; 1708 1709 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1710 1711 switch (event) { 1712 case AudioSystem::OUTPUT_OPENED: 1713 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1714 desc.channels = mChannelMask; 1715 desc.samplingRate = mSampleRate; 1716 desc.format = mFormat; 1717 desc.frameCount = mFrameCount; 1718 desc.latency = latency(); 1719 param2 = &desc; 1720 break; 1721 1722 case AudioSystem::STREAM_CONFIG_CHANGED: 1723 param2 = ¶m; 1724 case AudioSystem::OUTPUT_CLOSED: 1725 default: 1726 break; 1727 } 1728 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1729} 1730 1731void AudioFlinger::PlaybackThread::readOutputParameters() 1732{ 1733 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1734 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1735 mChannelCount = (uint16_t)popcount(mChannelMask); 1736 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1737 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1738 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1739 1740 // FIXME - Current mixer implementation only supports stereo output: Always 1741 // Allocate a stereo buffer even if HW output is mono. 1742 if (mMixBuffer != NULL) delete[] mMixBuffer; 1743 mMixBuffer = new int16_t[mFrameCount * 2]; 1744 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1745 1746 // force reconfiguration of effect chains and engines to take new buffer size and audio 1747 // parameters into account 1748 // Note that mLock is not held when readOutputParameters() is called from the constructor 1749 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1750 // matter. 1751 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1752 Vector< sp<EffectChain> > effectChains = mEffectChains; 1753 for (size_t i = 0; i < effectChains.size(); i ++) { 1754 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1755 } 1756} 1757 1758status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1759{ 1760 if (halFrames == 0 || dspFrames == 0) { 1761 return BAD_VALUE; 1762 } 1763 Mutex::Autolock _l(mLock); 1764 if (initCheck() != NO_ERROR) { 1765 return INVALID_OPERATION; 1766 } 1767 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1768 1769 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1770} 1771 1772uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1773{ 1774 Mutex::Autolock _l(mLock); 1775 uint32_t result = 0; 1776 if (getEffectChain_l(sessionId) != 0) { 1777 result = EFFECT_SESSION; 1778 } 1779 1780 for (size_t i = 0; i < mTracks.size(); ++i) { 1781 sp<Track> track = mTracks[i]; 1782 if (sessionId == track->sessionId() && 1783 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1784 result |= TRACK_SESSION; 1785 break; 1786 } 1787 } 1788 1789 return result; 1790} 1791 1792uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1793{ 1794 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1795 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1796 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1797 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1798 } 1799 for (size_t i = 0; i < mTracks.size(); i++) { 1800 sp<Track> track = mTracks[i]; 1801 if (sessionId == track->sessionId() && 1802 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1803 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1804 } 1805 } 1806 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1807} 1808 1809 1810AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1811{ 1812 Mutex::Autolock _l(mLock); 1813 return mOutput; 1814} 1815 1816AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1817{ 1818 Mutex::Autolock _l(mLock); 1819 AudioStreamOut *output = mOutput; 1820 mOutput = NULL; 1821 return output; 1822} 1823 1824// this method must always be called either with ThreadBase mLock held or inside the thread loop 1825audio_stream_t* AudioFlinger::PlaybackThread::stream() 1826{ 1827 if (mOutput == NULL) { 1828 return NULL; 1829 } 1830 return &mOutput->stream->common; 1831} 1832 1833uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1834{ 1835 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1836 // decoding and transfer time. So sleeping for half of the latency would likely cause 1837 // underruns 1838 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1839 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1840 } else { 1841 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1842 } 1843} 1844 1845// ---------------------------------------------------------------------------- 1846 1847AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1848 : PlaybackThread(audioFlinger, output, id, device), 1849 mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE) 1850{ 1851 mType = ThreadBase::MIXER; 1852 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1853 1854 // FIXME - Current mixer implementation only supports stereo output 1855 if (mChannelCount == 1) { 1856 ALOGE("Invalid audio hardware channel count"); 1857 } 1858} 1859 1860AudioFlinger::MixerThread::~MixerThread() 1861{ 1862 delete mAudioMixer; 1863} 1864 1865bool AudioFlinger::MixerThread::threadLoop() 1866{ 1867 Vector< sp<Track> > tracksToRemove; 1868 mixer_state mixerStatus = MIXER_IDLE; 1869 nsecs_t standbyTime = systemTime(); 1870 size_t mixBufferSize = mFrameCount * mFrameSize; 1871 // FIXME: Relaxed timing because of a certain device that can't meet latency 1872 // Should be reduced to 2x after the vendor fixes the driver issue 1873 // increase threshold again due to low power audio mode. The way this warning threshold is 1874 // calculated and its usefulness should be reconsidered anyway. 1875 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1876 nsecs_t lastWarning = 0; 1877 bool longStandbyExit = false; 1878 uint32_t activeSleepTime = activeSleepTimeUs(); 1879 uint32_t idleSleepTime = idleSleepTimeUs(); 1880 uint32_t sleepTime = idleSleepTime; 1881 uint32_t sleepTimeShift = 0; 1882 Vector< sp<EffectChain> > effectChains; 1883#ifdef DEBUG_CPU_USAGE 1884 ThreadCpuUsage cpu; 1885 const CentralTendencyStatistics& stats = cpu.statistics(); 1886#endif 1887 1888 acquireWakeLock(); 1889 1890 while (!exitPending()) 1891 { 1892#ifdef DEBUG_CPU_USAGE 1893 cpu.sampleAndEnable(); 1894 unsigned n = stats.n(); 1895 // cpu.elapsed() is expensive, so don't call it every loop 1896 if ((n & 127) == 1) { 1897 long long elapsed = cpu.elapsed(); 1898 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1899 double perLoop = elapsed / (double) n; 1900 double perLoop100 = perLoop * 0.01; 1901 double mean = stats.mean(); 1902 double stddev = stats.stddev(); 1903 double minimum = stats.minimum(); 1904 double maximum = stats.maximum(); 1905 cpu.resetStatistics(); 1906 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", 1907 elapsed * .000000001, n, perLoop * .000001, 1908 mean * .001, 1909 stddev * .001, 1910 minimum * .001, 1911 maximum * .001, 1912 mean / perLoop100, 1913 stddev / perLoop100, 1914 minimum / perLoop100, 1915 maximum / perLoop100); 1916 } 1917 } 1918#endif 1919 processConfigEvents(); 1920 1921 mixerStatus = MIXER_IDLE; 1922 { // scope for mLock 1923 1924 Mutex::Autolock _l(mLock); 1925 1926 if (checkForNewParameters_l()) { 1927 mixBufferSize = mFrameCount * mFrameSize; 1928 // FIXME: Relaxed timing because of a certain device that can't meet latency 1929 // Should be reduced to 2x after the vendor fixes the driver issue 1930 // increase threshold again due to low power audio mode. The way this warning 1931 // threshold is calculated and its usefulness should be reconsidered anyway. 1932 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1933 activeSleepTime = activeSleepTimeUs(); 1934 idleSleepTime = idleSleepTimeUs(); 1935 } 1936 1937 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1938 1939 // put audio hardware into standby after short delay 1940 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1941 mSuspended)) { 1942 if (!mStandby) { 1943 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1944 mOutput->stream->common.standby(&mOutput->stream->common); 1945 mStandby = true; 1946 mBytesWritten = 0; 1947 } 1948 1949 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1950 // we're about to wait, flush the binder command buffer 1951 IPCThreadState::self()->flushCommands(); 1952 1953 if (exitPending()) break; 1954 1955 releaseWakeLock_l(); 1956 // wait until we have something to do... 1957 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1958 mWaitWorkCV.wait(mLock); 1959 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1960 acquireWakeLock_l(); 1961 1962 mPrevMixerStatus = MIXER_IDLE; 1963 if (!mMasterMute) { 1964 char value[PROPERTY_VALUE_MAX]; 1965 property_get("ro.audio.silent", value, "0"); 1966 if (atoi(value)) { 1967 ALOGD("Silence is golden"); 1968 setMasterMute(true); 1969 } 1970 } 1971 1972 standbyTime = systemTime() + kStandbyTimeInNsecs; 1973 sleepTime = idleSleepTime; 1974 sleepTimeShift = 0; 1975 continue; 1976 } 1977 } 1978 1979 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1980 1981 // prevent any changes in effect chain list and in each effect chain 1982 // during mixing and effect process as the audio buffers could be deleted 1983 // or modified if an effect is created or deleted 1984 lockEffectChains_l(effectChains); 1985 } 1986 1987 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1988 // mix buffers... 1989 mAudioMixer->process(); 1990 // increase sleep time progressively when application underrun condition clears. 1991 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1992 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1993 // such that we would underrun the audio HAL. 1994 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 1995 sleepTimeShift--; 1996 } 1997 sleepTime = 0; 1998 standbyTime = systemTime() + kStandbyTimeInNsecs; 1999 //TODO: delay standby when effects have a tail 2000 } else { 2001 // If no tracks are ready, sleep once for the duration of an output 2002 // buffer size, then write 0s to the output 2003 if (sleepTime == 0) { 2004 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2005 sleepTime = activeSleepTime >> sleepTimeShift; 2006 if (sleepTime < kMinThreadSleepTimeUs) { 2007 sleepTime = kMinThreadSleepTimeUs; 2008 } 2009 // reduce sleep time in case of consecutive application underruns to avoid 2010 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2011 // duration we would end up writing less data than needed by the audio HAL if 2012 // the condition persists. 2013 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2014 sleepTimeShift++; 2015 } 2016 } else { 2017 sleepTime = idleSleepTime; 2018 } 2019 } else if (mBytesWritten != 0 || 2020 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2021 memset (mMixBuffer, 0, mixBufferSize); 2022 sleepTime = 0; 2023 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2024 } 2025 // TODO add standby time extension fct of effect tail 2026 } 2027 2028 if (mSuspended) { 2029 sleepTime = suspendSleepTimeUs(); 2030 } 2031 // sleepTime == 0 means we must write to audio hardware 2032 if (sleepTime == 0) { 2033 for (size_t i = 0; i < effectChains.size(); i ++) { 2034 effectChains[i]->process_l(); 2035 } 2036 // enable changes in effect chain 2037 unlockEffectChains(effectChains); 2038 mLastWriteTime = systemTime(); 2039 mInWrite = true; 2040 mBytesWritten += mixBufferSize; 2041 2042 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2043 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2044 mNumWrites++; 2045 mInWrite = false; 2046 nsecs_t now = systemTime(); 2047 nsecs_t delta = now - mLastWriteTime; 2048 if (!mStandby && delta > maxPeriod) { 2049 mNumDelayedWrites++; 2050 if ((now - lastWarning) > kWarningThrottleNs) { 2051 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2052 ns2ms(delta), mNumDelayedWrites, this); 2053 lastWarning = now; 2054 } 2055 if (mStandby) { 2056 longStandbyExit = true; 2057 } 2058 } 2059 mStandby = false; 2060 } else { 2061 // enable changes in effect chain 2062 unlockEffectChains(effectChains); 2063 usleep(sleepTime); 2064 } 2065 2066 // finally let go of all our tracks, without the lock held 2067 // since we can't guarantee the destructors won't acquire that 2068 // same lock. 2069 tracksToRemove.clear(); 2070 2071 // Effect chains will be actually deleted here if they were removed from 2072 // mEffectChains list during mixing or effects processing 2073 effectChains.clear(); 2074 } 2075 2076 if (!mStandby) { 2077 mOutput->stream->common.standby(&mOutput->stream->common); 2078 } 2079 2080 releaseWakeLock(); 2081 2082 ALOGV("MixerThread %p exiting", this); 2083 return false; 2084} 2085 2086// prepareTracks_l() must be called with ThreadBase::mLock held 2087AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2088 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2089{ 2090 2091 mixer_state mixerStatus = MIXER_IDLE; 2092 // find out which tracks need to be processed 2093 size_t count = activeTracks.size(); 2094 size_t mixedTracks = 0; 2095 size_t tracksWithEffect = 0; 2096 2097 float masterVolume = mMasterVolume; 2098 bool masterMute = mMasterMute; 2099 2100 if (masterMute) { 2101 masterVolume = 0; 2102 } 2103 // Delegate master volume control to effect in output mix effect chain if needed 2104 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2105 if (chain != 0) { 2106 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2107 chain->setVolume_l(&v, &v); 2108 masterVolume = (float)((v + (1 << 23)) >> 24); 2109 chain.clear(); 2110 } 2111 2112 for (size_t i=0 ; i<count ; i++) { 2113 sp<Track> t = activeTracks[i].promote(); 2114 if (t == 0) continue; 2115 2116 // this const just means the local variable doesn't change 2117 Track* const track = t.get(); 2118 audio_track_cblk_t* cblk = track->cblk(); 2119 2120 // The first time a track is added we wait 2121 // for all its buffers to be filled before processing it 2122 int name = track->name(); 2123 // make sure that we have enough frames to mix one full buffer. 2124 // enforce this condition only once to enable draining the buffer in case the client 2125 // app does not call stop() and relies on underrun to stop: 2126 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2127 // during last round 2128 uint32_t minFrames = 1; 2129 if (!track->isStopped() && !track->isPausing() && 2130 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2131 if (t->sampleRate() == (int)mSampleRate) { 2132 minFrames = mFrameCount; 2133 } else { 2134 // +1 for rounding and +1 for additional sample needed for interpolation 2135 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2136 // add frames already consumed but not yet released by the resampler 2137 // because cblk->framesReady() will include these frames 2138 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2139 // the minimum track buffer size is normally twice the number of frames necessary 2140 // to fill one buffer and the resampler should not leave more than one buffer worth 2141 // of unreleased frames after each pass, but just in case... 2142 ALOG_ASSERT(minFrames <= cblk->frameCount); 2143 } 2144 } 2145 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2146 !track->isPaused() && !track->isTerminated()) 2147 { 2148 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2149 2150 mixedTracks++; 2151 2152 // track->mainBuffer() != mMixBuffer means there is an effect chain 2153 // connected to the track 2154 chain.clear(); 2155 if (track->mainBuffer() != mMixBuffer) { 2156 chain = getEffectChain_l(track->sessionId()); 2157 // Delegate volume control to effect in track effect chain if needed 2158 if (chain != 0) { 2159 tracksWithEffect++; 2160 } else { 2161 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2162 name, track->sessionId()); 2163 } 2164 } 2165 2166 2167 int param = AudioMixer::VOLUME; 2168 if (track->mFillingUpStatus == Track::FS_FILLED) { 2169 // no ramp for the first volume setting 2170 track->mFillingUpStatus = Track::FS_ACTIVE; 2171 if (track->mState == TrackBase::RESUMING) { 2172 track->mState = TrackBase::ACTIVE; 2173 param = AudioMixer::RAMP_VOLUME; 2174 } 2175 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2176 } else if (cblk->server != 0) { 2177 // If the track is stopped before the first frame was mixed, 2178 // do not apply ramp 2179 param = AudioMixer::RAMP_VOLUME; 2180 } 2181 2182 // compute volume for this track 2183 uint32_t vl, vr, va; 2184 if (track->isMuted() || track->isPausing() || 2185 mStreamTypes[track->type()].mute) { 2186 vl = vr = va = 0; 2187 if (track->isPausing()) { 2188 track->setPaused(); 2189 } 2190 } else { 2191 2192 // read original volumes with volume control 2193 float typeVolume = mStreamTypes[track->type()].volume; 2194 float v = masterVolume * typeVolume; 2195 uint32_t vlr = cblk->volumeLR; 2196 vl = vlr & 0xFFFF; 2197 vr = vlr >> 16; 2198 // track volumes come from shared memory, so can't be trusted and must be clamped 2199 if (vl > MAX_GAIN_INT) { 2200 ALOGV("Track left volume out of range: %04X", vl); 2201 vl = MAX_GAIN_INT; 2202 } 2203 if (vr > MAX_GAIN_INT) { 2204 ALOGV("Track right volume out of range: %04X", vr); 2205 vr = MAX_GAIN_INT; 2206 } 2207 // now apply the master volume and stream type volume 2208 vl = (uint32_t)(v * vl) << 12; 2209 vr = (uint32_t)(v * vr) << 12; 2210 // assuming master volume and stream type volume each go up to 1.0, 2211 // vl and vr are now in 8.24 format 2212 2213 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2214 // send level comes from shared memory and so may be corrupt 2215 if (sendLevel >= MAX_GAIN_INT) { 2216 ALOGV("Track send level out of range: %04X", sendLevel); 2217 sendLevel = MAX_GAIN_INT; 2218 } 2219 va = (uint32_t)(v * sendLevel); 2220 } 2221 // Delegate volume control to effect in track effect chain if needed 2222 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2223 // Do not ramp volume if volume is controlled by effect 2224 param = AudioMixer::VOLUME; 2225 track->mHasVolumeController = true; 2226 } else { 2227 // force no volume ramp when volume controller was just disabled or removed 2228 // from effect chain to avoid volume spike 2229 if (track->mHasVolumeController) { 2230 param = AudioMixer::VOLUME; 2231 } 2232 track->mHasVolumeController = false; 2233 } 2234 2235 // Convert volumes from 8.24 to 4.12 format 2236 int16_t left, right, aux; 2237 // This additional clamping is needed in case chain->setVolume_l() overshot 2238 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2239 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2240 left = int16_t(v_clamped); 2241 v_clamped = (vr + (1 << 11)) >> 12; 2242 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2243 right = int16_t(v_clamped); 2244 2245 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2246 aux = int16_t(va); 2247 2248 // XXX: these things DON'T need to be done each time 2249 mAudioMixer->setBufferProvider(name, track); 2250 mAudioMixer->enable(name); 2251 2252 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2253 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2254 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2255 mAudioMixer->setParameter( 2256 name, 2257 AudioMixer::TRACK, 2258 AudioMixer::FORMAT, (void *)track->format()); 2259 mAudioMixer->setParameter( 2260 name, 2261 AudioMixer::TRACK, 2262 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2263 mAudioMixer->setParameter( 2264 name, 2265 AudioMixer::RESAMPLE, 2266 AudioMixer::SAMPLE_RATE, 2267 (void *)(cblk->sampleRate)); 2268 mAudioMixer->setParameter( 2269 name, 2270 AudioMixer::TRACK, 2271 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2272 mAudioMixer->setParameter( 2273 name, 2274 AudioMixer::TRACK, 2275 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2276 2277 // reset retry count 2278 track->mRetryCount = kMaxTrackRetries; 2279 // If one track is ready, set the mixer ready if: 2280 // - the mixer was not ready during previous round OR 2281 // - no other track is not ready 2282 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2283 mixerStatus != MIXER_TRACKS_ENABLED) { 2284 mixerStatus = MIXER_TRACKS_READY; 2285 } 2286 } else { 2287 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2288 if (track->isStopped()) { 2289 track->reset(); 2290 } 2291 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2292 // We have consumed all the buffers of this track. 2293 // Remove it from the list of active tracks. 2294 tracksToRemove->add(track); 2295 } else { 2296 // No buffers for this track. Give it a few chances to 2297 // fill a buffer, then remove it from active list. 2298 if (--(track->mRetryCount) <= 0) { 2299 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2300 tracksToRemove->add(track); 2301 // indicate to client process that the track was disabled because of underrun 2302 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2303 // If one track is not ready, mark the mixer also not ready if: 2304 // - the mixer was ready during previous round OR 2305 // - no other track is ready 2306 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2307 mixerStatus != MIXER_TRACKS_READY) { 2308 mixerStatus = MIXER_TRACKS_ENABLED; 2309 } 2310 } 2311 mAudioMixer->disable(name); 2312 } 2313 } 2314 2315 // remove all the tracks that need to be... 2316 count = tracksToRemove->size(); 2317 if (CC_UNLIKELY(count)) { 2318 for (size_t i=0 ; i<count ; i++) { 2319 const sp<Track>& track = tracksToRemove->itemAt(i); 2320 mActiveTracks.remove(track); 2321 if (track->mainBuffer() != mMixBuffer) { 2322 chain = getEffectChain_l(track->sessionId()); 2323 if (chain != 0) { 2324 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2325 chain->decActiveTrackCnt(); 2326 } 2327 } 2328 if (track->isTerminated()) { 2329 removeTrack_l(track); 2330 } 2331 } 2332 } 2333 2334 // mix buffer must be cleared if all tracks are connected to an 2335 // effect chain as in this case the mixer will not write to 2336 // mix buffer and track effects will accumulate into it 2337 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2338 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2339 } 2340 2341 mPrevMixerStatus = mixerStatus; 2342 return mixerStatus; 2343} 2344 2345void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2346{ 2347 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2348 this, streamType, mTracks.size()); 2349 Mutex::Autolock _l(mLock); 2350 2351 size_t size = mTracks.size(); 2352 for (size_t i = 0; i < size; i++) { 2353 sp<Track> t = mTracks[i]; 2354 if (t->type() == streamType) { 2355 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2356 t->mCblk->cv.signal(); 2357 } 2358 } 2359} 2360 2361void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2362{ 2363 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2364 this, streamType, valid); 2365 Mutex::Autolock _l(mLock); 2366 2367 mStreamTypes[streamType].valid = valid; 2368} 2369 2370// getTrackName_l() must be called with ThreadBase::mLock held 2371int AudioFlinger::MixerThread::getTrackName_l() 2372{ 2373 return mAudioMixer->getTrackName(); 2374} 2375 2376// deleteTrackName_l() must be called with ThreadBase::mLock held 2377void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2378{ 2379 ALOGV("remove track (%d) and delete from mixer", name); 2380 mAudioMixer->deleteTrackName(name); 2381} 2382 2383// checkForNewParameters_l() must be called with ThreadBase::mLock held 2384bool AudioFlinger::MixerThread::checkForNewParameters_l() 2385{ 2386 bool reconfig = false; 2387 2388 while (!mNewParameters.isEmpty()) { 2389 status_t status = NO_ERROR; 2390 String8 keyValuePair = mNewParameters[0]; 2391 AudioParameter param = AudioParameter(keyValuePair); 2392 int value; 2393 2394 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2395 reconfig = true; 2396 } 2397 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2398 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2399 status = BAD_VALUE; 2400 } else { 2401 reconfig = true; 2402 } 2403 } 2404 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2405 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2406 status = BAD_VALUE; 2407 } else { 2408 reconfig = true; 2409 } 2410 } 2411 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2412 // do not accept frame count changes if tracks are open as the track buffer 2413 // size depends on frame count and correct behavior would not be guaranteed 2414 // if frame count is changed after track creation 2415 if (!mTracks.isEmpty()) { 2416 status = INVALID_OPERATION; 2417 } else { 2418 reconfig = true; 2419 } 2420 } 2421 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2422 // when changing the audio output device, call addBatteryData to notify 2423 // the change 2424 if ((int)mDevice != value) { 2425 uint32_t params = 0; 2426 // check whether speaker is on 2427 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2428 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2429 } 2430 2431 int deviceWithoutSpeaker 2432 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2433 // check if any other device (except speaker) is on 2434 if (value & deviceWithoutSpeaker ) { 2435 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2436 } 2437 2438 if (params != 0) { 2439 addBatteryData(params); 2440 } 2441 } 2442 2443 // forward device change to effects that have requested to be 2444 // aware of attached audio device. 2445 mDevice = (uint32_t)value; 2446 for (size_t i = 0; i < mEffectChains.size(); i++) { 2447 mEffectChains[i]->setDevice_l(mDevice); 2448 } 2449 } 2450 2451 if (status == NO_ERROR) { 2452 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2453 keyValuePair.string()); 2454 if (!mStandby && status == INVALID_OPERATION) { 2455 mOutput->stream->common.standby(&mOutput->stream->common); 2456 mStandby = true; 2457 mBytesWritten = 0; 2458 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2459 keyValuePair.string()); 2460 } 2461 if (status == NO_ERROR && reconfig) { 2462 delete mAudioMixer; 2463 readOutputParameters(); 2464 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2465 for (size_t i = 0; i < mTracks.size() ; i++) { 2466 int name = getTrackName_l(); 2467 if (name < 0) break; 2468 mTracks[i]->mName = name; 2469 // limit track sample rate to 2 x new output sample rate 2470 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2471 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2472 } 2473 } 2474 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2475 } 2476 } 2477 2478 mNewParameters.removeAt(0); 2479 2480 mParamStatus = status; 2481 mParamCond.signal(); 2482 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2483 // already timed out waiting for the status and will never signal the condition. 2484 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2485 } 2486 return reconfig; 2487} 2488 2489status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2490{ 2491 const size_t SIZE = 256; 2492 char buffer[SIZE]; 2493 String8 result; 2494 2495 PlaybackThread::dumpInternals(fd, args); 2496 2497 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2498 result.append(buffer); 2499 write(fd, result.string(), result.size()); 2500 return NO_ERROR; 2501} 2502 2503uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2504{ 2505 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2506} 2507 2508uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2509{ 2510 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2511} 2512 2513// ---------------------------------------------------------------------------- 2514AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2515 : PlaybackThread(audioFlinger, output, id, device) 2516{ 2517 mType = ThreadBase::DIRECT; 2518} 2519 2520AudioFlinger::DirectOutputThread::~DirectOutputThread() 2521{ 2522} 2523 2524static inline 2525int32_t mul(int16_t in, int16_t v) 2526{ 2527#if defined(__arm__) && !defined(__thumb__) 2528 int32_t out; 2529 asm( "smulbb %[out], %[in], %[v] \n" 2530 : [out]"=r"(out) 2531 : [in]"%r"(in), [v]"r"(v) 2532 : ); 2533 return out; 2534#else 2535 return in * int32_t(v); 2536#endif 2537} 2538 2539void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2540{ 2541 // Do not apply volume on compressed audio 2542 if (!audio_is_linear_pcm(mFormat)) { 2543 return; 2544 } 2545 2546 // convert to signed 16 bit before volume calculation 2547 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2548 size_t count = mFrameCount * mChannelCount; 2549 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2550 int16_t *dst = mMixBuffer + count-1; 2551 while(count--) { 2552 *dst-- = (int16_t)(*src--^0x80) << 8; 2553 } 2554 } 2555 2556 size_t frameCount = mFrameCount; 2557 int16_t *out = mMixBuffer; 2558 if (ramp) { 2559 if (mChannelCount == 1) { 2560 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2561 int32_t vlInc = d / (int32_t)frameCount; 2562 int32_t vl = ((int32_t)mLeftVolShort << 16); 2563 do { 2564 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2565 out++; 2566 vl += vlInc; 2567 } while (--frameCount); 2568 2569 } else { 2570 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2571 int32_t vlInc = d / (int32_t)frameCount; 2572 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2573 int32_t vrInc = d / (int32_t)frameCount; 2574 int32_t vl = ((int32_t)mLeftVolShort << 16); 2575 int32_t vr = ((int32_t)mRightVolShort << 16); 2576 do { 2577 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2578 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2579 out += 2; 2580 vl += vlInc; 2581 vr += vrInc; 2582 } while (--frameCount); 2583 } 2584 } else { 2585 if (mChannelCount == 1) { 2586 do { 2587 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2588 out++; 2589 } while (--frameCount); 2590 } else { 2591 do { 2592 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2593 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2594 out += 2; 2595 } while (--frameCount); 2596 } 2597 } 2598 2599 // convert back to unsigned 8 bit after volume calculation 2600 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2601 size_t count = mFrameCount * mChannelCount; 2602 int16_t *src = mMixBuffer; 2603 uint8_t *dst = (uint8_t *)mMixBuffer; 2604 while(count--) { 2605 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2606 } 2607 } 2608 2609 mLeftVolShort = leftVol; 2610 mRightVolShort = rightVol; 2611} 2612 2613bool AudioFlinger::DirectOutputThread::threadLoop() 2614{ 2615 mixer_state mixerStatus = MIXER_IDLE; 2616 sp<Track> trackToRemove; 2617 sp<Track> activeTrack; 2618 nsecs_t standbyTime = systemTime(); 2619 int8_t *curBuf; 2620 size_t mixBufferSize = mFrameCount*mFrameSize; 2621 uint32_t activeSleepTime = activeSleepTimeUs(); 2622 uint32_t idleSleepTime = idleSleepTimeUs(); 2623 uint32_t sleepTime = idleSleepTime; 2624 // use shorter standby delay as on normal output to release 2625 // hardware resources as soon as possible 2626 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2627 2628 acquireWakeLock(); 2629 2630 while (!exitPending()) 2631 { 2632 bool rampVolume; 2633 uint16_t leftVol; 2634 uint16_t rightVol; 2635 Vector< sp<EffectChain> > effectChains; 2636 2637 processConfigEvents(); 2638 2639 mixerStatus = MIXER_IDLE; 2640 2641 { // scope for the mLock 2642 2643 Mutex::Autolock _l(mLock); 2644 2645 if (checkForNewParameters_l()) { 2646 mixBufferSize = mFrameCount*mFrameSize; 2647 activeSleepTime = activeSleepTimeUs(); 2648 idleSleepTime = idleSleepTimeUs(); 2649 standbyDelay = microseconds(activeSleepTime*2); 2650 } 2651 2652 // put audio hardware into standby after short delay 2653 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2654 mSuspended)) { 2655 // wait until we have something to do... 2656 if (!mStandby) { 2657 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2658 mOutput->stream->common.standby(&mOutput->stream->common); 2659 mStandby = true; 2660 mBytesWritten = 0; 2661 } 2662 2663 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2664 // we're about to wait, flush the binder command buffer 2665 IPCThreadState::self()->flushCommands(); 2666 2667 if (exitPending()) break; 2668 2669 releaseWakeLock_l(); 2670 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2671 mWaitWorkCV.wait(mLock); 2672 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2673 acquireWakeLock_l(); 2674 2675 if (!mMasterMute) { 2676 char value[PROPERTY_VALUE_MAX]; 2677 property_get("ro.audio.silent", value, "0"); 2678 if (atoi(value)) { 2679 ALOGD("Silence is golden"); 2680 setMasterMute(true); 2681 } 2682 } 2683 2684 standbyTime = systemTime() + standbyDelay; 2685 sleepTime = idleSleepTime; 2686 continue; 2687 } 2688 } 2689 2690 effectChains = mEffectChains; 2691 2692 // find out which tracks need to be processed 2693 if (mActiveTracks.size() != 0) { 2694 sp<Track> t = mActiveTracks[0].promote(); 2695 if (t == 0) continue; 2696 2697 Track* const track = t.get(); 2698 audio_track_cblk_t* cblk = track->cblk(); 2699 2700 // The first time a track is added we wait 2701 // for all its buffers to be filled before processing it 2702 if (cblk->framesReady() && track->isReady() && 2703 !track->isPaused() && !track->isTerminated()) 2704 { 2705 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2706 2707 if (track->mFillingUpStatus == Track::FS_FILLED) { 2708 track->mFillingUpStatus = Track::FS_ACTIVE; 2709 mLeftVolFloat = mRightVolFloat = 0; 2710 mLeftVolShort = mRightVolShort = 0; 2711 if (track->mState == TrackBase::RESUMING) { 2712 track->mState = TrackBase::ACTIVE; 2713 rampVolume = true; 2714 } 2715 } else if (cblk->server != 0) { 2716 // If the track is stopped before the first frame was mixed, 2717 // do not apply ramp 2718 rampVolume = true; 2719 } 2720 // compute volume for this track 2721 float left, right; 2722 if (track->isMuted() || mMasterMute || track->isPausing() || 2723 mStreamTypes[track->type()].mute) { 2724 left = right = 0; 2725 if (track->isPausing()) { 2726 track->setPaused(); 2727 } 2728 } else { 2729 float typeVolume = mStreamTypes[track->type()].volume; 2730 float v = mMasterVolume * typeVolume; 2731 uint32_t vlr = cblk->volumeLR; 2732 float v_clamped = v * (vlr & 0xFFFF); 2733 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2734 left = v_clamped/MAX_GAIN; 2735 v_clamped = v * (vlr >> 16); 2736 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2737 right = v_clamped/MAX_GAIN; 2738 } 2739 2740 if (left != mLeftVolFloat || right != mRightVolFloat) { 2741 mLeftVolFloat = left; 2742 mRightVolFloat = right; 2743 2744 // If audio HAL implements volume control, 2745 // force software volume to nominal value 2746 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2747 left = 1.0f; 2748 right = 1.0f; 2749 } 2750 2751 // Convert volumes from float to 8.24 2752 uint32_t vl = (uint32_t)(left * (1 << 24)); 2753 uint32_t vr = (uint32_t)(right * (1 << 24)); 2754 2755 // Delegate volume control to effect in track effect chain if needed 2756 // only one effect chain can be present on DirectOutputThread, so if 2757 // there is one, the track is connected to it 2758 if (!effectChains.isEmpty()) { 2759 // Do not ramp volume if volume is controlled by effect 2760 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2761 rampVolume = false; 2762 } 2763 } 2764 2765 // Convert volumes from 8.24 to 4.12 format 2766 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2767 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2768 leftVol = (uint16_t)v_clamped; 2769 v_clamped = (vr + (1 << 11)) >> 12; 2770 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2771 rightVol = (uint16_t)v_clamped; 2772 } else { 2773 leftVol = mLeftVolShort; 2774 rightVol = mRightVolShort; 2775 rampVolume = false; 2776 } 2777 2778 // reset retry count 2779 track->mRetryCount = kMaxTrackRetriesDirect; 2780 activeTrack = t; 2781 mixerStatus = MIXER_TRACKS_READY; 2782 } else { 2783 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2784 if (track->isStopped()) { 2785 track->reset(); 2786 } 2787 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2788 // We have consumed all the buffers of this track. 2789 // Remove it from the list of active tracks. 2790 trackToRemove = track; 2791 } else { 2792 // No buffers for this track. Give it a few chances to 2793 // fill a buffer, then remove it from active list. 2794 if (--(track->mRetryCount) <= 0) { 2795 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2796 trackToRemove = track; 2797 } else { 2798 mixerStatus = MIXER_TRACKS_ENABLED; 2799 } 2800 } 2801 } 2802 } 2803 2804 // remove all the tracks that need to be... 2805 if (CC_UNLIKELY(trackToRemove != 0)) { 2806 mActiveTracks.remove(trackToRemove); 2807 if (!effectChains.isEmpty()) { 2808 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2809 trackToRemove->sessionId()); 2810 effectChains[0]->decActiveTrackCnt(); 2811 } 2812 if (trackToRemove->isTerminated()) { 2813 removeTrack_l(trackToRemove); 2814 } 2815 } 2816 2817 lockEffectChains_l(effectChains); 2818 } 2819 2820 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2821 AudioBufferProvider::Buffer buffer; 2822 size_t frameCount = mFrameCount; 2823 curBuf = (int8_t *)mMixBuffer; 2824 // output audio to hardware 2825 while (frameCount) { 2826 buffer.frameCount = frameCount; 2827 activeTrack->getNextBuffer(&buffer); 2828 if (CC_UNLIKELY(buffer.raw == NULL)) { 2829 memset(curBuf, 0, frameCount * mFrameSize); 2830 break; 2831 } 2832 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2833 frameCount -= buffer.frameCount; 2834 curBuf += buffer.frameCount * mFrameSize; 2835 activeTrack->releaseBuffer(&buffer); 2836 } 2837 sleepTime = 0; 2838 standbyTime = systemTime() + standbyDelay; 2839 } else { 2840 if (sleepTime == 0) { 2841 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2842 sleepTime = activeSleepTime; 2843 } else { 2844 sleepTime = idleSleepTime; 2845 } 2846 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2847 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2848 sleepTime = 0; 2849 } 2850 } 2851 2852 if (mSuspended) { 2853 sleepTime = suspendSleepTimeUs(); 2854 } 2855 // sleepTime == 0 means we must write to audio hardware 2856 if (sleepTime == 0) { 2857 if (mixerStatus == MIXER_TRACKS_READY) { 2858 applyVolume(leftVol, rightVol, rampVolume); 2859 } 2860 for (size_t i = 0; i < effectChains.size(); i ++) { 2861 effectChains[i]->process_l(); 2862 } 2863 unlockEffectChains(effectChains); 2864 2865 mLastWriteTime = systemTime(); 2866 mInWrite = true; 2867 mBytesWritten += mixBufferSize; 2868 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2869 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2870 mNumWrites++; 2871 mInWrite = false; 2872 mStandby = false; 2873 } else { 2874 unlockEffectChains(effectChains); 2875 usleep(sleepTime); 2876 } 2877 2878 // finally let go of removed track, without the lock held 2879 // since we can't guarantee the destructors won't acquire that 2880 // same lock. 2881 trackToRemove.clear(); 2882 activeTrack.clear(); 2883 2884 // Effect chains will be actually deleted here if they were removed from 2885 // mEffectChains list during mixing or effects processing 2886 effectChains.clear(); 2887 } 2888 2889 if (!mStandby) { 2890 mOutput->stream->common.standby(&mOutput->stream->common); 2891 } 2892 2893 releaseWakeLock(); 2894 2895 ALOGV("DirectOutputThread %p exiting", this); 2896 return false; 2897} 2898 2899// getTrackName_l() must be called with ThreadBase::mLock held 2900int AudioFlinger::DirectOutputThread::getTrackName_l() 2901{ 2902 return 0; 2903} 2904 2905// deleteTrackName_l() must be called with ThreadBase::mLock held 2906void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2907{ 2908} 2909 2910// checkForNewParameters_l() must be called with ThreadBase::mLock held 2911bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2912{ 2913 bool reconfig = false; 2914 2915 while (!mNewParameters.isEmpty()) { 2916 status_t status = NO_ERROR; 2917 String8 keyValuePair = mNewParameters[0]; 2918 AudioParameter param = AudioParameter(keyValuePair); 2919 int value; 2920 2921 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2922 // do not accept frame count changes if tracks are open as the track buffer 2923 // size depends on frame count and correct behavior would not be garantied 2924 // if frame count is changed after track creation 2925 if (!mTracks.isEmpty()) { 2926 status = INVALID_OPERATION; 2927 } else { 2928 reconfig = true; 2929 } 2930 } 2931 if (status == NO_ERROR) { 2932 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2933 keyValuePair.string()); 2934 if (!mStandby && status == INVALID_OPERATION) { 2935 mOutput->stream->common.standby(&mOutput->stream->common); 2936 mStandby = true; 2937 mBytesWritten = 0; 2938 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2939 keyValuePair.string()); 2940 } 2941 if (status == NO_ERROR && reconfig) { 2942 readOutputParameters(); 2943 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2944 } 2945 } 2946 2947 mNewParameters.removeAt(0); 2948 2949 mParamStatus = status; 2950 mParamCond.signal(); 2951 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2952 // already timed out waiting for the status and will never signal the condition. 2953 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2954 } 2955 return reconfig; 2956} 2957 2958uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2959{ 2960 uint32_t time; 2961 if (audio_is_linear_pcm(mFormat)) { 2962 time = PlaybackThread::activeSleepTimeUs(); 2963 } else { 2964 time = 10000; 2965 } 2966 return time; 2967} 2968 2969uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2970{ 2971 uint32_t time; 2972 if (audio_is_linear_pcm(mFormat)) { 2973 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2974 } else { 2975 time = 10000; 2976 } 2977 return time; 2978} 2979 2980uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2981{ 2982 uint32_t time; 2983 if (audio_is_linear_pcm(mFormat)) { 2984 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2985 } else { 2986 time = 10000; 2987 } 2988 return time; 2989} 2990 2991 2992// ---------------------------------------------------------------------------- 2993 2994AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2995 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2996{ 2997 mType = ThreadBase::DUPLICATING; 2998 addOutputTrack(mainThread); 2999} 3000 3001AudioFlinger::DuplicatingThread::~DuplicatingThread() 3002{ 3003 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3004 mOutputTracks[i]->destroy(); 3005 } 3006 mOutputTracks.clear(); 3007} 3008 3009bool AudioFlinger::DuplicatingThread::threadLoop() 3010{ 3011 Vector< sp<Track> > tracksToRemove; 3012 mixer_state mixerStatus = MIXER_IDLE; 3013 nsecs_t standbyTime = systemTime(); 3014 size_t mixBufferSize = mFrameCount*mFrameSize; 3015 SortedVector< sp<OutputTrack> > outputTracks; 3016 uint32_t writeFrames = 0; 3017 uint32_t activeSleepTime = activeSleepTimeUs(); 3018 uint32_t idleSleepTime = idleSleepTimeUs(); 3019 uint32_t sleepTime = idleSleepTime; 3020 Vector< sp<EffectChain> > effectChains; 3021 3022 acquireWakeLock(); 3023 3024 while (!exitPending()) 3025 { 3026 processConfigEvents(); 3027 3028 mixerStatus = MIXER_IDLE; 3029 { // scope for the mLock 3030 3031 Mutex::Autolock _l(mLock); 3032 3033 if (checkForNewParameters_l()) { 3034 mixBufferSize = mFrameCount*mFrameSize; 3035 updateWaitTime(); 3036 activeSleepTime = activeSleepTimeUs(); 3037 idleSleepTime = idleSleepTimeUs(); 3038 } 3039 3040 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3041 3042 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3043 outputTracks.add(mOutputTracks[i]); 3044 } 3045 3046 // put audio hardware into standby after short delay 3047 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3048 mSuspended)) { 3049 if (!mStandby) { 3050 for (size_t i = 0; i < outputTracks.size(); i++) { 3051 outputTracks[i]->stop(); 3052 } 3053 mStandby = true; 3054 mBytesWritten = 0; 3055 } 3056 3057 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3058 // we're about to wait, flush the binder command buffer 3059 IPCThreadState::self()->flushCommands(); 3060 outputTracks.clear(); 3061 3062 if (exitPending()) break; 3063 3064 releaseWakeLock_l(); 3065 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3066 mWaitWorkCV.wait(mLock); 3067 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3068 acquireWakeLock_l(); 3069 3070 mPrevMixerStatus = MIXER_IDLE; 3071 if (!mMasterMute) { 3072 char value[PROPERTY_VALUE_MAX]; 3073 property_get("ro.audio.silent", value, "0"); 3074 if (atoi(value)) { 3075 ALOGD("Silence is golden"); 3076 setMasterMute(true); 3077 } 3078 } 3079 3080 standbyTime = systemTime() + kStandbyTimeInNsecs; 3081 sleepTime = idleSleepTime; 3082 continue; 3083 } 3084 } 3085 3086 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3087 3088 // prevent any changes in effect chain list and in each effect chain 3089 // during mixing and effect process as the audio buffers could be deleted 3090 // or modified if an effect is created or deleted 3091 lockEffectChains_l(effectChains); 3092 } 3093 3094 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3095 // mix buffers... 3096 if (outputsReady(outputTracks)) { 3097 mAudioMixer->process(); 3098 } else { 3099 memset(mMixBuffer, 0, mixBufferSize); 3100 } 3101 sleepTime = 0; 3102 writeFrames = mFrameCount; 3103 } else { 3104 if (sleepTime == 0) { 3105 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3106 sleepTime = activeSleepTime; 3107 } else { 3108 sleepTime = idleSleepTime; 3109 } 3110 } else if (mBytesWritten != 0) { 3111 // flush remaining overflow buffers in output tracks 3112 for (size_t i = 0; i < outputTracks.size(); i++) { 3113 if (outputTracks[i]->isActive()) { 3114 sleepTime = 0; 3115 writeFrames = 0; 3116 memset(mMixBuffer, 0, mixBufferSize); 3117 break; 3118 } 3119 } 3120 } 3121 } 3122 3123 if (mSuspended) { 3124 sleepTime = suspendSleepTimeUs(); 3125 } 3126 // sleepTime == 0 means we must write to audio hardware 3127 if (sleepTime == 0) { 3128 for (size_t i = 0; i < effectChains.size(); i ++) { 3129 effectChains[i]->process_l(); 3130 } 3131 // enable changes in effect chain 3132 unlockEffectChains(effectChains); 3133 3134 standbyTime = systemTime() + kStandbyTimeInNsecs; 3135 for (size_t i = 0; i < outputTracks.size(); i++) { 3136 outputTracks[i]->write(mMixBuffer, writeFrames); 3137 } 3138 mStandby = false; 3139 mBytesWritten += mixBufferSize; 3140 } else { 3141 // enable changes in effect chain 3142 unlockEffectChains(effectChains); 3143 usleep(sleepTime); 3144 } 3145 3146 // finally let go of all our tracks, without the lock held 3147 // since we can't guarantee the destructors won't acquire that 3148 // same lock. 3149 tracksToRemove.clear(); 3150 outputTracks.clear(); 3151 3152 // Effect chains will be actually deleted here if they were removed from 3153 // mEffectChains list during mixing or effects processing 3154 effectChains.clear(); 3155 } 3156 3157 releaseWakeLock(); 3158 3159 return false; 3160} 3161 3162void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3163{ 3164 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3165 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3166 this, 3167 mSampleRate, 3168 mFormat, 3169 mChannelMask, 3170 frameCount); 3171 if (outputTrack->cblk() != NULL) { 3172 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3173 mOutputTracks.add(outputTrack); 3174 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3175 updateWaitTime(); 3176 } 3177} 3178 3179void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3180{ 3181 Mutex::Autolock _l(mLock); 3182 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3183 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3184 mOutputTracks[i]->destroy(); 3185 mOutputTracks.removeAt(i); 3186 updateWaitTime(); 3187 return; 3188 } 3189 } 3190 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3191} 3192 3193void AudioFlinger::DuplicatingThread::updateWaitTime() 3194{ 3195 mWaitTimeMs = UINT_MAX; 3196 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3197 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3198 if (strong != NULL) { 3199 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3200 if (waitTimeMs < mWaitTimeMs) { 3201 mWaitTimeMs = waitTimeMs; 3202 } 3203 } 3204 } 3205} 3206 3207 3208bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3209{ 3210 for (size_t i = 0; i < outputTracks.size(); i++) { 3211 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3212 if (thread == 0) { 3213 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3214 return false; 3215 } 3216 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3217 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3218 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3219 return false; 3220 } 3221 } 3222 return true; 3223} 3224 3225uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3226{ 3227 return (mWaitTimeMs * 1000) / 2; 3228} 3229 3230// ---------------------------------------------------------------------------- 3231 3232// TrackBase constructor must be called with AudioFlinger::mLock held 3233AudioFlinger::ThreadBase::TrackBase::TrackBase( 3234 const wp<ThreadBase>& thread, 3235 const sp<Client>& client, 3236 uint32_t sampleRate, 3237 audio_format_t format, 3238 uint32_t channelMask, 3239 int frameCount, 3240 uint32_t flags, 3241 const sp<IMemory>& sharedBuffer, 3242 int sessionId) 3243 : RefBase(), 3244 mThread(thread), 3245 mClient(client), 3246 mCblk(0), 3247 mFrameCount(0), 3248 mState(IDLE), 3249 mClientTid(-1), 3250 mFormat(format), 3251 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3252 mSessionId(sessionId) 3253{ 3254 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3255 3256 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3257 size_t size = sizeof(audio_track_cblk_t); 3258 uint8_t channelCount = popcount(channelMask); 3259 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3260 if (sharedBuffer == 0) { 3261 size += bufferSize; 3262 } 3263 3264 if (client != NULL) { 3265 mCblkMemory = client->heap()->allocate(size); 3266 if (mCblkMemory != 0) { 3267 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3268 if (mCblk) { // construct the shared structure in-place. 3269 new(mCblk) audio_track_cblk_t(); 3270 // clear all buffers 3271 mCblk->frameCount = frameCount; 3272 mCblk->sampleRate = sampleRate; 3273 mChannelCount = channelCount; 3274 mChannelMask = channelMask; 3275 if (sharedBuffer == 0) { 3276 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3277 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3278 // Force underrun condition to avoid false underrun callback until first data is 3279 // written to buffer (other flags are cleared) 3280 mCblk->flags = CBLK_UNDERRUN_ON; 3281 } else { 3282 mBuffer = sharedBuffer->pointer(); 3283 } 3284 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3285 } 3286 } else { 3287 ALOGE("not enough memory for AudioTrack size=%u", size); 3288 client->heap()->dump("AudioTrack"); 3289 return; 3290 } 3291 } else { 3292 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3293 // construct the shared structure in-place. 3294 new(mCblk) audio_track_cblk_t(); 3295 // clear all buffers 3296 mCblk->frameCount = frameCount; 3297 mCblk->sampleRate = sampleRate; 3298 mChannelCount = channelCount; 3299 mChannelMask = channelMask; 3300 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3301 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3302 // Force underrun condition to avoid false underrun callback until first data is 3303 // written to buffer (other flags are cleared) 3304 mCblk->flags = CBLK_UNDERRUN_ON; 3305 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3306 } 3307} 3308 3309AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3310{ 3311 if (mCblk) { 3312 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3313 if (mClient == NULL) { 3314 delete mCblk; 3315 } 3316 } 3317 mCblkMemory.clear(); // and free the shared memory 3318 if (mClient != NULL) { 3319 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3320 mClient.clear(); 3321 } 3322} 3323 3324void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3325{ 3326 buffer->raw = NULL; 3327 mFrameCount = buffer->frameCount; 3328 step(); 3329 buffer->frameCount = 0; 3330} 3331 3332bool AudioFlinger::ThreadBase::TrackBase::step() { 3333 bool result; 3334 audio_track_cblk_t* cblk = this->cblk(); 3335 3336 result = cblk->stepServer(mFrameCount); 3337 if (!result) { 3338 ALOGV("stepServer failed acquiring cblk mutex"); 3339 mFlags |= STEPSERVER_FAILED; 3340 } 3341 return result; 3342} 3343 3344void AudioFlinger::ThreadBase::TrackBase::reset() { 3345 audio_track_cblk_t* cblk = this->cblk(); 3346 3347 cblk->user = 0; 3348 cblk->server = 0; 3349 cblk->userBase = 0; 3350 cblk->serverBase = 0; 3351 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3352 ALOGV("TrackBase::reset"); 3353} 3354 3355sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3356{ 3357 return mCblkMemory; 3358} 3359 3360int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3361 return (int)mCblk->sampleRate; 3362} 3363 3364int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3365 return (const int)mChannelCount; 3366} 3367 3368uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3369 return mChannelMask; 3370} 3371 3372void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3373 audio_track_cblk_t* cblk = this->cblk(); 3374 size_t frameSize = cblk->frameSize; 3375 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3376 int8_t *bufferEnd = bufferStart + frames * frameSize; 3377 3378 // Check validity of returned pointer in case the track control block would have been corrupted. 3379 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3380 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3381 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3382 server %d, serverBase %d, user %d, userBase %d", 3383 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3384 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3385 return 0; 3386 } 3387 3388 return bufferStart; 3389} 3390 3391// ---------------------------------------------------------------------------- 3392 3393// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3394AudioFlinger::PlaybackThread::Track::Track( 3395 const wp<ThreadBase>& thread, 3396 const sp<Client>& client, 3397 audio_stream_type_t streamType, 3398 uint32_t sampleRate, 3399 audio_format_t format, 3400 uint32_t channelMask, 3401 int frameCount, 3402 const sp<IMemory>& sharedBuffer, 3403 int sessionId) 3404 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3405 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3406 mAuxEffectId(0), mHasVolumeController(false) 3407{ 3408 if (mCblk != NULL) { 3409 sp<ThreadBase> baseThread = thread.promote(); 3410 if (baseThread != 0) { 3411 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3412 mName = playbackThread->getTrackName_l(); 3413 mMainBuffer = playbackThread->mixBuffer(); 3414 } 3415 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3416 if (mName < 0) { 3417 ALOGE("no more track names available"); 3418 } 3419 mStreamType = streamType; 3420 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3421 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3422 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3423 } 3424} 3425 3426AudioFlinger::PlaybackThread::Track::~Track() 3427{ 3428 ALOGV("PlaybackThread::Track destructor"); 3429 sp<ThreadBase> thread = mThread.promote(); 3430 if (thread != 0) { 3431 Mutex::Autolock _l(thread->mLock); 3432 mState = TERMINATED; 3433 } 3434} 3435 3436void AudioFlinger::PlaybackThread::Track::destroy() 3437{ 3438 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3439 // by removing it from mTracks vector, so there is a risk that this Tracks's 3440 // desctructor is called. As the destructor needs to lock mLock, 3441 // we must acquire a strong reference on this Track before locking mLock 3442 // here so that the destructor is called only when exiting this function. 3443 // On the other hand, as long as Track::destroy() is only called by 3444 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3445 // this Track with its member mTrack. 3446 sp<Track> keep(this); 3447 { // scope for mLock 3448 sp<ThreadBase> thread = mThread.promote(); 3449 if (thread != 0) { 3450 if (!isOutputTrack()) { 3451 if (mState == ACTIVE || mState == RESUMING) { 3452 AudioSystem::stopOutput(thread->id(), 3453 (audio_stream_type_t)mStreamType, 3454 mSessionId); 3455 3456 // to track the speaker usage 3457 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3458 } 3459 AudioSystem::releaseOutput(thread->id()); 3460 } 3461 Mutex::Autolock _l(thread->mLock); 3462 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3463 playbackThread->destroyTrack_l(this); 3464 } 3465 } 3466} 3467 3468void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3469{ 3470 uint32_t vlr = mCblk->volumeLR; 3471 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", 3472 mName - AudioMixer::TRACK0, 3473 (mClient == NULL) ? getpid() : mClient->pid(), 3474 mStreamType, 3475 mFormat, 3476 mChannelMask, 3477 mSessionId, 3478 mFrameCount, 3479 mState, 3480 mMute, 3481 mFillingUpStatus, 3482 mCblk->sampleRate, 3483 vlr & 0xFFFF, 3484 vlr >> 16, 3485 mCblk->server, 3486 mCblk->user, 3487 (int)mMainBuffer, 3488 (int)mAuxBuffer); 3489} 3490 3491status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3492{ 3493 audio_track_cblk_t* cblk = this->cblk(); 3494 uint32_t framesReady; 3495 uint32_t framesReq = buffer->frameCount; 3496 3497 // Check if last stepServer failed, try to step now 3498 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3499 if (!step()) goto getNextBuffer_exit; 3500 ALOGV("stepServer recovered"); 3501 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3502 } 3503 3504 framesReady = cblk->framesReady(); 3505 3506 if (CC_LIKELY(framesReady)) { 3507 uint32_t s = cblk->server; 3508 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3509 3510 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3511 if (framesReq > framesReady) { 3512 framesReq = framesReady; 3513 } 3514 if (s + framesReq > bufferEnd) { 3515 framesReq = bufferEnd - s; 3516 } 3517 3518 buffer->raw = getBuffer(s, framesReq); 3519 if (buffer->raw == NULL) goto getNextBuffer_exit; 3520 3521 buffer->frameCount = framesReq; 3522 return NO_ERROR; 3523 } 3524 3525getNextBuffer_exit: 3526 buffer->raw = NULL; 3527 buffer->frameCount = 0; 3528 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3529 return NOT_ENOUGH_DATA; 3530} 3531 3532bool AudioFlinger::PlaybackThread::Track::isReady() const { 3533 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3534 3535 if (mCblk->framesReady() >= mCblk->frameCount || 3536 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3537 mFillingUpStatus = FS_FILLED; 3538 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3539 return true; 3540 } 3541 return false; 3542} 3543 3544status_t AudioFlinger::PlaybackThread::Track::start() 3545{ 3546 status_t status = NO_ERROR; 3547 ALOGV("start(%d), calling thread %d session %d", 3548 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3549 sp<ThreadBase> thread = mThread.promote(); 3550 if (thread != 0) { 3551 Mutex::Autolock _l(thread->mLock); 3552 track_state state = mState; 3553 // here the track could be either new, or restarted 3554 // in both cases "unstop" the track 3555 if (mState == PAUSED) { 3556 mState = TrackBase::RESUMING; 3557 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3558 } else { 3559 mState = TrackBase::ACTIVE; 3560 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3561 } 3562 3563 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3564 thread->mLock.unlock(); 3565 status = AudioSystem::startOutput(thread->id(), 3566 (audio_stream_type_t)mStreamType, 3567 mSessionId); 3568 thread->mLock.lock(); 3569 3570 // to track the speaker usage 3571 if (status == NO_ERROR) { 3572 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3573 } 3574 } 3575 if (status == NO_ERROR) { 3576 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3577 playbackThread->addTrack_l(this); 3578 } else { 3579 mState = state; 3580 } 3581 } else { 3582 status = BAD_VALUE; 3583 } 3584 return status; 3585} 3586 3587void AudioFlinger::PlaybackThread::Track::stop() 3588{ 3589 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3590 sp<ThreadBase> thread = mThread.promote(); 3591 if (thread != 0) { 3592 Mutex::Autolock _l(thread->mLock); 3593 track_state state = mState; 3594 if (mState > STOPPED) { 3595 mState = STOPPED; 3596 // If the track is not active (PAUSED and buffers full), flush buffers 3597 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3598 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3599 reset(); 3600 } 3601 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3602 } 3603 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3604 thread->mLock.unlock(); 3605 AudioSystem::stopOutput(thread->id(), 3606 (audio_stream_type_t)mStreamType, 3607 mSessionId); 3608 thread->mLock.lock(); 3609 3610 // to track the speaker usage 3611 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3612 } 3613 } 3614} 3615 3616void AudioFlinger::PlaybackThread::Track::pause() 3617{ 3618 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3619 sp<ThreadBase> thread = mThread.promote(); 3620 if (thread != 0) { 3621 Mutex::Autolock _l(thread->mLock); 3622 if (mState == ACTIVE || mState == RESUMING) { 3623 mState = PAUSING; 3624 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3625 if (!isOutputTrack()) { 3626 thread->mLock.unlock(); 3627 AudioSystem::stopOutput(thread->id(), 3628 (audio_stream_type_t)mStreamType, 3629 mSessionId); 3630 thread->mLock.lock(); 3631 3632 // to track the speaker usage 3633 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3634 } 3635 } 3636 } 3637} 3638 3639void AudioFlinger::PlaybackThread::Track::flush() 3640{ 3641 ALOGV("flush(%d)", mName); 3642 sp<ThreadBase> thread = mThread.promote(); 3643 if (thread != 0) { 3644 Mutex::Autolock _l(thread->mLock); 3645 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3646 return; 3647 } 3648 // No point remaining in PAUSED state after a flush => go to 3649 // STOPPED state 3650 mState = STOPPED; 3651 3652 // do not reset the track if it is still in the process of being stopped or paused. 3653 // this will be done by prepareTracks_l() when the track is stopped. 3654 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3655 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3656 reset(); 3657 } 3658 } 3659} 3660 3661void AudioFlinger::PlaybackThread::Track::reset() 3662{ 3663 // Do not reset twice to avoid discarding data written just after a flush and before 3664 // the audioflinger thread detects the track is stopped. 3665 if (!mResetDone) { 3666 TrackBase::reset(); 3667 // Force underrun condition to avoid false underrun callback until first data is 3668 // written to buffer 3669 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3670 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3671 mFillingUpStatus = FS_FILLING; 3672 mResetDone = true; 3673 } 3674} 3675 3676void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3677{ 3678 mMute = muted; 3679} 3680 3681status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3682{ 3683 status_t status = DEAD_OBJECT; 3684 sp<ThreadBase> thread = mThread.promote(); 3685 if (thread != 0) { 3686 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3687 status = playbackThread->attachAuxEffect(this, EffectId); 3688 } 3689 return status; 3690} 3691 3692void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3693{ 3694 mAuxEffectId = EffectId; 3695 mAuxBuffer = buffer; 3696} 3697 3698// ---------------------------------------------------------------------------- 3699 3700// RecordTrack constructor must be called with AudioFlinger::mLock held 3701AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3702 const wp<ThreadBase>& thread, 3703 const sp<Client>& client, 3704 uint32_t sampleRate, 3705 audio_format_t format, 3706 uint32_t channelMask, 3707 int frameCount, 3708 uint32_t flags, 3709 int sessionId) 3710 : TrackBase(thread, client, sampleRate, format, 3711 channelMask, frameCount, flags, 0, sessionId), 3712 mOverflow(false) 3713{ 3714 if (mCblk != NULL) { 3715 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3716 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3717 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3718 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3719 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3720 } else { 3721 mCblk->frameSize = sizeof(int8_t); 3722 } 3723 } 3724} 3725 3726AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3727{ 3728 sp<ThreadBase> thread = mThread.promote(); 3729 if (thread != 0) { 3730 AudioSystem::releaseInput(thread->id()); 3731 } 3732} 3733 3734status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3735{ 3736 audio_track_cblk_t* cblk = this->cblk(); 3737 uint32_t framesAvail; 3738 uint32_t framesReq = buffer->frameCount; 3739 3740 // Check if last stepServer failed, try to step now 3741 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3742 if (!step()) goto getNextBuffer_exit; 3743 ALOGV("stepServer recovered"); 3744 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3745 } 3746 3747 framesAvail = cblk->framesAvailable_l(); 3748 3749 if (CC_LIKELY(framesAvail)) { 3750 uint32_t s = cblk->server; 3751 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3752 3753 if (framesReq > framesAvail) { 3754 framesReq = framesAvail; 3755 } 3756 if (s + framesReq > bufferEnd) { 3757 framesReq = bufferEnd - s; 3758 } 3759 3760 buffer->raw = getBuffer(s, framesReq); 3761 if (buffer->raw == NULL) goto getNextBuffer_exit; 3762 3763 buffer->frameCount = framesReq; 3764 return NO_ERROR; 3765 } 3766 3767getNextBuffer_exit: 3768 buffer->raw = NULL; 3769 buffer->frameCount = 0; 3770 return NOT_ENOUGH_DATA; 3771} 3772 3773status_t AudioFlinger::RecordThread::RecordTrack::start() 3774{ 3775 sp<ThreadBase> thread = mThread.promote(); 3776 if (thread != 0) { 3777 RecordThread *recordThread = (RecordThread *)thread.get(); 3778 return recordThread->start(this); 3779 } else { 3780 return BAD_VALUE; 3781 } 3782} 3783 3784void AudioFlinger::RecordThread::RecordTrack::stop() 3785{ 3786 sp<ThreadBase> thread = mThread.promote(); 3787 if (thread != 0) { 3788 RecordThread *recordThread = (RecordThread *)thread.get(); 3789 recordThread->stop(this); 3790 TrackBase::reset(); 3791 // Force overerrun condition to avoid false overrun callback until first data is 3792 // read from buffer 3793 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3794 } 3795} 3796 3797void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3798{ 3799 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3800 (mClient == NULL) ? getpid() : mClient->pid(), 3801 mFormat, 3802 mChannelMask, 3803 mSessionId, 3804 mFrameCount, 3805 mState, 3806 mCblk->sampleRate, 3807 mCblk->server, 3808 mCblk->user); 3809} 3810 3811 3812// ---------------------------------------------------------------------------- 3813 3814AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3815 const wp<ThreadBase>& thread, 3816 DuplicatingThread *sourceThread, 3817 uint32_t sampleRate, 3818 audio_format_t format, 3819 uint32_t channelMask, 3820 int frameCount) 3821 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3822 mActive(false), mSourceThread(sourceThread) 3823{ 3824 3825 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3826 if (mCblk != NULL) { 3827 mCblk->flags |= CBLK_DIRECTION_OUT; 3828 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3829 mCblk->volumeLR = (MAX_GAIN_INT << 16) | MAX_GAIN_INT; 3830 mOutBuffer.frameCount = 0; 3831 playbackThread->mTracks.add(this); 3832 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3833 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3834 mCblk, mBuffer, mCblk->buffers, 3835 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3836 } else { 3837 ALOGW("Error creating output track on thread %p", playbackThread); 3838 } 3839} 3840 3841AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3842{ 3843 clearBufferQueue(); 3844} 3845 3846status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3847{ 3848 status_t status = Track::start(); 3849 if (status != NO_ERROR) { 3850 return status; 3851 } 3852 3853 mActive = true; 3854 mRetryCount = 127; 3855 return status; 3856} 3857 3858void AudioFlinger::PlaybackThread::OutputTrack::stop() 3859{ 3860 Track::stop(); 3861 clearBufferQueue(); 3862 mOutBuffer.frameCount = 0; 3863 mActive = false; 3864} 3865 3866bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3867{ 3868 Buffer *pInBuffer; 3869 Buffer inBuffer; 3870 uint32_t channelCount = mChannelCount; 3871 bool outputBufferFull = false; 3872 inBuffer.frameCount = frames; 3873 inBuffer.i16 = data; 3874 3875 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3876 3877 if (!mActive && frames != 0) { 3878 start(); 3879 sp<ThreadBase> thread = mThread.promote(); 3880 if (thread != 0) { 3881 MixerThread *mixerThread = (MixerThread *)thread.get(); 3882 if (mCblk->frameCount > frames){ 3883 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3884 uint32_t startFrames = (mCblk->frameCount - frames); 3885 pInBuffer = new Buffer; 3886 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3887 pInBuffer->frameCount = startFrames; 3888 pInBuffer->i16 = pInBuffer->mBuffer; 3889 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3890 mBufferQueue.add(pInBuffer); 3891 } else { 3892 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3893 } 3894 } 3895 } 3896 } 3897 3898 while (waitTimeLeftMs) { 3899 // First write pending buffers, then new data 3900 if (mBufferQueue.size()) { 3901 pInBuffer = mBufferQueue.itemAt(0); 3902 } else { 3903 pInBuffer = &inBuffer; 3904 } 3905 3906 if (pInBuffer->frameCount == 0) { 3907 break; 3908 } 3909 3910 if (mOutBuffer.frameCount == 0) { 3911 mOutBuffer.frameCount = pInBuffer->frameCount; 3912 nsecs_t startTime = systemTime(); 3913 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3914 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3915 outputBufferFull = true; 3916 break; 3917 } 3918 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3919 if (waitTimeLeftMs >= waitTimeMs) { 3920 waitTimeLeftMs -= waitTimeMs; 3921 } else { 3922 waitTimeLeftMs = 0; 3923 } 3924 } 3925 3926 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3927 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3928 mCblk->stepUser(outFrames); 3929 pInBuffer->frameCount -= outFrames; 3930 pInBuffer->i16 += outFrames * channelCount; 3931 mOutBuffer.frameCount -= outFrames; 3932 mOutBuffer.i16 += outFrames * channelCount; 3933 3934 if (pInBuffer->frameCount == 0) { 3935 if (mBufferQueue.size()) { 3936 mBufferQueue.removeAt(0); 3937 delete [] pInBuffer->mBuffer; 3938 delete pInBuffer; 3939 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3940 } else { 3941 break; 3942 } 3943 } 3944 } 3945 3946 // If we could not write all frames, allocate a buffer and queue it for next time. 3947 if (inBuffer.frameCount) { 3948 sp<ThreadBase> thread = mThread.promote(); 3949 if (thread != 0 && !thread->standby()) { 3950 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3951 pInBuffer = new Buffer; 3952 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3953 pInBuffer->frameCount = inBuffer.frameCount; 3954 pInBuffer->i16 = pInBuffer->mBuffer; 3955 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3956 mBufferQueue.add(pInBuffer); 3957 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3958 } else { 3959 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3960 } 3961 } 3962 } 3963 3964 // Calling write() with a 0 length buffer, means that no more data will be written: 3965 // If no more buffers are pending, fill output track buffer to make sure it is started 3966 // by output mixer. 3967 if (frames == 0 && mBufferQueue.size() == 0) { 3968 if (mCblk->user < mCblk->frameCount) { 3969 frames = mCblk->frameCount - mCblk->user; 3970 pInBuffer = new Buffer; 3971 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3972 pInBuffer->frameCount = frames; 3973 pInBuffer->i16 = pInBuffer->mBuffer; 3974 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3975 mBufferQueue.add(pInBuffer); 3976 } else if (mActive) { 3977 stop(); 3978 } 3979 } 3980 3981 return outputBufferFull; 3982} 3983 3984status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3985{ 3986 int active; 3987 status_t result; 3988 audio_track_cblk_t* cblk = mCblk; 3989 uint32_t framesReq = buffer->frameCount; 3990 3991// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3992 buffer->frameCount = 0; 3993 3994 uint32_t framesAvail = cblk->framesAvailable(); 3995 3996 3997 if (framesAvail == 0) { 3998 Mutex::Autolock _l(cblk->lock); 3999 goto start_loop_here; 4000 while (framesAvail == 0) { 4001 active = mActive; 4002 if (CC_UNLIKELY(!active)) { 4003 ALOGV("Not active and NO_MORE_BUFFERS"); 4004 return NO_MORE_BUFFERS; 4005 } 4006 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4007 if (result != NO_ERROR) { 4008 return NO_MORE_BUFFERS; 4009 } 4010 // read the server count again 4011 start_loop_here: 4012 framesAvail = cblk->framesAvailable_l(); 4013 } 4014 } 4015 4016// if (framesAvail < framesReq) { 4017// return NO_MORE_BUFFERS; 4018// } 4019 4020 if (framesReq > framesAvail) { 4021 framesReq = framesAvail; 4022 } 4023 4024 uint32_t u = cblk->user; 4025 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4026 4027 if (u + framesReq > bufferEnd) { 4028 framesReq = bufferEnd - u; 4029 } 4030 4031 buffer->frameCount = framesReq; 4032 buffer->raw = (void *)cblk->buffer(u); 4033 return NO_ERROR; 4034} 4035 4036 4037void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4038{ 4039 size_t size = mBufferQueue.size(); 4040 Buffer *pBuffer; 4041 4042 for (size_t i = 0; i < size; i++) { 4043 pBuffer = mBufferQueue.itemAt(i); 4044 delete [] pBuffer->mBuffer; 4045 delete pBuffer; 4046 } 4047 mBufferQueue.clear(); 4048} 4049 4050// ---------------------------------------------------------------------------- 4051 4052AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4053 : RefBase(), 4054 mAudioFlinger(audioFlinger), 4055 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4056 mPid(pid) 4057{ 4058 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4059} 4060 4061// Client destructor must be called with AudioFlinger::mLock held 4062AudioFlinger::Client::~Client() 4063{ 4064 mAudioFlinger->removeClient_l(mPid); 4065} 4066 4067const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4068{ 4069 return mMemoryDealer; 4070} 4071 4072// ---------------------------------------------------------------------------- 4073 4074AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4075 const sp<IAudioFlingerClient>& client, 4076 pid_t pid) 4077 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4078{ 4079} 4080 4081AudioFlinger::NotificationClient::~NotificationClient() 4082{ 4083 mClient.clear(); 4084} 4085 4086void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4087{ 4088 sp<NotificationClient> keep(this); 4089 { 4090 mAudioFlinger->removeNotificationClient(mPid); 4091 } 4092} 4093 4094// ---------------------------------------------------------------------------- 4095 4096AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4097 : BnAudioTrack(), 4098 mTrack(track) 4099{ 4100} 4101 4102AudioFlinger::TrackHandle::~TrackHandle() { 4103 // just stop the track on deletion, associated resources 4104 // will be freed from the main thread once all pending buffers have 4105 // been played. Unless it's not in the active track list, in which 4106 // case we free everything now... 4107 mTrack->destroy(); 4108} 4109 4110sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4111 return mTrack->getCblk(); 4112} 4113 4114status_t AudioFlinger::TrackHandle::start() { 4115 return mTrack->start(); 4116} 4117 4118void AudioFlinger::TrackHandle::stop() { 4119 mTrack->stop(); 4120} 4121 4122void AudioFlinger::TrackHandle::flush() { 4123 mTrack->flush(); 4124} 4125 4126void AudioFlinger::TrackHandle::mute(bool e) { 4127 mTrack->mute(e); 4128} 4129 4130void AudioFlinger::TrackHandle::pause() { 4131 mTrack->pause(); 4132} 4133 4134status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4135{ 4136 return mTrack->attachAuxEffect(EffectId); 4137} 4138 4139status_t AudioFlinger::TrackHandle::onTransact( 4140 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4141{ 4142 return BnAudioTrack::onTransact(code, data, reply, flags); 4143} 4144 4145// ---------------------------------------------------------------------------- 4146 4147sp<IAudioRecord> AudioFlinger::openRecord( 4148 pid_t pid, 4149 int input, 4150 uint32_t sampleRate, 4151 audio_format_t format, 4152 uint32_t channelMask, 4153 int frameCount, 4154 uint32_t flags, 4155 int *sessionId, 4156 status_t *status) 4157{ 4158 sp<RecordThread::RecordTrack> recordTrack; 4159 sp<RecordHandle> recordHandle; 4160 sp<Client> client; 4161 wp<Client> wclient; 4162 status_t lStatus; 4163 RecordThread *thread; 4164 size_t inFrameCount; 4165 int lSessionId; 4166 4167 // check calling permissions 4168 if (!recordingAllowed()) { 4169 lStatus = PERMISSION_DENIED; 4170 goto Exit; 4171 } 4172 4173 // add client to list 4174 { // scope for mLock 4175 Mutex::Autolock _l(mLock); 4176 thread = checkRecordThread_l(input); 4177 if (thread == NULL) { 4178 lStatus = BAD_VALUE; 4179 goto Exit; 4180 } 4181 4182 wclient = mClients.valueFor(pid); 4183 if (wclient != NULL) { 4184 client = wclient.promote(); 4185 } else { 4186 client = new Client(this, pid); 4187 mClients.add(pid, client); 4188 } 4189 4190 // If no audio session id is provided, create one here 4191 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4192 lSessionId = *sessionId; 4193 } else { 4194 lSessionId = nextUniqueId(); 4195 if (sessionId != NULL) { 4196 *sessionId = lSessionId; 4197 } 4198 } 4199 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4200 recordTrack = thread->createRecordTrack_l(client, 4201 sampleRate, 4202 format, 4203 channelMask, 4204 frameCount, 4205 flags, 4206 lSessionId, 4207 &lStatus); 4208 } 4209 if (lStatus != NO_ERROR) { 4210 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4211 // destructor is called by the TrackBase destructor with mLock held 4212 client.clear(); 4213 recordTrack.clear(); 4214 goto Exit; 4215 } 4216 4217 // return to handle to client 4218 recordHandle = new RecordHandle(recordTrack); 4219 lStatus = NO_ERROR; 4220 4221Exit: 4222 if (status) { 4223 *status = lStatus; 4224 } 4225 return recordHandle; 4226} 4227 4228// ---------------------------------------------------------------------------- 4229 4230AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4231 : BnAudioRecord(), 4232 mRecordTrack(recordTrack) 4233{ 4234} 4235 4236AudioFlinger::RecordHandle::~RecordHandle() { 4237 stop(); 4238} 4239 4240sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4241 return mRecordTrack->getCblk(); 4242} 4243 4244status_t AudioFlinger::RecordHandle::start() { 4245 ALOGV("RecordHandle::start()"); 4246 return mRecordTrack->start(); 4247} 4248 4249void AudioFlinger::RecordHandle::stop() { 4250 ALOGV("RecordHandle::stop()"); 4251 mRecordTrack->stop(); 4252} 4253 4254status_t AudioFlinger::RecordHandle::onTransact( 4255 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4256{ 4257 return BnAudioRecord::onTransact(code, data, reply, flags); 4258} 4259 4260// ---------------------------------------------------------------------------- 4261 4262AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4263 AudioStreamIn *input, 4264 uint32_t sampleRate, 4265 uint32_t channels, 4266 int id, 4267 uint32_t device) : 4268 ThreadBase(audioFlinger, id, device), 4269 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4270{ 4271 mType = ThreadBase::RECORD; 4272 4273 snprintf(mName, kNameLength, "AudioIn_%d", id); 4274 4275 mReqChannelCount = popcount(channels); 4276 mReqSampleRate = sampleRate; 4277 readInputParameters(); 4278} 4279 4280 4281AudioFlinger::RecordThread::~RecordThread() 4282{ 4283 delete[] mRsmpInBuffer; 4284 if (mResampler != NULL) { 4285 delete mResampler; 4286 delete[] mRsmpOutBuffer; 4287 } 4288} 4289 4290void AudioFlinger::RecordThread::onFirstRef() 4291{ 4292 run(mName, PRIORITY_URGENT_AUDIO); 4293} 4294 4295status_t AudioFlinger::RecordThread::readyToRun() 4296{ 4297 status_t status = initCheck(); 4298 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4299 return status; 4300} 4301 4302bool AudioFlinger::RecordThread::threadLoop() 4303{ 4304 AudioBufferProvider::Buffer buffer; 4305 sp<RecordTrack> activeTrack; 4306 Vector< sp<EffectChain> > effectChains; 4307 4308 nsecs_t lastWarning = 0; 4309 4310 acquireWakeLock(); 4311 4312 // start recording 4313 while (!exitPending()) { 4314 4315 processConfigEvents(); 4316 4317 { // scope for mLock 4318 Mutex::Autolock _l(mLock); 4319 checkForNewParameters_l(); 4320 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4321 if (!mStandby) { 4322 mInput->stream->common.standby(&mInput->stream->common); 4323 mStandby = true; 4324 } 4325 4326 if (exitPending()) break; 4327 4328 releaseWakeLock_l(); 4329 ALOGV("RecordThread: loop stopping"); 4330 // go to sleep 4331 mWaitWorkCV.wait(mLock); 4332 ALOGV("RecordThread: loop starting"); 4333 acquireWakeLock_l(); 4334 continue; 4335 } 4336 if (mActiveTrack != 0) { 4337 if (mActiveTrack->mState == TrackBase::PAUSING) { 4338 if (!mStandby) { 4339 mInput->stream->common.standby(&mInput->stream->common); 4340 mStandby = true; 4341 } 4342 mActiveTrack.clear(); 4343 mStartStopCond.broadcast(); 4344 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4345 if (mReqChannelCount != mActiveTrack->channelCount()) { 4346 mActiveTrack.clear(); 4347 mStartStopCond.broadcast(); 4348 } else if (mBytesRead != 0) { 4349 // record start succeeds only if first read from audio input 4350 // succeeds 4351 if (mBytesRead > 0) { 4352 mActiveTrack->mState = TrackBase::ACTIVE; 4353 } else { 4354 mActiveTrack.clear(); 4355 } 4356 mStartStopCond.broadcast(); 4357 } 4358 mStandby = false; 4359 } 4360 } 4361 lockEffectChains_l(effectChains); 4362 } 4363 4364 if (mActiveTrack != 0) { 4365 if (mActiveTrack->mState != TrackBase::ACTIVE && 4366 mActiveTrack->mState != TrackBase::RESUMING) { 4367 unlockEffectChains(effectChains); 4368 usleep(kRecordThreadSleepUs); 4369 continue; 4370 } 4371 for (size_t i = 0; i < effectChains.size(); i ++) { 4372 effectChains[i]->process_l(); 4373 } 4374 4375 buffer.frameCount = mFrameCount; 4376 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4377 size_t framesOut = buffer.frameCount; 4378 if (mResampler == NULL) { 4379 // no resampling 4380 while (framesOut) { 4381 size_t framesIn = mFrameCount - mRsmpInIndex; 4382 if (framesIn) { 4383 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4384 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4385 if (framesIn > framesOut) 4386 framesIn = framesOut; 4387 mRsmpInIndex += framesIn; 4388 framesOut -= framesIn; 4389 if ((int)mChannelCount == mReqChannelCount || 4390 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4391 memcpy(dst, src, framesIn * mFrameSize); 4392 } else { 4393 int16_t *src16 = (int16_t *)src; 4394 int16_t *dst16 = (int16_t *)dst; 4395 if (mChannelCount == 1) { 4396 while (framesIn--) { 4397 *dst16++ = *src16; 4398 *dst16++ = *src16++; 4399 } 4400 } else { 4401 while (framesIn--) { 4402 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4403 src16 += 2; 4404 } 4405 } 4406 } 4407 } 4408 if (framesOut && mFrameCount == mRsmpInIndex) { 4409 if (framesOut == mFrameCount && 4410 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4411 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4412 framesOut = 0; 4413 } else { 4414 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4415 mRsmpInIndex = 0; 4416 } 4417 if (mBytesRead < 0) { 4418 ALOGE("Error reading audio input"); 4419 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4420 // Force input into standby so that it tries to 4421 // recover at next read attempt 4422 mInput->stream->common.standby(&mInput->stream->common); 4423 usleep(kRecordThreadSleepUs); 4424 } 4425 mRsmpInIndex = mFrameCount; 4426 framesOut = 0; 4427 buffer.frameCount = 0; 4428 } 4429 } 4430 } 4431 } else { 4432 // resampling 4433 4434 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4435 // alter output frame count as if we were expecting stereo samples 4436 if (mChannelCount == 1 && mReqChannelCount == 1) { 4437 framesOut >>= 1; 4438 } 4439 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4440 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4441 // are 32 bit aligned which should be always true. 4442 if (mChannelCount == 2 && mReqChannelCount == 1) { 4443 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4444 // the resampler always outputs stereo samples: do post stereo to mono conversion 4445 int16_t *src = (int16_t *)mRsmpOutBuffer; 4446 int16_t *dst = buffer.i16; 4447 while (framesOut--) { 4448 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4449 src += 2; 4450 } 4451 } else { 4452 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4453 } 4454 4455 } 4456 mActiveTrack->releaseBuffer(&buffer); 4457 mActiveTrack->overflow(); 4458 } 4459 // client isn't retrieving buffers fast enough 4460 else { 4461 if (!mActiveTrack->setOverflow()) { 4462 nsecs_t now = systemTime(); 4463 if ((now - lastWarning) > kWarningThrottleNs) { 4464 ALOGW("RecordThread: buffer overflow"); 4465 lastWarning = now; 4466 } 4467 } 4468 // Release the processor for a while before asking for a new buffer. 4469 // This will give the application more chance to read from the buffer and 4470 // clear the overflow. 4471 usleep(kRecordThreadSleepUs); 4472 } 4473 } 4474 // enable changes in effect chain 4475 unlockEffectChains(effectChains); 4476 effectChains.clear(); 4477 } 4478 4479 if (!mStandby) { 4480 mInput->stream->common.standby(&mInput->stream->common); 4481 } 4482 mActiveTrack.clear(); 4483 4484 mStartStopCond.broadcast(); 4485 4486 releaseWakeLock(); 4487 4488 ALOGV("RecordThread %p exiting", this); 4489 return false; 4490} 4491 4492 4493sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4494 const sp<AudioFlinger::Client>& client, 4495 uint32_t sampleRate, 4496 audio_format_t format, 4497 int channelMask, 4498 int frameCount, 4499 uint32_t flags, 4500 int sessionId, 4501 status_t *status) 4502{ 4503 sp<RecordTrack> track; 4504 status_t lStatus; 4505 4506 lStatus = initCheck(); 4507 if (lStatus != NO_ERROR) { 4508 ALOGE("Audio driver not initialized."); 4509 goto Exit; 4510 } 4511 4512 { // scope for mLock 4513 Mutex::Autolock _l(mLock); 4514 4515 track = new RecordTrack(this, client, sampleRate, 4516 format, channelMask, frameCount, flags, sessionId); 4517 4518 if (track->getCblk() == NULL) { 4519 lStatus = NO_MEMORY; 4520 goto Exit; 4521 } 4522 4523 mTrack = track.get(); 4524 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4525 bool suspend = audio_is_bluetooth_sco_device( 4526 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4527 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4528 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4529 } 4530 lStatus = NO_ERROR; 4531 4532Exit: 4533 if (status) { 4534 *status = lStatus; 4535 } 4536 return track; 4537} 4538 4539status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4540{ 4541 ALOGV("RecordThread::start"); 4542 sp <ThreadBase> strongMe = this; 4543 status_t status = NO_ERROR; 4544 { 4545 AutoMutex lock(mLock); 4546 if (mActiveTrack != 0) { 4547 if (recordTrack != mActiveTrack.get()) { 4548 status = -EBUSY; 4549 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4550 mActiveTrack->mState = TrackBase::ACTIVE; 4551 } 4552 return status; 4553 } 4554 4555 recordTrack->mState = TrackBase::IDLE; 4556 mActiveTrack = recordTrack; 4557 mLock.unlock(); 4558 status_t status = AudioSystem::startInput(mId); 4559 mLock.lock(); 4560 if (status != NO_ERROR) { 4561 mActiveTrack.clear(); 4562 return status; 4563 } 4564 mRsmpInIndex = mFrameCount; 4565 mBytesRead = 0; 4566 if (mResampler != NULL) { 4567 mResampler->reset(); 4568 } 4569 mActiveTrack->mState = TrackBase::RESUMING; 4570 // signal thread to start 4571 ALOGV("Signal record thread"); 4572 mWaitWorkCV.signal(); 4573 // do not wait for mStartStopCond if exiting 4574 if (mExiting) { 4575 mActiveTrack.clear(); 4576 status = INVALID_OPERATION; 4577 goto startError; 4578 } 4579 mStartStopCond.wait(mLock); 4580 if (mActiveTrack == 0) { 4581 ALOGV("Record failed to start"); 4582 status = BAD_VALUE; 4583 goto startError; 4584 } 4585 ALOGV("Record started OK"); 4586 return status; 4587 } 4588startError: 4589 AudioSystem::stopInput(mId); 4590 return status; 4591} 4592 4593void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4594 ALOGV("RecordThread::stop"); 4595 sp <ThreadBase> strongMe = this; 4596 { 4597 AutoMutex lock(mLock); 4598 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4599 mActiveTrack->mState = TrackBase::PAUSING; 4600 // do not wait for mStartStopCond if exiting 4601 if (mExiting) { 4602 return; 4603 } 4604 mStartStopCond.wait(mLock); 4605 // if we have been restarted, recordTrack == mActiveTrack.get() here 4606 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4607 mLock.unlock(); 4608 AudioSystem::stopInput(mId); 4609 mLock.lock(); 4610 ALOGV("Record stopped OK"); 4611 } 4612 } 4613 } 4614} 4615 4616status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4617{ 4618 const size_t SIZE = 256; 4619 char buffer[SIZE]; 4620 String8 result; 4621 pid_t pid = 0; 4622 4623 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4624 result.append(buffer); 4625 4626 if (mActiveTrack != 0) { 4627 result.append("Active Track:\n"); 4628 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4629 mActiveTrack->dump(buffer, SIZE); 4630 result.append(buffer); 4631 4632 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4633 result.append(buffer); 4634 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4635 result.append(buffer); 4636 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4637 result.append(buffer); 4638 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4639 result.append(buffer); 4640 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4641 result.append(buffer); 4642 4643 4644 } else { 4645 result.append("No record client\n"); 4646 } 4647 write(fd, result.string(), result.size()); 4648 4649 dumpBase(fd, args); 4650 dumpEffectChains(fd, args); 4651 4652 return NO_ERROR; 4653} 4654 4655status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4656{ 4657 size_t framesReq = buffer->frameCount; 4658 size_t framesReady = mFrameCount - mRsmpInIndex; 4659 int channelCount; 4660 4661 if (framesReady == 0) { 4662 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4663 if (mBytesRead < 0) { 4664 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4665 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4666 // Force input into standby so that it tries to 4667 // recover at next read attempt 4668 mInput->stream->common.standby(&mInput->stream->common); 4669 usleep(kRecordThreadSleepUs); 4670 } 4671 buffer->raw = NULL; 4672 buffer->frameCount = 0; 4673 return NOT_ENOUGH_DATA; 4674 } 4675 mRsmpInIndex = 0; 4676 framesReady = mFrameCount; 4677 } 4678 4679 if (framesReq > framesReady) { 4680 framesReq = framesReady; 4681 } 4682 4683 if (mChannelCount == 1 && mReqChannelCount == 2) { 4684 channelCount = 1; 4685 } else { 4686 channelCount = 2; 4687 } 4688 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4689 buffer->frameCount = framesReq; 4690 return NO_ERROR; 4691} 4692 4693void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4694{ 4695 mRsmpInIndex += buffer->frameCount; 4696 buffer->frameCount = 0; 4697} 4698 4699bool AudioFlinger::RecordThread::checkForNewParameters_l() 4700{ 4701 bool reconfig = false; 4702 4703 while (!mNewParameters.isEmpty()) { 4704 status_t status = NO_ERROR; 4705 String8 keyValuePair = mNewParameters[0]; 4706 AudioParameter param = AudioParameter(keyValuePair); 4707 int value; 4708 audio_format_t reqFormat = mFormat; 4709 int reqSamplingRate = mReqSampleRate; 4710 int reqChannelCount = mReqChannelCount; 4711 4712 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4713 reqSamplingRate = value; 4714 reconfig = true; 4715 } 4716 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4717 reqFormat = (audio_format_t) value; 4718 reconfig = true; 4719 } 4720 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4721 reqChannelCount = popcount(value); 4722 reconfig = true; 4723 } 4724 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4725 // do not accept frame count changes if tracks are open as the track buffer 4726 // size depends on frame count and correct behavior would not be garantied 4727 // if frame count is changed after track creation 4728 if (mActiveTrack != 0) { 4729 status = INVALID_OPERATION; 4730 } else { 4731 reconfig = true; 4732 } 4733 } 4734 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4735 // forward device change to effects that have requested to be 4736 // aware of attached audio device. 4737 for (size_t i = 0; i < mEffectChains.size(); i++) { 4738 mEffectChains[i]->setDevice_l(value); 4739 } 4740 // store input device and output device but do not forward output device to audio HAL. 4741 // Note that status is ignored by the caller for output device 4742 // (see AudioFlinger::setParameters() 4743 if (value & AUDIO_DEVICE_OUT_ALL) { 4744 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4745 status = BAD_VALUE; 4746 } else { 4747 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4748 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4749 if (mTrack != NULL) { 4750 bool suspend = audio_is_bluetooth_sco_device( 4751 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4752 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4753 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4754 } 4755 } 4756 mDevice |= (uint32_t)value; 4757 } 4758 if (status == NO_ERROR) { 4759 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4760 if (status == INVALID_OPERATION) { 4761 mInput->stream->common.standby(&mInput->stream->common); 4762 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4763 } 4764 if (reconfig) { 4765 if (status == BAD_VALUE && 4766 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4767 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4768 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4769 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4770 (reqChannelCount < 3)) { 4771 status = NO_ERROR; 4772 } 4773 if (status == NO_ERROR) { 4774 readInputParameters(); 4775 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4776 } 4777 } 4778 } 4779 4780 mNewParameters.removeAt(0); 4781 4782 mParamStatus = status; 4783 mParamCond.signal(); 4784 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4785 // already timed out waiting for the status and will never signal the condition. 4786 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4787 } 4788 return reconfig; 4789} 4790 4791String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4792{ 4793 char *s; 4794 String8 out_s8 = String8(); 4795 4796 Mutex::Autolock _l(mLock); 4797 if (initCheck() != NO_ERROR) { 4798 return out_s8; 4799 } 4800 4801 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4802 out_s8 = String8(s); 4803 free(s); 4804 return out_s8; 4805} 4806 4807void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4808 AudioSystem::OutputDescriptor desc; 4809 void *param2 = 0; 4810 4811 switch (event) { 4812 case AudioSystem::INPUT_OPENED: 4813 case AudioSystem::INPUT_CONFIG_CHANGED: 4814 desc.channels = mChannelMask; 4815 desc.samplingRate = mSampleRate; 4816 desc.format = mFormat; 4817 desc.frameCount = mFrameCount; 4818 desc.latency = 0; 4819 param2 = &desc; 4820 break; 4821 4822 case AudioSystem::INPUT_CLOSED: 4823 default: 4824 break; 4825 } 4826 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4827} 4828 4829void AudioFlinger::RecordThread::readInputParameters() 4830{ 4831 if (mRsmpInBuffer) delete mRsmpInBuffer; 4832 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4833 if (mResampler) delete mResampler; 4834 mResampler = NULL; 4835 4836 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4837 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4838 mChannelCount = (uint16_t)popcount(mChannelMask); 4839 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4840 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4841 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4842 mFrameCount = mInputBytes / mFrameSize; 4843 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4844 4845 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4846 { 4847 int channelCount; 4848 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4849 // stereo to mono post process as the resampler always outputs stereo. 4850 if (mChannelCount == 1 && mReqChannelCount == 2) { 4851 channelCount = 1; 4852 } else { 4853 channelCount = 2; 4854 } 4855 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4856 mResampler->setSampleRate(mSampleRate); 4857 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4858 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4859 4860 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4861 if (mChannelCount == 1 && mReqChannelCount == 1) { 4862 mFrameCount >>= 1; 4863 } 4864 4865 } 4866 mRsmpInIndex = mFrameCount; 4867} 4868 4869unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4870{ 4871 Mutex::Autolock _l(mLock); 4872 if (initCheck() != NO_ERROR) { 4873 return 0; 4874 } 4875 4876 return mInput->stream->get_input_frames_lost(mInput->stream); 4877} 4878 4879uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4880{ 4881 Mutex::Autolock _l(mLock); 4882 uint32_t result = 0; 4883 if (getEffectChain_l(sessionId) != 0) { 4884 result = EFFECT_SESSION; 4885 } 4886 4887 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4888 result |= TRACK_SESSION; 4889 } 4890 4891 return result; 4892} 4893 4894AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4895{ 4896 Mutex::Autolock _l(mLock); 4897 return mTrack; 4898} 4899 4900AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4901{ 4902 Mutex::Autolock _l(mLock); 4903 return mInput; 4904} 4905 4906AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4907{ 4908 Mutex::Autolock _l(mLock); 4909 AudioStreamIn *input = mInput; 4910 mInput = NULL; 4911 return input; 4912} 4913 4914// this method must always be called either with ThreadBase mLock held or inside the thread loop 4915audio_stream_t* AudioFlinger::RecordThread::stream() 4916{ 4917 if (mInput == NULL) { 4918 return NULL; 4919 } 4920 return &mInput->stream->common; 4921} 4922 4923 4924// ---------------------------------------------------------------------------- 4925 4926int AudioFlinger::openOutput(uint32_t *pDevices, 4927 uint32_t *pSamplingRate, 4928 audio_format_t *pFormat, 4929 uint32_t *pChannels, 4930 uint32_t *pLatencyMs, 4931 uint32_t flags) 4932{ 4933 status_t status; 4934 PlaybackThread *thread = NULL; 4935 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4936 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4937 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4938 uint32_t channels = pChannels ? *pChannels : 0; 4939 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4940 audio_stream_out_t *outStream; 4941 audio_hw_device_t *outHwDev; 4942 4943 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4944 pDevices ? *pDevices : 0, 4945 samplingRate, 4946 format, 4947 channels, 4948 flags); 4949 4950 if (pDevices == NULL || *pDevices == 0) { 4951 return 0; 4952 } 4953 4954 Mutex::Autolock _l(mLock); 4955 4956 outHwDev = findSuitableHwDev_l(*pDevices); 4957 if (outHwDev == NULL) 4958 return 0; 4959 4960 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4961 &channels, &samplingRate, &outStream); 4962 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4963 outStream, 4964 samplingRate, 4965 format, 4966 channels, 4967 status); 4968 4969 mHardwareStatus = AUDIO_HW_IDLE; 4970 if (outStream != NULL) { 4971 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4972 int id = nextUniqueId(); 4973 4974 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4975 (format != AUDIO_FORMAT_PCM_16_BIT) || 4976 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4977 thread = new DirectOutputThread(this, output, id, *pDevices); 4978 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4979 } else { 4980 thread = new MixerThread(this, output, id, *pDevices); 4981 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4982 } 4983 mPlaybackThreads.add(id, thread); 4984 4985 if (pSamplingRate) *pSamplingRate = samplingRate; 4986 if (pFormat) *pFormat = format; 4987 if (pChannels) *pChannels = channels; 4988 if (pLatencyMs) *pLatencyMs = thread->latency(); 4989 4990 // notify client processes of the new output creation 4991 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4992 return id; 4993 } 4994 4995 return 0; 4996} 4997 4998int AudioFlinger::openDuplicateOutput(int output1, int output2) 4999{ 5000 Mutex::Autolock _l(mLock); 5001 MixerThread *thread1 = checkMixerThread_l(output1); 5002 MixerThread *thread2 = checkMixerThread_l(output2); 5003 5004 if (thread1 == NULL || thread2 == NULL) { 5005 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5006 return 0; 5007 } 5008 5009 int id = nextUniqueId(); 5010 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5011 thread->addOutputTrack(thread2); 5012 mPlaybackThreads.add(id, thread); 5013 // notify client processes of the new output creation 5014 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5015 return id; 5016} 5017 5018status_t AudioFlinger::closeOutput(int output) 5019{ 5020 // keep strong reference on the playback thread so that 5021 // it is not destroyed while exit() is executed 5022 sp <PlaybackThread> thread; 5023 { 5024 Mutex::Autolock _l(mLock); 5025 thread = checkPlaybackThread_l(output); 5026 if (thread == NULL) { 5027 return BAD_VALUE; 5028 } 5029 5030 ALOGV("closeOutput() %d", output); 5031 5032 if (thread->type() == ThreadBase::MIXER) { 5033 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5034 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5035 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5036 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5037 } 5038 } 5039 } 5040 void *param2 = 0; 5041 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5042 mPlaybackThreads.removeItem(output); 5043 } 5044 thread->exit(); 5045 5046 if (thread->type() != ThreadBase::DUPLICATING) { 5047 AudioStreamOut *out = thread->clearOutput(); 5048 assert(out != NULL); 5049 // from now on thread->mOutput is NULL 5050 out->hwDev->close_output_stream(out->hwDev, out->stream); 5051 delete out; 5052 } 5053 return NO_ERROR; 5054} 5055 5056status_t AudioFlinger::suspendOutput(int output) 5057{ 5058 Mutex::Autolock _l(mLock); 5059 PlaybackThread *thread = checkPlaybackThread_l(output); 5060 5061 if (thread == NULL) { 5062 return BAD_VALUE; 5063 } 5064 5065 ALOGV("suspendOutput() %d", output); 5066 thread->suspend(); 5067 5068 return NO_ERROR; 5069} 5070 5071status_t AudioFlinger::restoreOutput(int output) 5072{ 5073 Mutex::Autolock _l(mLock); 5074 PlaybackThread *thread = checkPlaybackThread_l(output); 5075 5076 if (thread == NULL) { 5077 return BAD_VALUE; 5078 } 5079 5080 ALOGV("restoreOutput() %d", output); 5081 5082 thread->restore(); 5083 5084 return NO_ERROR; 5085} 5086 5087int AudioFlinger::openInput(uint32_t *pDevices, 5088 uint32_t *pSamplingRate, 5089 audio_format_t *pFormat, 5090 uint32_t *pChannels, 5091 uint32_t acoustics) 5092{ 5093 status_t status; 5094 RecordThread *thread = NULL; 5095 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5096 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5097 uint32_t channels = pChannels ? *pChannels : 0; 5098 uint32_t reqSamplingRate = samplingRate; 5099 audio_format_t reqFormat = format; 5100 uint32_t reqChannels = channels; 5101 audio_stream_in_t *inStream; 5102 audio_hw_device_t *inHwDev; 5103 5104 if (pDevices == NULL || *pDevices == 0) { 5105 return 0; 5106 } 5107 5108 Mutex::Autolock _l(mLock); 5109 5110 inHwDev = findSuitableHwDev_l(*pDevices); 5111 if (inHwDev == NULL) 5112 return 0; 5113 5114 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5115 &channels, &samplingRate, 5116 (audio_in_acoustics_t)acoustics, 5117 &inStream); 5118 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5119 inStream, 5120 samplingRate, 5121 format, 5122 channels, 5123 acoustics, 5124 status); 5125 5126 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5127 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5128 // or stereo to mono conversions on 16 bit PCM inputs. 5129 if (inStream == NULL && status == BAD_VALUE && 5130 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5131 (samplingRate <= 2 * reqSamplingRate) && 5132 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5133 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5134 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5135 &channels, &samplingRate, 5136 (audio_in_acoustics_t)acoustics, 5137 &inStream); 5138 } 5139 5140 if (inStream != NULL) { 5141 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5142 5143 int id = nextUniqueId(); 5144 // Start record thread 5145 // RecorThread require both input and output device indication to forward to audio 5146 // pre processing modules 5147 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5148 thread = new RecordThread(this, 5149 input, 5150 reqSamplingRate, 5151 reqChannels, 5152 id, 5153 device); 5154 mRecordThreads.add(id, thread); 5155 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5156 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5157 if (pFormat) *pFormat = format; 5158 if (pChannels) *pChannels = reqChannels; 5159 5160 input->stream->common.standby(&input->stream->common); 5161 5162 // notify client processes of the new input creation 5163 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5164 return id; 5165 } 5166 5167 return 0; 5168} 5169 5170status_t AudioFlinger::closeInput(int input) 5171{ 5172 // keep strong reference on the record thread so that 5173 // it is not destroyed while exit() is executed 5174 sp <RecordThread> thread; 5175 { 5176 Mutex::Autolock _l(mLock); 5177 thread = checkRecordThread_l(input); 5178 if (thread == NULL) { 5179 return BAD_VALUE; 5180 } 5181 5182 ALOGV("closeInput() %d", input); 5183 void *param2 = 0; 5184 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5185 mRecordThreads.removeItem(input); 5186 } 5187 thread->exit(); 5188 5189 AudioStreamIn *in = thread->clearInput(); 5190 assert(in != NULL); 5191 // from now on thread->mInput is NULL 5192 in->hwDev->close_input_stream(in->hwDev, in->stream); 5193 delete in; 5194 5195 return NO_ERROR; 5196} 5197 5198status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5199{ 5200 Mutex::Autolock _l(mLock); 5201 MixerThread *dstThread = checkMixerThread_l(output); 5202 if (dstThread == NULL) { 5203 ALOGW("setStreamOutput() bad output id %d", output); 5204 return BAD_VALUE; 5205 } 5206 5207 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5208 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5209 5210 dstThread->setStreamValid(stream, true); 5211 5212 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5213 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5214 if (thread != dstThread && 5215 thread->type() != ThreadBase::DIRECT) { 5216 MixerThread *srcThread = (MixerThread *)thread; 5217 srcThread->setStreamValid(stream, false); 5218 srcThread->invalidateTracks(stream); 5219 } 5220 } 5221 5222 return NO_ERROR; 5223} 5224 5225 5226int AudioFlinger::newAudioSessionId() 5227{ 5228 return nextUniqueId(); 5229} 5230 5231void AudioFlinger::acquireAudioSessionId(int audioSession) 5232{ 5233 Mutex::Autolock _l(mLock); 5234 int caller = IPCThreadState::self()->getCallingPid(); 5235 ALOGV("acquiring %d from %d", audioSession, caller); 5236 int num = mAudioSessionRefs.size(); 5237 for (int i = 0; i< num; i++) { 5238 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5239 if (ref->sessionid == audioSession && ref->pid == caller) { 5240 ref->cnt++; 5241 ALOGV(" incremented refcount to %d", ref->cnt); 5242 return; 5243 } 5244 } 5245 AudioSessionRef *ref = new AudioSessionRef(); 5246 ref->sessionid = audioSession; 5247 ref->pid = caller; 5248 ref->cnt = 1; 5249 mAudioSessionRefs.push(ref); 5250 ALOGV(" added new entry for %d", ref->sessionid); 5251} 5252 5253void AudioFlinger::releaseAudioSessionId(int audioSession) 5254{ 5255 Mutex::Autolock _l(mLock); 5256 int caller = IPCThreadState::self()->getCallingPid(); 5257 ALOGV("releasing %d from %d", audioSession, caller); 5258 int num = mAudioSessionRefs.size(); 5259 for (int i = 0; i< num; i++) { 5260 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5261 if (ref->sessionid == audioSession && ref->pid == caller) { 5262 ref->cnt--; 5263 ALOGV(" decremented refcount to %d", ref->cnt); 5264 if (ref->cnt == 0) { 5265 mAudioSessionRefs.removeAt(i); 5266 delete ref; 5267 purgeStaleEffects_l(); 5268 } 5269 return; 5270 } 5271 } 5272 ALOGW("session id %d not found for pid %d", audioSession, caller); 5273} 5274 5275void AudioFlinger::purgeStaleEffects_l() { 5276 5277 ALOGV("purging stale effects"); 5278 5279 Vector< sp<EffectChain> > chains; 5280 5281 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5282 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5283 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5284 sp<EffectChain> ec = t->mEffectChains[j]; 5285 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5286 chains.push(ec); 5287 } 5288 } 5289 } 5290 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5291 sp<RecordThread> t = mRecordThreads.valueAt(i); 5292 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5293 sp<EffectChain> ec = t->mEffectChains[j]; 5294 chains.push(ec); 5295 } 5296 } 5297 5298 for (size_t i = 0; i < chains.size(); i++) { 5299 sp<EffectChain> ec = chains[i]; 5300 int sessionid = ec->sessionId(); 5301 sp<ThreadBase> t = ec->mThread.promote(); 5302 if (t == 0) { 5303 continue; 5304 } 5305 size_t numsessionrefs = mAudioSessionRefs.size(); 5306 bool found = false; 5307 for (size_t k = 0; k < numsessionrefs; k++) { 5308 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5309 if (ref->sessionid == sessionid) { 5310 ALOGV(" session %d still exists for %d with %d refs", 5311 sessionid, ref->pid, ref->cnt); 5312 found = true; 5313 break; 5314 } 5315 } 5316 if (!found) { 5317 // remove all effects from the chain 5318 while (ec->mEffects.size()) { 5319 sp<EffectModule> effect = ec->mEffects[0]; 5320 effect->unPin(); 5321 Mutex::Autolock _l (t->mLock); 5322 t->removeEffect_l(effect); 5323 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5324 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5325 if (handle != 0) { 5326 handle->mEffect.clear(); 5327 if (handle->mHasControl && handle->mEnabled) { 5328 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5329 } 5330 } 5331 } 5332 AudioSystem::unregisterEffect(effect->id()); 5333 } 5334 } 5335 } 5336 return; 5337} 5338 5339// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5340AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5341{ 5342 PlaybackThread *thread = NULL; 5343 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5344 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5345 } 5346 return thread; 5347} 5348 5349// checkMixerThread_l() must be called with AudioFlinger::mLock held 5350AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5351{ 5352 PlaybackThread *thread = checkPlaybackThread_l(output); 5353 if (thread != NULL) { 5354 if (thread->type() == ThreadBase::DIRECT) { 5355 thread = NULL; 5356 } 5357 } 5358 return (MixerThread *)thread; 5359} 5360 5361// checkRecordThread_l() must be called with AudioFlinger::mLock held 5362AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5363{ 5364 RecordThread *thread = NULL; 5365 if (mRecordThreads.indexOfKey(input) >= 0) { 5366 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5367 } 5368 return thread; 5369} 5370 5371uint32_t AudioFlinger::nextUniqueId() 5372{ 5373 return android_atomic_inc(&mNextUniqueId); 5374} 5375 5376AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5377{ 5378 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5379 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5380 AudioStreamOut *output = thread->getOutput(); 5381 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5382 return thread; 5383 } 5384 } 5385 return NULL; 5386} 5387 5388uint32_t AudioFlinger::primaryOutputDevice_l() 5389{ 5390 PlaybackThread *thread = primaryPlaybackThread_l(); 5391 5392 if (thread == NULL) { 5393 return 0; 5394 } 5395 5396 return thread->device(); 5397} 5398 5399 5400// ---------------------------------------------------------------------------- 5401// Effect management 5402// ---------------------------------------------------------------------------- 5403 5404 5405status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5406{ 5407 Mutex::Autolock _l(mLock); 5408 return EffectQueryNumberEffects(numEffects); 5409} 5410 5411status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5412{ 5413 Mutex::Autolock _l(mLock); 5414 return EffectQueryEffect(index, descriptor); 5415} 5416 5417status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5418{ 5419 Mutex::Autolock _l(mLock); 5420 return EffectGetDescriptor(pUuid, descriptor); 5421} 5422 5423 5424sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5425 effect_descriptor_t *pDesc, 5426 const sp<IEffectClient>& effectClient, 5427 int32_t priority, 5428 int io, 5429 int sessionId, 5430 status_t *status, 5431 int *id, 5432 int *enabled) 5433{ 5434 status_t lStatus = NO_ERROR; 5435 sp<EffectHandle> handle; 5436 effect_descriptor_t desc; 5437 sp<Client> client; 5438 wp<Client> wclient; 5439 5440 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5441 pid, effectClient.get(), priority, sessionId, io); 5442 5443 if (pDesc == NULL) { 5444 lStatus = BAD_VALUE; 5445 goto Exit; 5446 } 5447 5448 // check audio settings permission for global effects 5449 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5450 lStatus = PERMISSION_DENIED; 5451 goto Exit; 5452 } 5453 5454 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5455 // that can only be created by audio policy manager (running in same process) 5456 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5457 lStatus = PERMISSION_DENIED; 5458 goto Exit; 5459 } 5460 5461 if (io == 0) { 5462 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5463 // output must be specified by AudioPolicyManager when using session 5464 // AUDIO_SESSION_OUTPUT_STAGE 5465 lStatus = BAD_VALUE; 5466 goto Exit; 5467 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5468 // if the output returned by getOutputForEffect() is removed before we lock the 5469 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5470 // and we will exit safely 5471 io = AudioSystem::getOutputForEffect(&desc); 5472 } 5473 } 5474 5475 { 5476 Mutex::Autolock _l(mLock); 5477 5478 5479 if (!EffectIsNullUuid(&pDesc->uuid)) { 5480 // if uuid is specified, request effect descriptor 5481 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5482 if (lStatus < 0) { 5483 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5484 goto Exit; 5485 } 5486 } else { 5487 // if uuid is not specified, look for an available implementation 5488 // of the required type in effect factory 5489 if (EffectIsNullUuid(&pDesc->type)) { 5490 ALOGW("createEffect() no effect type"); 5491 lStatus = BAD_VALUE; 5492 goto Exit; 5493 } 5494 uint32_t numEffects = 0; 5495 effect_descriptor_t d; 5496 d.flags = 0; // prevent compiler warning 5497 bool found = false; 5498 5499 lStatus = EffectQueryNumberEffects(&numEffects); 5500 if (lStatus < 0) { 5501 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5502 goto Exit; 5503 } 5504 for (uint32_t i = 0; i < numEffects; i++) { 5505 lStatus = EffectQueryEffect(i, &desc); 5506 if (lStatus < 0) { 5507 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5508 continue; 5509 } 5510 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5511 // If matching type found save effect descriptor. If the session is 5512 // 0 and the effect is not auxiliary, continue enumeration in case 5513 // an auxiliary version of this effect type is available 5514 found = true; 5515 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5516 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5517 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5518 break; 5519 } 5520 } 5521 } 5522 if (!found) { 5523 lStatus = BAD_VALUE; 5524 ALOGW("createEffect() effect not found"); 5525 goto Exit; 5526 } 5527 // For same effect type, chose auxiliary version over insert version if 5528 // connect to output mix (Compliance to OpenSL ES) 5529 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5530 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5531 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5532 } 5533 } 5534 5535 // Do not allow auxiliary effects on a session different from 0 (output mix) 5536 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5537 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5538 lStatus = INVALID_OPERATION; 5539 goto Exit; 5540 } 5541 5542 // check recording permission for visualizer 5543 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5544 !recordingAllowed()) { 5545 lStatus = PERMISSION_DENIED; 5546 goto Exit; 5547 } 5548 5549 // return effect descriptor 5550 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5551 5552 // If output is not specified try to find a matching audio session ID in one of the 5553 // output threads. 5554 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5555 // because of code checking output when entering the function. 5556 // Note: io is never 0 when creating an effect on an input 5557 if (io == 0) { 5558 // look for the thread where the specified audio session is present 5559 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5560 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5561 io = mPlaybackThreads.keyAt(i); 5562 break; 5563 } 5564 } 5565 if (io == 0) { 5566 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5567 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5568 io = mRecordThreads.keyAt(i); 5569 break; 5570 } 5571 } 5572 } 5573 // If no output thread contains the requested session ID, default to 5574 // first output. The effect chain will be moved to the correct output 5575 // thread when a track with the same session ID is created 5576 if (io == 0 && mPlaybackThreads.size()) { 5577 io = mPlaybackThreads.keyAt(0); 5578 } 5579 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5580 } 5581 ThreadBase *thread = checkRecordThread_l(io); 5582 if (thread == NULL) { 5583 thread = checkPlaybackThread_l(io); 5584 if (thread == NULL) { 5585 ALOGE("createEffect() unknown output thread"); 5586 lStatus = BAD_VALUE; 5587 goto Exit; 5588 } 5589 } 5590 5591 wclient = mClients.valueFor(pid); 5592 5593 if (wclient != NULL) { 5594 client = wclient.promote(); 5595 } else { 5596 client = new Client(this, pid); 5597 mClients.add(pid, client); 5598 } 5599 5600 // create effect on selected output thread 5601 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5602 &desc, enabled, &lStatus); 5603 if (handle != 0 && id != NULL) { 5604 *id = handle->id(); 5605 } 5606 } 5607 5608Exit: 5609 if(status) { 5610 *status = lStatus; 5611 } 5612 return handle; 5613} 5614 5615status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5616{ 5617 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5618 sessionId, srcOutput, dstOutput); 5619 Mutex::Autolock _l(mLock); 5620 if (srcOutput == dstOutput) { 5621 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5622 return NO_ERROR; 5623 } 5624 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5625 if (srcThread == NULL) { 5626 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5627 return BAD_VALUE; 5628 } 5629 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5630 if (dstThread == NULL) { 5631 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5632 return BAD_VALUE; 5633 } 5634 5635 Mutex::Autolock _dl(dstThread->mLock); 5636 Mutex::Autolock _sl(srcThread->mLock); 5637 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5638 5639 return NO_ERROR; 5640} 5641 5642// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5643status_t AudioFlinger::moveEffectChain_l(int sessionId, 5644 AudioFlinger::PlaybackThread *srcThread, 5645 AudioFlinger::PlaybackThread *dstThread, 5646 bool reRegister) 5647{ 5648 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5649 sessionId, srcThread, dstThread); 5650 5651 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5652 if (chain == 0) { 5653 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5654 sessionId, srcThread); 5655 return INVALID_OPERATION; 5656 } 5657 5658 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5659 // so that a new chain is created with correct parameters when first effect is added. This is 5660 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5661 // removed. 5662 srcThread->removeEffectChain_l(chain); 5663 5664 // transfer all effects one by one so that new effect chain is created on new thread with 5665 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5666 int dstOutput = dstThread->id(); 5667 sp<EffectChain> dstChain; 5668 uint32_t strategy = 0; // prevent compiler warning 5669 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5670 while (effect != 0) { 5671 srcThread->removeEffect_l(effect); 5672 dstThread->addEffect_l(effect); 5673 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5674 if (effect->state() == EffectModule::ACTIVE || 5675 effect->state() == EffectModule::STOPPING) { 5676 effect->start(); 5677 } 5678 // if the move request is not received from audio policy manager, the effect must be 5679 // re-registered with the new strategy and output 5680 if (dstChain == 0) { 5681 dstChain = effect->chain().promote(); 5682 if (dstChain == 0) { 5683 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5684 srcThread->addEffect_l(effect); 5685 return NO_INIT; 5686 } 5687 strategy = dstChain->strategy(); 5688 } 5689 if (reRegister) { 5690 AudioSystem::unregisterEffect(effect->id()); 5691 AudioSystem::registerEffect(&effect->desc(), 5692 dstOutput, 5693 strategy, 5694 sessionId, 5695 effect->id()); 5696 } 5697 effect = chain->getEffectFromId_l(0); 5698 } 5699 5700 return NO_ERROR; 5701} 5702 5703 5704// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5705sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5706 const sp<AudioFlinger::Client>& client, 5707 const sp<IEffectClient>& effectClient, 5708 int32_t priority, 5709 int sessionId, 5710 effect_descriptor_t *desc, 5711 int *enabled, 5712 status_t *status 5713 ) 5714{ 5715 sp<EffectModule> effect; 5716 sp<EffectHandle> handle; 5717 status_t lStatus; 5718 sp<EffectChain> chain; 5719 bool chainCreated = false; 5720 bool effectCreated = false; 5721 bool effectRegistered = false; 5722 5723 lStatus = initCheck(); 5724 if (lStatus != NO_ERROR) { 5725 ALOGW("createEffect_l() Audio driver not initialized."); 5726 goto Exit; 5727 } 5728 5729 // Do not allow effects with session ID 0 on direct output or duplicating threads 5730 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5731 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5732 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5733 desc->name, sessionId); 5734 lStatus = BAD_VALUE; 5735 goto Exit; 5736 } 5737 // Only Pre processor effects are allowed on input threads and only on input threads 5738 if ((mType == RECORD && 5739 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5740 (mType != RECORD && 5741 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5742 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5743 desc->name, desc->flags, mType); 5744 lStatus = BAD_VALUE; 5745 goto Exit; 5746 } 5747 5748 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5749 5750 { // scope for mLock 5751 Mutex::Autolock _l(mLock); 5752 5753 // check for existing effect chain with the requested audio session 5754 chain = getEffectChain_l(sessionId); 5755 if (chain == 0) { 5756 // create a new chain for this session 5757 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5758 chain = new EffectChain(this, sessionId); 5759 addEffectChain_l(chain); 5760 chain->setStrategy(getStrategyForSession_l(sessionId)); 5761 chainCreated = true; 5762 } else { 5763 effect = chain->getEffectFromDesc_l(desc); 5764 } 5765 5766 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5767 5768 if (effect == 0) { 5769 int id = mAudioFlinger->nextUniqueId(); 5770 // Check CPU and memory usage 5771 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5772 if (lStatus != NO_ERROR) { 5773 goto Exit; 5774 } 5775 effectRegistered = true; 5776 // create a new effect module if none present in the chain 5777 effect = new EffectModule(this, chain, desc, id, sessionId); 5778 lStatus = effect->status(); 5779 if (lStatus != NO_ERROR) { 5780 goto Exit; 5781 } 5782 lStatus = chain->addEffect_l(effect); 5783 if (lStatus != NO_ERROR) { 5784 goto Exit; 5785 } 5786 effectCreated = true; 5787 5788 effect->setDevice(mDevice); 5789 effect->setMode(mAudioFlinger->getMode()); 5790 } 5791 // create effect handle and connect it to effect module 5792 handle = new EffectHandle(effect, client, effectClient, priority); 5793 lStatus = effect->addHandle(handle); 5794 if (enabled) { 5795 *enabled = (int)effect->isEnabled(); 5796 } 5797 } 5798 5799Exit: 5800 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5801 Mutex::Autolock _l(mLock); 5802 if (effectCreated) { 5803 chain->removeEffect_l(effect); 5804 } 5805 if (effectRegistered) { 5806 AudioSystem::unregisterEffect(effect->id()); 5807 } 5808 if (chainCreated) { 5809 removeEffectChain_l(chain); 5810 } 5811 handle.clear(); 5812 } 5813 5814 if(status) { 5815 *status = lStatus; 5816 } 5817 return handle; 5818} 5819 5820sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5821{ 5822 sp<EffectChain> chain = getEffectChain_l(sessionId); 5823 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 5824} 5825 5826// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5827// PlaybackThread::mLock held 5828status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5829{ 5830 // check for existing effect chain with the requested audio session 5831 int sessionId = effect->sessionId(); 5832 sp<EffectChain> chain = getEffectChain_l(sessionId); 5833 bool chainCreated = false; 5834 5835 if (chain == 0) { 5836 // create a new chain for this session 5837 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5838 chain = new EffectChain(this, sessionId); 5839 addEffectChain_l(chain); 5840 chain->setStrategy(getStrategyForSession_l(sessionId)); 5841 chainCreated = true; 5842 } 5843 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5844 5845 if (chain->getEffectFromId_l(effect->id()) != 0) { 5846 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5847 this, effect->desc().name, chain.get()); 5848 return BAD_VALUE; 5849 } 5850 5851 status_t status = chain->addEffect_l(effect); 5852 if (status != NO_ERROR) { 5853 if (chainCreated) { 5854 removeEffectChain_l(chain); 5855 } 5856 return status; 5857 } 5858 5859 effect->setDevice(mDevice); 5860 effect->setMode(mAudioFlinger->getMode()); 5861 return NO_ERROR; 5862} 5863 5864void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5865 5866 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5867 effect_descriptor_t desc = effect->desc(); 5868 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5869 detachAuxEffect_l(effect->id()); 5870 } 5871 5872 sp<EffectChain> chain = effect->chain().promote(); 5873 if (chain != 0) { 5874 // remove effect chain if removing last effect 5875 if (chain->removeEffect_l(effect) == 0) { 5876 removeEffectChain_l(chain); 5877 } 5878 } else { 5879 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5880 } 5881} 5882 5883void AudioFlinger::ThreadBase::lockEffectChains_l( 5884 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5885{ 5886 effectChains = mEffectChains; 5887 for (size_t i = 0; i < mEffectChains.size(); i++) { 5888 mEffectChains[i]->lock(); 5889 } 5890} 5891 5892void AudioFlinger::ThreadBase::unlockEffectChains( 5893 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5894{ 5895 for (size_t i = 0; i < effectChains.size(); i++) { 5896 effectChains[i]->unlock(); 5897 } 5898} 5899 5900sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5901{ 5902 Mutex::Autolock _l(mLock); 5903 return getEffectChain_l(sessionId); 5904} 5905 5906sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5907{ 5908 size_t size = mEffectChains.size(); 5909 for (size_t i = 0; i < size; i++) { 5910 if (mEffectChains[i]->sessionId() == sessionId) { 5911 return mEffectChains[i]; 5912 } 5913 } 5914 return 0; 5915} 5916 5917void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5918{ 5919 Mutex::Autolock _l(mLock); 5920 size_t size = mEffectChains.size(); 5921 for (size_t i = 0; i < size; i++) { 5922 mEffectChains[i]->setMode_l(mode); 5923 } 5924} 5925 5926void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5927 const wp<EffectHandle>& handle, 5928 bool unpiniflast) { 5929 5930 Mutex::Autolock _l(mLock); 5931 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5932 // delete the effect module if removing last handle on it 5933 if (effect->removeHandle(handle) == 0) { 5934 if (!effect->isPinned() || unpiniflast) { 5935 removeEffect_l(effect); 5936 AudioSystem::unregisterEffect(effect->id()); 5937 } 5938 } 5939} 5940 5941status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5942{ 5943 int session = chain->sessionId(); 5944 int16_t *buffer = mMixBuffer; 5945 bool ownsBuffer = false; 5946 5947 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5948 if (session > 0) { 5949 // Only one effect chain can be present in direct output thread and it uses 5950 // the mix buffer as input 5951 if (mType != DIRECT) { 5952 size_t numSamples = mFrameCount * mChannelCount; 5953 buffer = new int16_t[numSamples]; 5954 memset(buffer, 0, numSamples * sizeof(int16_t)); 5955 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5956 ownsBuffer = true; 5957 } 5958 5959 // Attach all tracks with same session ID to this chain. 5960 for (size_t i = 0; i < mTracks.size(); ++i) { 5961 sp<Track> track = mTracks[i]; 5962 if (session == track->sessionId()) { 5963 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5964 track->setMainBuffer(buffer); 5965 chain->incTrackCnt(); 5966 } 5967 } 5968 5969 // indicate all active tracks in the chain 5970 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5971 sp<Track> track = mActiveTracks[i].promote(); 5972 if (track == 0) continue; 5973 if (session == track->sessionId()) { 5974 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5975 chain->incActiveTrackCnt(); 5976 } 5977 } 5978 } 5979 5980 chain->setInBuffer(buffer, ownsBuffer); 5981 chain->setOutBuffer(mMixBuffer); 5982 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5983 // chains list in order to be processed last as it contains output stage effects 5984 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5985 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5986 // after track specific effects and before output stage 5987 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5988 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5989 // Effect chain for other sessions are inserted at beginning of effect 5990 // chains list to be processed before output mix effects. Relative order between other 5991 // sessions is not important 5992 size_t size = mEffectChains.size(); 5993 size_t i = 0; 5994 for (i = 0; i < size; i++) { 5995 if (mEffectChains[i]->sessionId() < session) break; 5996 } 5997 mEffectChains.insertAt(chain, i); 5998 checkSuspendOnAddEffectChain_l(chain); 5999 6000 return NO_ERROR; 6001} 6002 6003size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6004{ 6005 int session = chain->sessionId(); 6006 6007 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6008 6009 for (size_t i = 0; i < mEffectChains.size(); i++) { 6010 if (chain == mEffectChains[i]) { 6011 mEffectChains.removeAt(i); 6012 // detach all active tracks from the chain 6013 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6014 sp<Track> track = mActiveTracks[i].promote(); 6015 if (track == 0) continue; 6016 if (session == track->sessionId()) { 6017 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6018 chain.get(), session); 6019 chain->decActiveTrackCnt(); 6020 } 6021 } 6022 6023 // detach all tracks with same session ID from this chain 6024 for (size_t i = 0; i < mTracks.size(); ++i) { 6025 sp<Track> track = mTracks[i]; 6026 if (session == track->sessionId()) { 6027 track->setMainBuffer(mMixBuffer); 6028 chain->decTrackCnt(); 6029 } 6030 } 6031 break; 6032 } 6033 } 6034 return mEffectChains.size(); 6035} 6036 6037status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6038 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6039{ 6040 Mutex::Autolock _l(mLock); 6041 return attachAuxEffect_l(track, EffectId); 6042} 6043 6044status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6045 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6046{ 6047 status_t status = NO_ERROR; 6048 6049 if (EffectId == 0) { 6050 track->setAuxBuffer(0, NULL); 6051 } else { 6052 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6053 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6054 if (effect != 0) { 6055 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6056 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6057 } else { 6058 status = INVALID_OPERATION; 6059 } 6060 } else { 6061 status = BAD_VALUE; 6062 } 6063 } 6064 return status; 6065} 6066 6067void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6068{ 6069 for (size_t i = 0; i < mTracks.size(); ++i) { 6070 sp<Track> track = mTracks[i]; 6071 if (track->auxEffectId() == effectId) { 6072 attachAuxEffect_l(track, 0); 6073 } 6074 } 6075} 6076 6077status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6078{ 6079 // only one chain per input thread 6080 if (mEffectChains.size() != 0) { 6081 return INVALID_OPERATION; 6082 } 6083 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6084 6085 chain->setInBuffer(NULL); 6086 chain->setOutBuffer(NULL); 6087 6088 checkSuspendOnAddEffectChain_l(chain); 6089 6090 mEffectChains.add(chain); 6091 6092 return NO_ERROR; 6093} 6094 6095size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6096{ 6097 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6098 ALOGW_IF(mEffectChains.size() != 1, 6099 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6100 chain.get(), mEffectChains.size(), this); 6101 if (mEffectChains.size() == 1) { 6102 mEffectChains.removeAt(0); 6103 } 6104 return 0; 6105} 6106 6107// ---------------------------------------------------------------------------- 6108// EffectModule implementation 6109// ---------------------------------------------------------------------------- 6110 6111#undef LOG_TAG 6112#define LOG_TAG "AudioFlinger::EffectModule" 6113 6114AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6115 const wp<AudioFlinger::EffectChain>& chain, 6116 effect_descriptor_t *desc, 6117 int id, 6118 int sessionId) 6119 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6120 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6121{ 6122 ALOGV("Constructor %p", this); 6123 int lStatus; 6124 sp<ThreadBase> thread = mThread.promote(); 6125 if (thread == 0) { 6126 return; 6127 } 6128 6129 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6130 6131 // create effect engine from effect factory 6132 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6133 6134 if (mStatus != NO_ERROR) { 6135 return; 6136 } 6137 lStatus = init(); 6138 if (lStatus < 0) { 6139 mStatus = lStatus; 6140 goto Error; 6141 } 6142 6143 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6144 mPinned = true; 6145 } 6146 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6147 return; 6148Error: 6149 EffectRelease(mEffectInterface); 6150 mEffectInterface = NULL; 6151 ALOGV("Constructor Error %d", mStatus); 6152} 6153 6154AudioFlinger::EffectModule::~EffectModule() 6155{ 6156 ALOGV("Destructor %p", this); 6157 if (mEffectInterface != NULL) { 6158 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6159 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6160 sp<ThreadBase> thread = mThread.promote(); 6161 if (thread != 0) { 6162 audio_stream_t *stream = thread->stream(); 6163 if (stream != NULL) { 6164 stream->remove_audio_effect(stream, mEffectInterface); 6165 } 6166 } 6167 } 6168 // release effect engine 6169 EffectRelease(mEffectInterface); 6170 } 6171} 6172 6173status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6174{ 6175 status_t status; 6176 6177 Mutex::Autolock _l(mLock); 6178 // First handle in mHandles has highest priority and controls the effect module 6179 int priority = handle->priority(); 6180 size_t size = mHandles.size(); 6181 sp<EffectHandle> h; 6182 size_t i; 6183 for (i = 0; i < size; i++) { 6184 h = mHandles[i].promote(); 6185 if (h == 0) continue; 6186 if (h->priority() <= priority) break; 6187 } 6188 // if inserted in first place, move effect control from previous owner to this handle 6189 if (i == 0) { 6190 bool enabled = false; 6191 if (h != 0) { 6192 enabled = h->enabled(); 6193 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6194 } 6195 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6196 status = NO_ERROR; 6197 } else { 6198 status = ALREADY_EXISTS; 6199 } 6200 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6201 mHandles.insertAt(handle, i); 6202 return status; 6203} 6204 6205size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6206{ 6207 Mutex::Autolock _l(mLock); 6208 size_t size = mHandles.size(); 6209 size_t i; 6210 for (i = 0; i < size; i++) { 6211 if (mHandles[i] == handle) break; 6212 } 6213 if (i == size) { 6214 return size; 6215 } 6216 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6217 6218 bool enabled = false; 6219 EffectHandle *hdl = handle.unsafe_get(); 6220 if (hdl) { 6221 ALOGV("removeHandle() unsafe_get OK"); 6222 enabled = hdl->enabled(); 6223 } 6224 mHandles.removeAt(i); 6225 size = mHandles.size(); 6226 // if removed from first place, move effect control from this handle to next in line 6227 if (i == 0 && size != 0) { 6228 sp<EffectHandle> h = mHandles[0].promote(); 6229 if (h != 0) { 6230 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6231 } 6232 } 6233 6234 // Prevent calls to process() and other functions on effect interface from now on. 6235 // The effect engine will be released by the destructor when the last strong reference on 6236 // this object is released which can happen after next process is called. 6237 if (size == 0 && !mPinned) { 6238 mState = DESTROYED; 6239 } 6240 6241 return size; 6242} 6243 6244sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6245{ 6246 Mutex::Autolock _l(mLock); 6247 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6248} 6249 6250void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6251{ 6252 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6253 // keep a strong reference on this EffectModule to avoid calling the 6254 // destructor before we exit 6255 sp<EffectModule> keep(this); 6256 { 6257 sp<ThreadBase> thread = mThread.promote(); 6258 if (thread != 0) { 6259 thread->disconnectEffect(keep, handle, unpiniflast); 6260 } 6261 } 6262} 6263 6264void AudioFlinger::EffectModule::updateState() { 6265 Mutex::Autolock _l(mLock); 6266 6267 switch (mState) { 6268 case RESTART: 6269 reset_l(); 6270 // FALL THROUGH 6271 6272 case STARTING: 6273 // clear auxiliary effect input buffer for next accumulation 6274 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6275 memset(mConfig.inputCfg.buffer.raw, 6276 0, 6277 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6278 } 6279 start_l(); 6280 mState = ACTIVE; 6281 break; 6282 case STOPPING: 6283 stop_l(); 6284 mDisableWaitCnt = mMaxDisableWaitCnt; 6285 mState = STOPPED; 6286 break; 6287 case STOPPED: 6288 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6289 // turn off sequence. 6290 if (--mDisableWaitCnt == 0) { 6291 reset_l(); 6292 mState = IDLE; 6293 } 6294 break; 6295 default: //IDLE , ACTIVE, DESTROYED 6296 break; 6297 } 6298} 6299 6300void AudioFlinger::EffectModule::process() 6301{ 6302 Mutex::Autolock _l(mLock); 6303 6304 if (mState == DESTROYED || mEffectInterface == NULL || 6305 mConfig.inputCfg.buffer.raw == NULL || 6306 mConfig.outputCfg.buffer.raw == NULL) { 6307 return; 6308 } 6309 6310 if (isProcessEnabled()) { 6311 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6312 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6313 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6314 mConfig.inputCfg.buffer.s32, 6315 mConfig.inputCfg.buffer.frameCount/2); 6316 } 6317 6318 // do the actual processing in the effect engine 6319 int ret = (*mEffectInterface)->process(mEffectInterface, 6320 &mConfig.inputCfg.buffer, 6321 &mConfig.outputCfg.buffer); 6322 6323 // force transition to IDLE state when engine is ready 6324 if (mState == STOPPED && ret == -ENODATA) { 6325 mDisableWaitCnt = 1; 6326 } 6327 6328 // clear auxiliary effect input buffer for next accumulation 6329 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6330 memset(mConfig.inputCfg.buffer.raw, 0, 6331 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6332 } 6333 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6334 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6335 // If an insert effect is idle and input buffer is different from output buffer, 6336 // accumulate input onto output 6337 sp<EffectChain> chain = mChain.promote(); 6338 if (chain != 0 && chain->activeTrackCnt() != 0) { 6339 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6340 int16_t *in = mConfig.inputCfg.buffer.s16; 6341 int16_t *out = mConfig.outputCfg.buffer.s16; 6342 for (size_t i = 0; i < frameCnt; i++) { 6343 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6344 } 6345 } 6346 } 6347} 6348 6349void AudioFlinger::EffectModule::reset_l() 6350{ 6351 if (mEffectInterface == NULL) { 6352 return; 6353 } 6354 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6355} 6356 6357status_t AudioFlinger::EffectModule::configure() 6358{ 6359 uint32_t channels; 6360 if (mEffectInterface == NULL) { 6361 return NO_INIT; 6362 } 6363 6364 sp<ThreadBase> thread = mThread.promote(); 6365 if (thread == 0) { 6366 return DEAD_OBJECT; 6367 } 6368 6369 // TODO: handle configuration of effects replacing track process 6370 if (thread->channelCount() == 1) { 6371 channels = AUDIO_CHANNEL_OUT_MONO; 6372 } else { 6373 channels = AUDIO_CHANNEL_OUT_STEREO; 6374 } 6375 6376 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6377 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6378 } else { 6379 mConfig.inputCfg.channels = channels; 6380 } 6381 mConfig.outputCfg.channels = channels; 6382 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6383 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6384 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6385 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6386 mConfig.inputCfg.bufferProvider.cookie = NULL; 6387 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6388 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6389 mConfig.outputCfg.bufferProvider.cookie = NULL; 6390 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6391 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6392 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6393 // Insert effect: 6394 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6395 // always overwrites output buffer: input buffer == output buffer 6396 // - in other sessions: 6397 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6398 // other effect: overwrites output buffer: input buffer == output buffer 6399 // Auxiliary effect: 6400 // accumulates in output buffer: input buffer != output buffer 6401 // Therefore: accumulate <=> input buffer != output buffer 6402 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6403 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6404 } else { 6405 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6406 } 6407 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6408 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6409 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6410 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6411 6412 ALOGV("configure() %p thread %p buffer %p framecount %d", 6413 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6414 6415 status_t cmdStatus; 6416 uint32_t size = sizeof(int); 6417 status_t status = (*mEffectInterface)->command(mEffectInterface, 6418 EFFECT_CMD_SET_CONFIG, 6419 sizeof(effect_config_t), 6420 &mConfig, 6421 &size, 6422 &cmdStatus); 6423 if (status == 0) { 6424 status = cmdStatus; 6425 } 6426 6427 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6428 (1000 * mConfig.outputCfg.buffer.frameCount); 6429 6430 return status; 6431} 6432 6433status_t AudioFlinger::EffectModule::init() 6434{ 6435 Mutex::Autolock _l(mLock); 6436 if (mEffectInterface == NULL) { 6437 return NO_INIT; 6438 } 6439 status_t cmdStatus; 6440 uint32_t size = sizeof(status_t); 6441 status_t status = (*mEffectInterface)->command(mEffectInterface, 6442 EFFECT_CMD_INIT, 6443 0, 6444 NULL, 6445 &size, 6446 &cmdStatus); 6447 if (status == 0) { 6448 status = cmdStatus; 6449 } 6450 return status; 6451} 6452 6453status_t AudioFlinger::EffectModule::start() 6454{ 6455 Mutex::Autolock _l(mLock); 6456 return start_l(); 6457} 6458 6459status_t AudioFlinger::EffectModule::start_l() 6460{ 6461 if (mEffectInterface == NULL) { 6462 return NO_INIT; 6463 } 6464 status_t cmdStatus; 6465 uint32_t size = sizeof(status_t); 6466 status_t status = (*mEffectInterface)->command(mEffectInterface, 6467 EFFECT_CMD_ENABLE, 6468 0, 6469 NULL, 6470 &size, 6471 &cmdStatus); 6472 if (status == 0) { 6473 status = cmdStatus; 6474 } 6475 if (status == 0 && 6476 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6477 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6478 sp<ThreadBase> thread = mThread.promote(); 6479 if (thread != 0) { 6480 audio_stream_t *stream = thread->stream(); 6481 if (stream != NULL) { 6482 stream->add_audio_effect(stream, mEffectInterface); 6483 } 6484 } 6485 } 6486 return status; 6487} 6488 6489status_t AudioFlinger::EffectModule::stop() 6490{ 6491 Mutex::Autolock _l(mLock); 6492 return stop_l(); 6493} 6494 6495status_t AudioFlinger::EffectModule::stop_l() 6496{ 6497 if (mEffectInterface == NULL) { 6498 return NO_INIT; 6499 } 6500 status_t cmdStatus; 6501 uint32_t size = sizeof(status_t); 6502 status_t status = (*mEffectInterface)->command(mEffectInterface, 6503 EFFECT_CMD_DISABLE, 6504 0, 6505 NULL, 6506 &size, 6507 &cmdStatus); 6508 if (status == 0) { 6509 status = cmdStatus; 6510 } 6511 if (status == 0 && 6512 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6513 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6514 sp<ThreadBase> thread = mThread.promote(); 6515 if (thread != 0) { 6516 audio_stream_t *stream = thread->stream(); 6517 if (stream != NULL) { 6518 stream->remove_audio_effect(stream, mEffectInterface); 6519 } 6520 } 6521 } 6522 return status; 6523} 6524 6525status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6526 uint32_t cmdSize, 6527 void *pCmdData, 6528 uint32_t *replySize, 6529 void *pReplyData) 6530{ 6531 Mutex::Autolock _l(mLock); 6532// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6533 6534 if (mState == DESTROYED || mEffectInterface == NULL) { 6535 return NO_INIT; 6536 } 6537 status_t status = (*mEffectInterface)->command(mEffectInterface, 6538 cmdCode, 6539 cmdSize, 6540 pCmdData, 6541 replySize, 6542 pReplyData); 6543 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6544 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6545 for (size_t i = 1; i < mHandles.size(); i++) { 6546 sp<EffectHandle> h = mHandles[i].promote(); 6547 if (h != 0) { 6548 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6549 } 6550 } 6551 } 6552 return status; 6553} 6554 6555status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6556{ 6557 6558 Mutex::Autolock _l(mLock); 6559 ALOGV("setEnabled %p enabled %d", this, enabled); 6560 6561 if (enabled != isEnabled()) { 6562 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6563 if (enabled && status != NO_ERROR) { 6564 return status; 6565 } 6566 6567 switch (mState) { 6568 // going from disabled to enabled 6569 case IDLE: 6570 mState = STARTING; 6571 break; 6572 case STOPPED: 6573 mState = RESTART; 6574 break; 6575 case STOPPING: 6576 mState = ACTIVE; 6577 break; 6578 6579 // going from enabled to disabled 6580 case RESTART: 6581 mState = STOPPED; 6582 break; 6583 case STARTING: 6584 mState = IDLE; 6585 break; 6586 case ACTIVE: 6587 mState = STOPPING; 6588 break; 6589 case DESTROYED: 6590 return NO_ERROR; // simply ignore as we are being destroyed 6591 } 6592 for (size_t i = 1; i < mHandles.size(); i++) { 6593 sp<EffectHandle> h = mHandles[i].promote(); 6594 if (h != 0) { 6595 h->setEnabled(enabled); 6596 } 6597 } 6598 } 6599 return NO_ERROR; 6600} 6601 6602bool AudioFlinger::EffectModule::isEnabled() 6603{ 6604 switch (mState) { 6605 case RESTART: 6606 case STARTING: 6607 case ACTIVE: 6608 return true; 6609 case IDLE: 6610 case STOPPING: 6611 case STOPPED: 6612 case DESTROYED: 6613 default: 6614 return false; 6615 } 6616} 6617 6618bool AudioFlinger::EffectModule::isProcessEnabled() 6619{ 6620 switch (mState) { 6621 case RESTART: 6622 case ACTIVE: 6623 case STOPPING: 6624 case STOPPED: 6625 return true; 6626 case IDLE: 6627 case STARTING: 6628 case DESTROYED: 6629 default: 6630 return false; 6631 } 6632} 6633 6634status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6635{ 6636 Mutex::Autolock _l(mLock); 6637 status_t status = NO_ERROR; 6638 6639 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6640 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6641 if (isProcessEnabled() && 6642 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6643 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6644 status_t cmdStatus; 6645 uint32_t volume[2]; 6646 uint32_t *pVolume = NULL; 6647 uint32_t size = sizeof(volume); 6648 volume[0] = *left; 6649 volume[1] = *right; 6650 if (controller) { 6651 pVolume = volume; 6652 } 6653 status = (*mEffectInterface)->command(mEffectInterface, 6654 EFFECT_CMD_SET_VOLUME, 6655 size, 6656 volume, 6657 &size, 6658 pVolume); 6659 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6660 *left = volume[0]; 6661 *right = volume[1]; 6662 } 6663 } 6664 return status; 6665} 6666 6667status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6668{ 6669 Mutex::Autolock _l(mLock); 6670 status_t status = NO_ERROR; 6671 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6672 // audio pre processing modules on RecordThread can receive both output and 6673 // input device indication in the same call 6674 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6675 if (dev) { 6676 status_t cmdStatus; 6677 uint32_t size = sizeof(status_t); 6678 6679 status = (*mEffectInterface)->command(mEffectInterface, 6680 EFFECT_CMD_SET_DEVICE, 6681 sizeof(uint32_t), 6682 &dev, 6683 &size, 6684 &cmdStatus); 6685 if (status == NO_ERROR) { 6686 status = cmdStatus; 6687 } 6688 } 6689 dev = device & AUDIO_DEVICE_IN_ALL; 6690 if (dev) { 6691 status_t cmdStatus; 6692 uint32_t size = sizeof(status_t); 6693 6694 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6695 EFFECT_CMD_SET_INPUT_DEVICE, 6696 sizeof(uint32_t), 6697 &dev, 6698 &size, 6699 &cmdStatus); 6700 if (status2 == NO_ERROR) { 6701 status2 = cmdStatus; 6702 } 6703 if (status == NO_ERROR) { 6704 status = status2; 6705 } 6706 } 6707 } 6708 return status; 6709} 6710 6711status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6712{ 6713 Mutex::Autolock _l(mLock); 6714 status_t status = NO_ERROR; 6715 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6716 status_t cmdStatus; 6717 uint32_t size = sizeof(status_t); 6718 status = (*mEffectInterface)->command(mEffectInterface, 6719 EFFECT_CMD_SET_AUDIO_MODE, 6720 sizeof(audio_mode_t), 6721 &mode, 6722 &size, 6723 &cmdStatus); 6724 if (status == NO_ERROR) { 6725 status = cmdStatus; 6726 } 6727 } 6728 return status; 6729} 6730 6731void AudioFlinger::EffectModule::setSuspended(bool suspended) 6732{ 6733 Mutex::Autolock _l(mLock); 6734 mSuspended = suspended; 6735} 6736 6737bool AudioFlinger::EffectModule::suspended() const 6738{ 6739 Mutex::Autolock _l(mLock); 6740 return mSuspended; 6741} 6742 6743status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6744{ 6745 const size_t SIZE = 256; 6746 char buffer[SIZE]; 6747 String8 result; 6748 6749 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6750 result.append(buffer); 6751 6752 bool locked = tryLock(mLock); 6753 // failed to lock - AudioFlinger is probably deadlocked 6754 if (!locked) { 6755 result.append("\t\tCould not lock Fx mutex:\n"); 6756 } 6757 6758 result.append("\t\tSession Status State Engine:\n"); 6759 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6760 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6761 result.append(buffer); 6762 6763 result.append("\t\tDescriptor:\n"); 6764 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6765 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6766 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6767 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6768 result.append(buffer); 6769 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6770 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6771 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6772 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6773 result.append(buffer); 6774 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6775 mDescriptor.apiVersion, 6776 mDescriptor.flags); 6777 result.append(buffer); 6778 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6779 mDescriptor.name); 6780 result.append(buffer); 6781 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6782 mDescriptor.implementor); 6783 result.append(buffer); 6784 6785 result.append("\t\t- Input configuration:\n"); 6786 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6787 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6788 (uint32_t)mConfig.inputCfg.buffer.raw, 6789 mConfig.inputCfg.buffer.frameCount, 6790 mConfig.inputCfg.samplingRate, 6791 mConfig.inputCfg.channels, 6792 mConfig.inputCfg.format); 6793 result.append(buffer); 6794 6795 result.append("\t\t- Output configuration:\n"); 6796 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6797 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6798 (uint32_t)mConfig.outputCfg.buffer.raw, 6799 mConfig.outputCfg.buffer.frameCount, 6800 mConfig.outputCfg.samplingRate, 6801 mConfig.outputCfg.channels, 6802 mConfig.outputCfg.format); 6803 result.append(buffer); 6804 6805 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6806 result.append(buffer); 6807 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6808 for (size_t i = 0; i < mHandles.size(); ++i) { 6809 sp<EffectHandle> handle = mHandles[i].promote(); 6810 if (handle != 0) { 6811 handle->dump(buffer, SIZE); 6812 result.append(buffer); 6813 } 6814 } 6815 6816 result.append("\n"); 6817 6818 write(fd, result.string(), result.length()); 6819 6820 if (locked) { 6821 mLock.unlock(); 6822 } 6823 6824 return NO_ERROR; 6825} 6826 6827// ---------------------------------------------------------------------------- 6828// EffectHandle implementation 6829// ---------------------------------------------------------------------------- 6830 6831#undef LOG_TAG 6832#define LOG_TAG "AudioFlinger::EffectHandle" 6833 6834AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6835 const sp<AudioFlinger::Client>& client, 6836 const sp<IEffectClient>& effectClient, 6837 int32_t priority) 6838 : BnEffect(), 6839 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6840 mPriority(priority), mHasControl(false), mEnabled(false) 6841{ 6842 ALOGV("constructor %p", this); 6843 6844 if (client == 0) { 6845 return; 6846 } 6847 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6848 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6849 if (mCblkMemory != 0) { 6850 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6851 6852 if (mCblk) { 6853 new(mCblk) effect_param_cblk_t(); 6854 mBuffer = (uint8_t *)mCblk + bufOffset; 6855 } 6856 } else { 6857 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6858 return; 6859 } 6860} 6861 6862AudioFlinger::EffectHandle::~EffectHandle() 6863{ 6864 ALOGV("Destructor %p", this); 6865 disconnect(false); 6866 ALOGV("Destructor DONE %p", this); 6867} 6868 6869status_t AudioFlinger::EffectHandle::enable() 6870{ 6871 ALOGV("enable %p", this); 6872 if (!mHasControl) return INVALID_OPERATION; 6873 if (mEffect == 0) return DEAD_OBJECT; 6874 6875 if (mEnabled) { 6876 return NO_ERROR; 6877 } 6878 6879 mEnabled = true; 6880 6881 sp<ThreadBase> thread = mEffect->thread().promote(); 6882 if (thread != 0) { 6883 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6884 } 6885 6886 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6887 if (mEffect->suspended()) { 6888 return NO_ERROR; 6889 } 6890 6891 status_t status = mEffect->setEnabled(true); 6892 if (status != NO_ERROR) { 6893 if (thread != 0) { 6894 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6895 } 6896 mEnabled = false; 6897 } 6898 return status; 6899} 6900 6901status_t AudioFlinger::EffectHandle::disable() 6902{ 6903 ALOGV("disable %p", this); 6904 if (!mHasControl) return INVALID_OPERATION; 6905 if (mEffect == 0) return DEAD_OBJECT; 6906 6907 if (!mEnabled) { 6908 return NO_ERROR; 6909 } 6910 mEnabled = false; 6911 6912 if (mEffect->suspended()) { 6913 return NO_ERROR; 6914 } 6915 6916 status_t status = mEffect->setEnabled(false); 6917 6918 sp<ThreadBase> thread = mEffect->thread().promote(); 6919 if (thread != 0) { 6920 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6921 } 6922 6923 return status; 6924} 6925 6926void AudioFlinger::EffectHandle::disconnect() 6927{ 6928 disconnect(true); 6929} 6930 6931void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6932{ 6933 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6934 if (mEffect == 0) { 6935 return; 6936 } 6937 mEffect->disconnect(this, unpiniflast); 6938 6939 if (mHasControl && mEnabled) { 6940 sp<ThreadBase> thread = mEffect->thread().promote(); 6941 if (thread != 0) { 6942 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6943 } 6944 } 6945 6946 // release sp on module => module destructor can be called now 6947 mEffect.clear(); 6948 if (mClient != 0) { 6949 if (mCblk) { 6950 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6951 } 6952 mCblkMemory.clear(); // and free the shared memory 6953 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6954 mClient.clear(); 6955 } 6956} 6957 6958status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6959 uint32_t cmdSize, 6960 void *pCmdData, 6961 uint32_t *replySize, 6962 void *pReplyData) 6963{ 6964// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6965// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6966 6967 // only get parameter command is permitted for applications not controlling the effect 6968 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6969 return INVALID_OPERATION; 6970 } 6971 if (mEffect == 0) return DEAD_OBJECT; 6972 if (mClient == 0) return INVALID_OPERATION; 6973 6974 // handle commands that are not forwarded transparently to effect engine 6975 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6976 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6977 // no risk to block the whole media server process or mixer threads is we are stuck here 6978 Mutex::Autolock _l(mCblk->lock); 6979 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6980 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6981 mCblk->serverIndex = 0; 6982 mCblk->clientIndex = 0; 6983 return BAD_VALUE; 6984 } 6985 status_t status = NO_ERROR; 6986 while (mCblk->serverIndex < mCblk->clientIndex) { 6987 int reply; 6988 uint32_t rsize = sizeof(int); 6989 int *p = (int *)(mBuffer + mCblk->serverIndex); 6990 int size = *p++; 6991 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6992 ALOGW("command(): invalid parameter block size"); 6993 break; 6994 } 6995 effect_param_t *param = (effect_param_t *)p; 6996 if (param->psize == 0 || param->vsize == 0) { 6997 ALOGW("command(): null parameter or value size"); 6998 mCblk->serverIndex += size; 6999 continue; 7000 } 7001 uint32_t psize = sizeof(effect_param_t) + 7002 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7003 param->vsize; 7004 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7005 psize, 7006 p, 7007 &rsize, 7008 &reply); 7009 // stop at first error encountered 7010 if (ret != NO_ERROR) { 7011 status = ret; 7012 *(int *)pReplyData = reply; 7013 break; 7014 } else if (reply != NO_ERROR) { 7015 *(int *)pReplyData = reply; 7016 break; 7017 } 7018 mCblk->serverIndex += size; 7019 } 7020 mCblk->serverIndex = 0; 7021 mCblk->clientIndex = 0; 7022 return status; 7023 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7024 *(int *)pReplyData = NO_ERROR; 7025 return enable(); 7026 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7027 *(int *)pReplyData = NO_ERROR; 7028 return disable(); 7029 } 7030 7031 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7032} 7033 7034sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7035 return mCblkMemory; 7036} 7037 7038void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7039{ 7040 ALOGV("setControl %p control %d", this, hasControl); 7041 7042 mHasControl = hasControl; 7043 mEnabled = enabled; 7044 7045 if (signal && mEffectClient != 0) { 7046 mEffectClient->controlStatusChanged(hasControl); 7047 } 7048} 7049 7050void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7051 uint32_t cmdSize, 7052 void *pCmdData, 7053 uint32_t replySize, 7054 void *pReplyData) 7055{ 7056 if (mEffectClient != 0) { 7057 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7058 } 7059} 7060 7061 7062 7063void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7064{ 7065 if (mEffectClient != 0) { 7066 mEffectClient->enableStatusChanged(enabled); 7067 } 7068} 7069 7070status_t AudioFlinger::EffectHandle::onTransact( 7071 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7072{ 7073 return BnEffect::onTransact(code, data, reply, flags); 7074} 7075 7076 7077void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7078{ 7079 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7080 7081 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7082 (mClient == NULL) ? getpid() : mClient->pid(), 7083 mPriority, 7084 mHasControl, 7085 !locked, 7086 mCblk ? mCblk->clientIndex : 0, 7087 mCblk ? mCblk->serverIndex : 0 7088 ); 7089 7090 if (locked) { 7091 mCblk->lock.unlock(); 7092 } 7093} 7094 7095#undef LOG_TAG 7096#define LOG_TAG "AudioFlinger::EffectChain" 7097 7098AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7099 int sessionId) 7100 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7101 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7102 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7103{ 7104 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7105 sp<ThreadBase> thread = mThread.promote(); 7106 if (thread == 0) { 7107 return; 7108 } 7109 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7110 thread->frameCount(); 7111} 7112 7113AudioFlinger::EffectChain::~EffectChain() 7114{ 7115 if (mOwnInBuffer) { 7116 delete mInBuffer; 7117 } 7118 7119} 7120 7121// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7122sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7123{ 7124 size_t size = mEffects.size(); 7125 7126 for (size_t i = 0; i < size; i++) { 7127 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7128 return mEffects[i]; 7129 } 7130 } 7131 return 0; 7132} 7133 7134// getEffectFromId_l() must be called with ThreadBase::mLock held 7135sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7136{ 7137 size_t size = mEffects.size(); 7138 7139 for (size_t i = 0; i < size; i++) { 7140 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7141 if (id == 0 || mEffects[i]->id() == id) { 7142 return mEffects[i]; 7143 } 7144 } 7145 return 0; 7146} 7147 7148// getEffectFromType_l() must be called with ThreadBase::mLock held 7149sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7150 const effect_uuid_t *type) 7151{ 7152 size_t size = mEffects.size(); 7153 7154 for (size_t i = 0; i < size; i++) { 7155 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7156 return mEffects[i]; 7157 } 7158 } 7159 return 0; 7160} 7161 7162// Must be called with EffectChain::mLock locked 7163void AudioFlinger::EffectChain::process_l() 7164{ 7165 sp<ThreadBase> thread = mThread.promote(); 7166 if (thread == 0) { 7167 ALOGW("process_l(): cannot promote mixer thread"); 7168 return; 7169 } 7170 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7171 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7172 // always process effects unless no more tracks are on the session and the effect tail 7173 // has been rendered 7174 bool doProcess = true; 7175 if (!isGlobalSession) { 7176 bool tracksOnSession = (trackCnt() != 0); 7177 7178 if (!tracksOnSession && mTailBufferCount == 0) { 7179 doProcess = false; 7180 } 7181 7182 if (activeTrackCnt() == 0) { 7183 // if no track is active and the effect tail has not been rendered, 7184 // the input buffer must be cleared here as the mixer process will not do it 7185 if (tracksOnSession || mTailBufferCount > 0) { 7186 size_t numSamples = thread->frameCount() * thread->channelCount(); 7187 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7188 if (mTailBufferCount > 0) { 7189 mTailBufferCount--; 7190 } 7191 } 7192 } 7193 } 7194 7195 size_t size = mEffects.size(); 7196 if (doProcess) { 7197 for (size_t i = 0; i < size; i++) { 7198 mEffects[i]->process(); 7199 } 7200 } 7201 for (size_t i = 0; i < size; i++) { 7202 mEffects[i]->updateState(); 7203 } 7204} 7205 7206// addEffect_l() must be called with PlaybackThread::mLock held 7207status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7208{ 7209 effect_descriptor_t desc = effect->desc(); 7210 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7211 7212 Mutex::Autolock _l(mLock); 7213 effect->setChain(this); 7214 sp<ThreadBase> thread = mThread.promote(); 7215 if (thread == 0) { 7216 return NO_INIT; 7217 } 7218 effect->setThread(thread); 7219 7220 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7221 // Auxiliary effects are inserted at the beginning of mEffects vector as 7222 // they are processed first and accumulated in chain input buffer 7223 mEffects.insertAt(effect, 0); 7224 7225 // the input buffer for auxiliary effect contains mono samples in 7226 // 32 bit format. This is to avoid saturation in AudoMixer 7227 // accumulation stage. Saturation is done in EffectModule::process() before 7228 // calling the process in effect engine 7229 size_t numSamples = thread->frameCount(); 7230 int32_t *buffer = new int32_t[numSamples]; 7231 memset(buffer, 0, numSamples * sizeof(int32_t)); 7232 effect->setInBuffer((int16_t *)buffer); 7233 // auxiliary effects output samples to chain input buffer for further processing 7234 // by insert effects 7235 effect->setOutBuffer(mInBuffer); 7236 } else { 7237 // Insert effects are inserted at the end of mEffects vector as they are processed 7238 // after track and auxiliary effects. 7239 // Insert effect order as a function of indicated preference: 7240 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7241 // another effect is present 7242 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7243 // last effect claiming first position 7244 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7245 // first effect claiming last position 7246 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7247 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7248 // already present 7249 7250 int size = (int)mEffects.size(); 7251 int idx_insert = size; 7252 int idx_insert_first = -1; 7253 int idx_insert_last = -1; 7254 7255 for (int i = 0; i < size; i++) { 7256 effect_descriptor_t d = mEffects[i]->desc(); 7257 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7258 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7259 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7260 // check invalid effect chaining combinations 7261 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7262 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7263 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7264 return INVALID_OPERATION; 7265 } 7266 // remember position of first insert effect and by default 7267 // select this as insert position for new effect 7268 if (idx_insert == size) { 7269 idx_insert = i; 7270 } 7271 // remember position of last insert effect claiming 7272 // first position 7273 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7274 idx_insert_first = i; 7275 } 7276 // remember position of first insert effect claiming 7277 // last position 7278 if (iPref == EFFECT_FLAG_INSERT_LAST && 7279 idx_insert_last == -1) { 7280 idx_insert_last = i; 7281 } 7282 } 7283 } 7284 7285 // modify idx_insert from first position if needed 7286 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7287 if (idx_insert_last != -1) { 7288 idx_insert = idx_insert_last; 7289 } else { 7290 idx_insert = size; 7291 } 7292 } else { 7293 if (idx_insert_first != -1) { 7294 idx_insert = idx_insert_first + 1; 7295 } 7296 } 7297 7298 // always read samples from chain input buffer 7299 effect->setInBuffer(mInBuffer); 7300 7301 // if last effect in the chain, output samples to chain 7302 // output buffer, otherwise to chain input buffer 7303 if (idx_insert == size) { 7304 if (idx_insert != 0) { 7305 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7306 mEffects[idx_insert-1]->configure(); 7307 } 7308 effect->setOutBuffer(mOutBuffer); 7309 } else { 7310 effect->setOutBuffer(mInBuffer); 7311 } 7312 mEffects.insertAt(effect, idx_insert); 7313 7314 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7315 } 7316 effect->configure(); 7317 return NO_ERROR; 7318} 7319 7320// removeEffect_l() must be called with PlaybackThread::mLock held 7321size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7322{ 7323 Mutex::Autolock _l(mLock); 7324 int size = (int)mEffects.size(); 7325 int i; 7326 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7327 7328 for (i = 0; i < size; i++) { 7329 if (effect == mEffects[i]) { 7330 // calling stop here will remove pre-processing effect from the audio HAL. 7331 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7332 // the middle of a read from audio HAL 7333 if (mEffects[i]->state() == EffectModule::ACTIVE || 7334 mEffects[i]->state() == EffectModule::STOPPING) { 7335 mEffects[i]->stop(); 7336 } 7337 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7338 delete[] effect->inBuffer(); 7339 } else { 7340 if (i == size - 1 && i != 0) { 7341 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7342 mEffects[i - 1]->configure(); 7343 } 7344 } 7345 mEffects.removeAt(i); 7346 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7347 break; 7348 } 7349 } 7350 7351 return mEffects.size(); 7352} 7353 7354// setDevice_l() must be called with PlaybackThread::mLock held 7355void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7356{ 7357 size_t size = mEffects.size(); 7358 for (size_t i = 0; i < size; i++) { 7359 mEffects[i]->setDevice(device); 7360 } 7361} 7362 7363// setMode_l() must be called with PlaybackThread::mLock held 7364void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7365{ 7366 size_t size = mEffects.size(); 7367 for (size_t i = 0; i < size; i++) { 7368 mEffects[i]->setMode(mode); 7369 } 7370} 7371 7372// setVolume_l() must be called with PlaybackThread::mLock held 7373bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7374{ 7375 uint32_t newLeft = *left; 7376 uint32_t newRight = *right; 7377 bool hasControl = false; 7378 int ctrlIdx = -1; 7379 size_t size = mEffects.size(); 7380 7381 // first update volume controller 7382 for (size_t i = size; i > 0; i--) { 7383 if (mEffects[i - 1]->isProcessEnabled() && 7384 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7385 ctrlIdx = i - 1; 7386 hasControl = true; 7387 break; 7388 } 7389 } 7390 7391 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7392 if (hasControl) { 7393 *left = mNewLeftVolume; 7394 *right = mNewRightVolume; 7395 } 7396 return hasControl; 7397 } 7398 7399 mVolumeCtrlIdx = ctrlIdx; 7400 mLeftVolume = newLeft; 7401 mRightVolume = newRight; 7402 7403 // second get volume update from volume controller 7404 if (ctrlIdx >= 0) { 7405 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7406 mNewLeftVolume = newLeft; 7407 mNewRightVolume = newRight; 7408 } 7409 // then indicate volume to all other effects in chain. 7410 // Pass altered volume to effects before volume controller 7411 // and requested volume to effects after controller 7412 uint32_t lVol = newLeft; 7413 uint32_t rVol = newRight; 7414 7415 for (size_t i = 0; i < size; i++) { 7416 if ((int)i == ctrlIdx) continue; 7417 // this also works for ctrlIdx == -1 when there is no volume controller 7418 if ((int)i > ctrlIdx) { 7419 lVol = *left; 7420 rVol = *right; 7421 } 7422 mEffects[i]->setVolume(&lVol, &rVol, false); 7423 } 7424 *left = newLeft; 7425 *right = newRight; 7426 7427 return hasControl; 7428} 7429 7430status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7431{ 7432 const size_t SIZE = 256; 7433 char buffer[SIZE]; 7434 String8 result; 7435 7436 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7437 result.append(buffer); 7438 7439 bool locked = tryLock(mLock); 7440 // failed to lock - AudioFlinger is probably deadlocked 7441 if (!locked) { 7442 result.append("\tCould not lock mutex:\n"); 7443 } 7444 7445 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7446 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7447 mEffects.size(), 7448 (uint32_t)mInBuffer, 7449 (uint32_t)mOutBuffer, 7450 mActiveTrackCnt); 7451 result.append(buffer); 7452 write(fd, result.string(), result.size()); 7453 7454 for (size_t i = 0; i < mEffects.size(); ++i) { 7455 sp<EffectModule> effect = mEffects[i]; 7456 if (effect != 0) { 7457 effect->dump(fd, args); 7458 } 7459 } 7460 7461 if (locked) { 7462 mLock.unlock(); 7463 } 7464 7465 return NO_ERROR; 7466} 7467 7468// must be called with ThreadBase::mLock held 7469void AudioFlinger::EffectChain::setEffectSuspended_l( 7470 const effect_uuid_t *type, bool suspend) 7471{ 7472 sp<SuspendedEffectDesc> desc; 7473 // use effect type UUID timelow as key as there is no real risk of identical 7474 // timeLow fields among effect type UUIDs. 7475 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7476 if (suspend) { 7477 if (index >= 0) { 7478 desc = mSuspendedEffects.valueAt(index); 7479 } else { 7480 desc = new SuspendedEffectDesc(); 7481 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7482 mSuspendedEffects.add(type->timeLow, desc); 7483 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7484 } 7485 if (desc->mRefCount++ == 0) { 7486 sp<EffectModule> effect = getEffectIfEnabled(type); 7487 if (effect != 0) { 7488 desc->mEffect = effect; 7489 effect->setSuspended(true); 7490 effect->setEnabled(false); 7491 } 7492 } 7493 } else { 7494 if (index < 0) { 7495 return; 7496 } 7497 desc = mSuspendedEffects.valueAt(index); 7498 if (desc->mRefCount <= 0) { 7499 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7500 desc->mRefCount = 1; 7501 } 7502 if (--desc->mRefCount == 0) { 7503 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7504 if (desc->mEffect != 0) { 7505 sp<EffectModule> effect = desc->mEffect.promote(); 7506 if (effect != 0) { 7507 effect->setSuspended(false); 7508 sp<EffectHandle> handle = effect->controlHandle(); 7509 if (handle != 0) { 7510 effect->setEnabled(handle->enabled()); 7511 } 7512 } 7513 desc->mEffect.clear(); 7514 } 7515 mSuspendedEffects.removeItemsAt(index); 7516 } 7517 } 7518} 7519 7520// must be called with ThreadBase::mLock held 7521void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7522{ 7523 sp<SuspendedEffectDesc> desc; 7524 7525 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7526 if (suspend) { 7527 if (index >= 0) { 7528 desc = mSuspendedEffects.valueAt(index); 7529 } else { 7530 desc = new SuspendedEffectDesc(); 7531 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7532 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7533 } 7534 if (desc->mRefCount++ == 0) { 7535 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7536 for (size_t i = 0; i < effects.size(); i++) { 7537 setEffectSuspended_l(&effects[i]->desc().type, true); 7538 } 7539 } 7540 } else { 7541 if (index < 0) { 7542 return; 7543 } 7544 desc = mSuspendedEffects.valueAt(index); 7545 if (desc->mRefCount <= 0) { 7546 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7547 desc->mRefCount = 1; 7548 } 7549 if (--desc->mRefCount == 0) { 7550 Vector<const effect_uuid_t *> types; 7551 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7552 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7553 continue; 7554 } 7555 types.add(&mSuspendedEffects.valueAt(i)->mType); 7556 } 7557 for (size_t i = 0; i < types.size(); i++) { 7558 setEffectSuspended_l(types[i], false); 7559 } 7560 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7561 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7562 } 7563 } 7564} 7565 7566 7567// The volume effect is used for automated tests only 7568#ifndef OPENSL_ES_H_ 7569static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7570 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7571const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7572#endif //OPENSL_ES_H_ 7573 7574bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7575{ 7576 // auxiliary effects and visualizer are never suspended on output mix 7577 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7578 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7579 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7580 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7581 return false; 7582 } 7583 return true; 7584} 7585 7586Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7587{ 7588 Vector< sp<EffectModule> > effects; 7589 for (size_t i = 0; i < mEffects.size(); i++) { 7590 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7591 continue; 7592 } 7593 effects.add(mEffects[i]); 7594 } 7595 return effects; 7596} 7597 7598sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7599 const effect_uuid_t *type) 7600{ 7601 sp<EffectModule> effect = getEffectFromType_l(type); 7602 return effect != 0 && effect->isEnabled() ? effect : 0; 7603} 7604 7605void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7606 bool enabled) 7607{ 7608 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7609 if (enabled) { 7610 if (index < 0) { 7611 // if the effect is not suspend check if all effects are suspended 7612 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7613 if (index < 0) { 7614 return; 7615 } 7616 if (!isEffectEligibleForSuspend(effect->desc())) { 7617 return; 7618 } 7619 setEffectSuspended_l(&effect->desc().type, enabled); 7620 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7621 if (index < 0) { 7622 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7623 return; 7624 } 7625 } 7626 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7627 effect->desc().type.timeLow); 7628 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7629 // if effect is requested to suspended but was not yet enabled, supend it now. 7630 if (desc->mEffect == 0) { 7631 desc->mEffect = effect; 7632 effect->setEnabled(false); 7633 effect->setSuspended(true); 7634 } 7635 } else { 7636 if (index < 0) { 7637 return; 7638 } 7639 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7640 effect->desc().type.timeLow); 7641 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7642 desc->mEffect.clear(); 7643 effect->setSuspended(false); 7644 } 7645} 7646 7647#undef LOG_TAG 7648#define LOG_TAG "AudioFlinger" 7649 7650// ---------------------------------------------------------------------------- 7651 7652status_t AudioFlinger::onTransact( 7653 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7654{ 7655 return BnAudioFlinger::onTransact(code, data, reply, flags); 7656} 7657 7658}; // namespace android 7659