AudioFlinger.cpp revision a3a2cd4072aaa2d93c91251a786eb7323f8d2c27
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; 1246 chain = getEffectChain_l(sessionId); 1247 if (chain != 0) { 1248 if (type != NULL) { 1249 chain->setEffectSuspended_l(type, suspend); 1250 } else { 1251 chain->setEffectSuspendedAll_l(suspend); 1252 } 1253 } 1254 1255 updateSuspendedSessions_l(type, suspend, sessionId); 1256} 1257 1258void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1259{ 1260 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1261 if (index < 0) { 1262 return; 1263 } 1264 1265 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1266 mSuspendedSessions.editValueAt(index); 1267 1268 for (size_t i = 0; i < sessionEffects.size(); i++) { 1269 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1270 for (int j = 0; j < desc->mRefCount; j++) { 1271 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1272 chain->setEffectSuspendedAll_l(true); 1273 } else { 1274 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1275 desc->mType.timeLow); 1276 chain->setEffectSuspended_l(&desc->mType, true); 1277 } 1278 } 1279 } 1280} 1281 1282void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1283 bool suspend, 1284 int sessionId) 1285{ 1286 int index = mSuspendedSessions.indexOfKey(sessionId); 1287 1288 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1289 1290 if (suspend) { 1291 if (index >= 0) { 1292 sessionEffects = mSuspendedSessions.editValueAt(index); 1293 } else { 1294 mSuspendedSessions.add(sessionId, sessionEffects); 1295 } 1296 } else { 1297 if (index < 0) { 1298 return; 1299 } 1300 sessionEffects = mSuspendedSessions.editValueAt(index); 1301 } 1302 1303 1304 int key = EffectChain::kKeyForSuspendAll; 1305 if (type != NULL) { 1306 key = type->timeLow; 1307 } 1308 index = sessionEffects.indexOfKey(key); 1309 1310 sp <SuspendedSessionDesc> desc; 1311 if (suspend) { 1312 if (index >= 0) { 1313 desc = sessionEffects.valueAt(index); 1314 } else { 1315 desc = new SuspendedSessionDesc(); 1316 if (type != NULL) { 1317 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1318 } 1319 sessionEffects.add(key, desc); 1320 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1321 } 1322 desc->mRefCount++; 1323 } else { 1324 if (index < 0) { 1325 return; 1326 } 1327 desc = sessionEffects.valueAt(index); 1328 if (--desc->mRefCount == 0) { 1329 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1330 sessionEffects.removeItemsAt(index); 1331 if (sessionEffects.isEmpty()) { 1332 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1333 sessionId); 1334 mSuspendedSessions.removeItem(sessionId); 1335 } 1336 } 1337 } 1338 if (!sessionEffects.isEmpty()) { 1339 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1340 } 1341} 1342 1343void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1344 bool enabled, 1345 int sessionId) 1346{ 1347 Mutex::Autolock _l(mLock); 1348 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1349} 1350 1351void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1352 bool enabled, 1353 int sessionId) 1354{ 1355 if (mType != RECORD) { 1356 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1357 // another session. This gives the priority to well behaved effect control panels 1358 // and applications not using global effects. 1359 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1360 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1361 } 1362 } 1363 1364 sp<EffectChain> chain = getEffectChain_l(sessionId); 1365 if (chain != 0) { 1366 chain->checkSuspendOnEffectEnabled(effect, enabled); 1367 } 1368} 1369 1370// ---------------------------------------------------------------------------- 1371 1372AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1373 AudioStreamOut* output, 1374 int id, 1375 uint32_t device) 1376 : ThreadBase(audioFlinger, id, device), 1377 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1378 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1379{ 1380 snprintf(mName, kNameLength, "AudioOut_%d", id); 1381 1382 readOutputParameters(); 1383 1384 // Assumes constructor is called by AudioFlinger with it's mLock held, 1385 // but it would be safer to explicitly pass these as parameters 1386 mMasterVolume = mAudioFlinger->masterVolume_l(); 1387 mMasterMute = mAudioFlinger->masterMute_l(); 1388 1389 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1390 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1391 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1392 stream = (audio_stream_type_t) (stream + 1)) { 1393 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1394 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1395 // initialized by stream_type_t default constructor 1396 // mStreamTypes[stream].valid = true; 1397 } 1398} 1399 1400AudioFlinger::PlaybackThread::~PlaybackThread() 1401{ 1402 delete [] mMixBuffer; 1403} 1404 1405status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1406{ 1407 dumpInternals(fd, args); 1408 dumpTracks(fd, args); 1409 dumpEffectChains(fd, args); 1410 return NO_ERROR; 1411} 1412 1413status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1414{ 1415 const size_t SIZE = 256; 1416 char buffer[SIZE]; 1417 String8 result; 1418 1419 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1420 result.append(buffer); 1421 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1422 for (size_t i = 0; i < mTracks.size(); ++i) { 1423 sp<Track> track = mTracks[i]; 1424 if (track != 0) { 1425 track->dump(buffer, SIZE); 1426 result.append(buffer); 1427 } 1428 } 1429 1430 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1431 result.append(buffer); 1432 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1433 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1434 wp<Track> wTrack = mActiveTracks[i]; 1435 if (wTrack != 0) { 1436 sp<Track> track = wTrack.promote(); 1437 if (track != 0) { 1438 track->dump(buffer, SIZE); 1439 result.append(buffer); 1440 } 1441 } 1442 } 1443 write(fd, result.string(), result.size()); 1444 return NO_ERROR; 1445} 1446 1447status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1448{ 1449 const size_t SIZE = 256; 1450 char buffer[SIZE]; 1451 String8 result; 1452 1453 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1454 result.append(buffer); 1455 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1456 result.append(buffer); 1457 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1458 result.append(buffer); 1459 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1460 result.append(buffer); 1461 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1462 result.append(buffer); 1463 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1464 result.append(buffer); 1465 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1466 result.append(buffer); 1467 write(fd, result.string(), result.size()); 1468 1469 dumpBase(fd, args); 1470 1471 return NO_ERROR; 1472} 1473 1474// Thread virtuals 1475status_t AudioFlinger::PlaybackThread::readyToRun() 1476{ 1477 status_t status = initCheck(); 1478 if (status == NO_ERROR) { 1479 ALOGI("AudioFlinger's thread %p ready to run", this); 1480 } else { 1481 ALOGE("No working audio driver found."); 1482 } 1483 return status; 1484} 1485 1486void AudioFlinger::PlaybackThread::onFirstRef() 1487{ 1488 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1489} 1490 1491// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1492sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1493 const sp<AudioFlinger::Client>& client, 1494 audio_stream_type_t streamType, 1495 uint32_t sampleRate, 1496 audio_format_t format, 1497 uint32_t channelMask, 1498 int frameCount, 1499 const sp<IMemory>& sharedBuffer, 1500 int sessionId, 1501 status_t *status) 1502{ 1503 sp<Track> track; 1504 status_t lStatus; 1505 1506 if (mType == DIRECT) { 1507 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1508 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1509 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1510 "for output %p with format %d", 1511 sampleRate, format, channelMask, mOutput, mFormat); 1512 lStatus = BAD_VALUE; 1513 goto Exit; 1514 } 1515 } 1516 } else { 1517 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1518 if (sampleRate > mSampleRate*2) { 1519 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1520 lStatus = BAD_VALUE; 1521 goto Exit; 1522 } 1523 } 1524 1525 lStatus = initCheck(); 1526 if (lStatus != NO_ERROR) { 1527 ALOGE("Audio driver not initialized."); 1528 goto Exit; 1529 } 1530 1531 { // scope for mLock 1532 Mutex::Autolock _l(mLock); 1533 1534 // all tracks in same audio session must share the same routing strategy otherwise 1535 // conflicts will happen when tracks are moved from one output to another by audio policy 1536 // manager 1537 uint32_t strategy = 1538 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1539 for (size_t i = 0; i < mTracks.size(); ++i) { 1540 sp<Track> t = mTracks[i]; 1541 if (t != 0) { 1542 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1543 if (sessionId == t->sessionId() && strategy != actual) { 1544 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1545 strategy, actual); 1546 lStatus = BAD_VALUE; 1547 goto Exit; 1548 } 1549 } 1550 } 1551 1552 track = new Track(this, client, streamType, sampleRate, format, 1553 channelMask, frameCount, sharedBuffer, sessionId); 1554 if (track->getCblk() == NULL || track->name() < 0) { 1555 lStatus = NO_MEMORY; 1556 goto Exit; 1557 } 1558 mTracks.add(track); 1559 1560 sp<EffectChain> chain = getEffectChain_l(sessionId); 1561 if (chain != 0) { 1562 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1563 track->setMainBuffer(chain->inBuffer()); 1564 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1565 chain->incTrackCnt(); 1566 } 1567 1568 // invalidate track immediately if the stream type was moved to another thread since 1569 // createTrack() was called by the client process. 1570 if (!mStreamTypes[streamType].valid) { 1571 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1572 this, streamType); 1573 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1574 } 1575 } 1576 lStatus = NO_ERROR; 1577 1578Exit: 1579 if(status) { 1580 *status = lStatus; 1581 } 1582 return track; 1583} 1584 1585uint32_t AudioFlinger::PlaybackThread::latency() const 1586{ 1587 Mutex::Autolock _l(mLock); 1588 if (initCheck() == NO_ERROR) { 1589 return mOutput->stream->get_latency(mOutput->stream); 1590 } else { 1591 return 0; 1592 } 1593} 1594 1595status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1596{ 1597 mMasterVolume = value; 1598 return NO_ERROR; 1599} 1600 1601status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1602{ 1603 mMasterMute = muted; 1604 return NO_ERROR; 1605} 1606 1607float AudioFlinger::PlaybackThread::masterVolume() const 1608{ 1609 return mMasterVolume; 1610} 1611 1612bool AudioFlinger::PlaybackThread::masterMute() const 1613{ 1614 return mMasterMute; 1615} 1616 1617status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1618{ 1619 mStreamTypes[stream].volume = value; 1620 return NO_ERROR; 1621} 1622 1623status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1624{ 1625 mStreamTypes[stream].mute = muted; 1626 return NO_ERROR; 1627} 1628 1629float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1630{ 1631 return mStreamTypes[stream].volume; 1632} 1633 1634bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1635{ 1636 return mStreamTypes[stream].mute; 1637} 1638 1639// addTrack_l() must be called with ThreadBase::mLock held 1640status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1641{ 1642 status_t status = ALREADY_EXISTS; 1643 1644 // set retry count for buffer fill 1645 track->mRetryCount = kMaxTrackStartupRetries; 1646 if (mActiveTracks.indexOf(track) < 0) { 1647 // the track is newly added, make sure it fills up all its 1648 // buffers before playing. This is to ensure the client will 1649 // effectively get the latency it requested. 1650 track->mFillingUpStatus = Track::FS_FILLING; 1651 track->mResetDone = false; 1652 mActiveTracks.add(track); 1653 if (track->mainBuffer() != mMixBuffer) { 1654 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1655 if (chain != 0) { 1656 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1657 chain->incActiveTrackCnt(); 1658 } 1659 } 1660 1661 status = NO_ERROR; 1662 } 1663 1664 ALOGV("mWaitWorkCV.broadcast"); 1665 mWaitWorkCV.broadcast(); 1666 1667 return status; 1668} 1669 1670// destroyTrack_l() must be called with ThreadBase::mLock held 1671void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1672{ 1673 track->mState = TrackBase::TERMINATED; 1674 if (mActiveTracks.indexOf(track) < 0) { 1675 removeTrack_l(track); 1676 } 1677} 1678 1679void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1680{ 1681 mTracks.remove(track); 1682 deleteTrackName_l(track->name()); 1683 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1684 if (chain != 0) { 1685 chain->decTrackCnt(); 1686 } 1687} 1688 1689String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1690{ 1691 String8 out_s8 = String8(""); 1692 char *s; 1693 1694 Mutex::Autolock _l(mLock); 1695 if (initCheck() != NO_ERROR) { 1696 return out_s8; 1697 } 1698 1699 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1700 out_s8 = String8(s); 1701 free(s); 1702 return out_s8; 1703} 1704 1705// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1706void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1707 AudioSystem::OutputDescriptor desc; 1708 void *param2 = 0; 1709 1710 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1711 1712 switch (event) { 1713 case AudioSystem::OUTPUT_OPENED: 1714 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1715 desc.channels = mChannelMask; 1716 desc.samplingRate = mSampleRate; 1717 desc.format = mFormat; 1718 desc.frameCount = mFrameCount; 1719 desc.latency = latency(); 1720 param2 = &desc; 1721 break; 1722 1723 case AudioSystem::STREAM_CONFIG_CHANGED: 1724 param2 = ¶m; 1725 case AudioSystem::OUTPUT_CLOSED: 1726 default: 1727 break; 1728 } 1729 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1730} 1731 1732void AudioFlinger::PlaybackThread::readOutputParameters() 1733{ 1734 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1735 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1736 mChannelCount = (uint16_t)popcount(mChannelMask); 1737 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1738 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1739 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1740 1741 // FIXME - Current mixer implementation only supports stereo output: Always 1742 // Allocate a stereo buffer even if HW output is mono. 1743 if (mMixBuffer != NULL) delete[] mMixBuffer; 1744 mMixBuffer = new int16_t[mFrameCount * 2]; 1745 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1746 1747 // force reconfiguration of effect chains and engines to take new buffer size and audio 1748 // parameters into account 1749 // Note that mLock is not held when readOutputParameters() is called from the constructor 1750 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1751 // matter. 1752 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1753 Vector< sp<EffectChain> > effectChains = mEffectChains; 1754 for (size_t i = 0; i < effectChains.size(); i ++) { 1755 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1756 } 1757} 1758 1759status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1760{ 1761 if (halFrames == 0 || dspFrames == 0) { 1762 return BAD_VALUE; 1763 } 1764 Mutex::Autolock _l(mLock); 1765 if (initCheck() != NO_ERROR) { 1766 return INVALID_OPERATION; 1767 } 1768 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1769 1770 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1771} 1772 1773uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1774{ 1775 Mutex::Autolock _l(mLock); 1776 uint32_t result = 0; 1777 if (getEffectChain_l(sessionId) != 0) { 1778 result = EFFECT_SESSION; 1779 } 1780 1781 for (size_t i = 0; i < mTracks.size(); ++i) { 1782 sp<Track> track = mTracks[i]; 1783 if (sessionId == track->sessionId() && 1784 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1785 result |= TRACK_SESSION; 1786 break; 1787 } 1788 } 1789 1790 return result; 1791} 1792 1793uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1794{ 1795 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1796 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1797 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1798 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1799 } 1800 for (size_t i = 0; i < mTracks.size(); i++) { 1801 sp<Track> track = mTracks[i]; 1802 if (sessionId == track->sessionId() && 1803 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1804 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1805 } 1806 } 1807 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1808} 1809 1810 1811AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1812{ 1813 Mutex::Autolock _l(mLock); 1814 return mOutput; 1815} 1816 1817AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1818{ 1819 Mutex::Autolock _l(mLock); 1820 AudioStreamOut *output = mOutput; 1821 mOutput = NULL; 1822 return output; 1823} 1824 1825// this method must always be called either with ThreadBase mLock held or inside the thread loop 1826audio_stream_t* AudioFlinger::PlaybackThread::stream() 1827{ 1828 if (mOutput == NULL) { 1829 return NULL; 1830 } 1831 return &mOutput->stream->common; 1832} 1833 1834uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1835{ 1836 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1837 // decoding and transfer time. So sleeping for half of the latency would likely cause 1838 // underruns 1839 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1840 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1841 } else { 1842 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1843 } 1844} 1845 1846// ---------------------------------------------------------------------------- 1847 1848AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1849 : PlaybackThread(audioFlinger, output, id, device), 1850 mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE) 1851{ 1852 mType = ThreadBase::MIXER; 1853 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1854 1855 // FIXME - Current mixer implementation only supports stereo output 1856 if (mChannelCount == 1) { 1857 ALOGE("Invalid audio hardware channel count"); 1858 } 1859} 1860 1861AudioFlinger::MixerThread::~MixerThread() 1862{ 1863 delete mAudioMixer; 1864} 1865 1866bool AudioFlinger::MixerThread::threadLoop() 1867{ 1868 Vector< sp<Track> > tracksToRemove; 1869 mixer_state mixerStatus = MIXER_IDLE; 1870 nsecs_t standbyTime = systemTime(); 1871 size_t mixBufferSize = mFrameCount * mFrameSize; 1872 // FIXME: Relaxed timing because of a certain device that can't meet latency 1873 // Should be reduced to 2x after the vendor fixes the driver issue 1874 // increase threshold again due to low power audio mode. The way this warning threshold is 1875 // calculated and its usefulness should be reconsidered anyway. 1876 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1877 nsecs_t lastWarning = 0; 1878 bool longStandbyExit = false; 1879 uint32_t activeSleepTime = activeSleepTimeUs(); 1880 uint32_t idleSleepTime = idleSleepTimeUs(); 1881 uint32_t sleepTime = idleSleepTime; 1882 uint32_t sleepTimeShift = 0; 1883 Vector< sp<EffectChain> > effectChains; 1884#ifdef DEBUG_CPU_USAGE 1885 ThreadCpuUsage cpu; 1886 const CentralTendencyStatistics& stats = cpu.statistics(); 1887#endif 1888 1889 acquireWakeLock(); 1890 1891 while (!exitPending()) 1892 { 1893#ifdef DEBUG_CPU_USAGE 1894 cpu.sampleAndEnable(); 1895 unsigned n = stats.n(); 1896 // cpu.elapsed() is expensive, so don't call it every loop 1897 if ((n & 127) == 1) { 1898 long long elapsed = cpu.elapsed(); 1899 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1900 double perLoop = elapsed / (double) n; 1901 double perLoop100 = perLoop * 0.01; 1902 double mean = stats.mean(); 1903 double stddev = stats.stddev(); 1904 double minimum = stats.minimum(); 1905 double maximum = stats.maximum(); 1906 cpu.resetStatistics(); 1907 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", 1908 elapsed * .000000001, n, perLoop * .000001, 1909 mean * .001, 1910 stddev * .001, 1911 minimum * .001, 1912 maximum * .001, 1913 mean / perLoop100, 1914 stddev / perLoop100, 1915 minimum / perLoop100, 1916 maximum / perLoop100); 1917 } 1918 } 1919#endif 1920 processConfigEvents(); 1921 1922 mixerStatus = MIXER_IDLE; 1923 { // scope for mLock 1924 1925 Mutex::Autolock _l(mLock); 1926 1927 if (checkForNewParameters_l()) { 1928 mixBufferSize = mFrameCount * mFrameSize; 1929 // FIXME: Relaxed timing because of a certain device that can't meet latency 1930 // Should be reduced to 2x after the vendor fixes the driver issue 1931 // increase threshold again due to low power audio mode. The way this warning 1932 // threshold is calculated and its usefulness should be reconsidered anyway. 1933 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1934 activeSleepTime = activeSleepTimeUs(); 1935 idleSleepTime = idleSleepTimeUs(); 1936 } 1937 1938 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1939 1940 // put audio hardware into standby after short delay 1941 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1942 mSuspended)) { 1943 if (!mStandby) { 1944 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1945 mOutput->stream->common.standby(&mOutput->stream->common); 1946 mStandby = true; 1947 mBytesWritten = 0; 1948 } 1949 1950 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1951 // we're about to wait, flush the binder command buffer 1952 IPCThreadState::self()->flushCommands(); 1953 1954 if (exitPending()) break; 1955 1956 releaseWakeLock_l(); 1957 // wait until we have something to do... 1958 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1959 mWaitWorkCV.wait(mLock); 1960 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1961 acquireWakeLock_l(); 1962 1963 mPrevMixerStatus = MIXER_IDLE; 1964 if (!mMasterMute) { 1965 char value[PROPERTY_VALUE_MAX]; 1966 property_get("ro.audio.silent", value, "0"); 1967 if (atoi(value)) { 1968 ALOGD("Silence is golden"); 1969 setMasterMute(true); 1970 } 1971 } 1972 1973 standbyTime = systemTime() + kStandbyTimeInNsecs; 1974 sleepTime = idleSleepTime; 1975 sleepTimeShift = 0; 1976 continue; 1977 } 1978 } 1979 1980 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1981 1982 // prevent any changes in effect chain list and in each effect chain 1983 // during mixing and effect process as the audio buffers could be deleted 1984 // or modified if an effect is created or deleted 1985 lockEffectChains_l(effectChains); 1986 } 1987 1988 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1989 // mix buffers... 1990 mAudioMixer->process(); 1991 // increase sleep time progressively when application underrun condition clears. 1992 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1993 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1994 // such that we would underrun the audio HAL. 1995 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 1996 sleepTimeShift--; 1997 } 1998 sleepTime = 0; 1999 standbyTime = systemTime() + kStandbyTimeInNsecs; 2000 //TODO: delay standby when effects have a tail 2001 } else { 2002 // If no tracks are ready, sleep once for the duration of an output 2003 // buffer size, then write 0s to the output 2004 if (sleepTime == 0) { 2005 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2006 sleepTime = activeSleepTime >> sleepTimeShift; 2007 if (sleepTime < kMinThreadSleepTimeUs) { 2008 sleepTime = kMinThreadSleepTimeUs; 2009 } 2010 // reduce sleep time in case of consecutive application underruns to avoid 2011 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2012 // duration we would end up writing less data than needed by the audio HAL if 2013 // the condition persists. 2014 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2015 sleepTimeShift++; 2016 } 2017 } else { 2018 sleepTime = idleSleepTime; 2019 } 2020 } else if (mBytesWritten != 0 || 2021 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2022 memset (mMixBuffer, 0, mixBufferSize); 2023 sleepTime = 0; 2024 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2025 } 2026 // TODO add standby time extension fct of effect tail 2027 } 2028 2029 if (mSuspended) { 2030 sleepTime = suspendSleepTimeUs(); 2031 } 2032 // sleepTime == 0 means we must write to audio hardware 2033 if (sleepTime == 0) { 2034 for (size_t i = 0; i < effectChains.size(); i ++) { 2035 effectChains[i]->process_l(); 2036 } 2037 // enable changes in effect chain 2038 unlockEffectChains(effectChains); 2039 mLastWriteTime = systemTime(); 2040 mInWrite = true; 2041 mBytesWritten += mixBufferSize; 2042 2043 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2044 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2045 mNumWrites++; 2046 mInWrite = false; 2047 nsecs_t now = systemTime(); 2048 nsecs_t delta = now - mLastWriteTime; 2049 if (!mStandby && delta > maxPeriod) { 2050 mNumDelayedWrites++; 2051 if ((now - lastWarning) > kWarningThrottleNs) { 2052 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2053 ns2ms(delta), mNumDelayedWrites, this); 2054 lastWarning = now; 2055 } 2056 if (mStandby) { 2057 longStandbyExit = true; 2058 } 2059 } 2060 mStandby = false; 2061 } else { 2062 // enable changes in effect chain 2063 unlockEffectChains(effectChains); 2064 usleep(sleepTime); 2065 } 2066 2067 // finally let go of all our tracks, without the lock held 2068 // since we can't guarantee the destructors won't acquire that 2069 // same lock. 2070 tracksToRemove.clear(); 2071 2072 // Effect chains will be actually deleted here if they were removed from 2073 // mEffectChains list during mixing or effects processing 2074 effectChains.clear(); 2075 } 2076 2077 if (!mStandby) { 2078 mOutput->stream->common.standby(&mOutput->stream->common); 2079 } 2080 2081 releaseWakeLock(); 2082 2083 ALOGV("MixerThread %p exiting", this); 2084 return false; 2085} 2086 2087// prepareTracks_l() must be called with ThreadBase::mLock held 2088AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2089 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2090{ 2091 2092 mixer_state mixerStatus = MIXER_IDLE; 2093 // find out which tracks need to be processed 2094 size_t count = activeTracks.size(); 2095 size_t mixedTracks = 0; 2096 size_t tracksWithEffect = 0; 2097 2098 float masterVolume = mMasterVolume; 2099 bool masterMute = mMasterMute; 2100 2101 if (masterMute) { 2102 masterVolume = 0; 2103 } 2104 // Delegate master volume control to effect in output mix effect chain if needed 2105 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2106 if (chain != 0) { 2107 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2108 chain->setVolume_l(&v, &v); 2109 masterVolume = (float)((v + (1 << 23)) >> 24); 2110 chain.clear(); 2111 } 2112 2113 for (size_t i=0 ; i<count ; i++) { 2114 sp<Track> t = activeTracks[i].promote(); 2115 if (t == 0) continue; 2116 2117 // this const just means the local variable doesn't change 2118 Track* const track = t.get(); 2119 audio_track_cblk_t* cblk = track->cblk(); 2120 2121 // The first time a track is added we wait 2122 // for all its buffers to be filled before processing it 2123 int name = track->name(); 2124 // make sure that we have enough frames to mix one full buffer. 2125 // enforce this condition only once to enable draining the buffer in case the client 2126 // app does not call stop() and relies on underrun to stop: 2127 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2128 // during last round 2129 uint32_t minFrames = 1; 2130 if (!track->isStopped() && !track->isPausing() && 2131 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2132 if (t->sampleRate() == (int)mSampleRate) { 2133 minFrames = mFrameCount; 2134 } else { 2135 // +1 for rounding and +1 for additional sample needed for interpolation 2136 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2137 // add frames already consumed but not yet released by the resampler 2138 // because cblk->framesReady() will include these frames 2139 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2140 // the minimum track buffer size is normally twice the number of frames necessary 2141 // to fill one buffer and the resampler should not leave more than one buffer worth 2142 // of unreleased frames after each pass, but just in case... 2143 ALOG_ASSERT(minFrames <= cblk->frameCount); 2144 } 2145 } 2146 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2147 !track->isPaused() && !track->isTerminated()) 2148 { 2149 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2150 2151 mixedTracks++; 2152 2153 // track->mainBuffer() != mMixBuffer means there is an effect chain 2154 // connected to the track 2155 chain.clear(); 2156 if (track->mainBuffer() != mMixBuffer) { 2157 chain = getEffectChain_l(track->sessionId()); 2158 // Delegate volume control to effect in track effect chain if needed 2159 if (chain != 0) { 2160 tracksWithEffect++; 2161 } else { 2162 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2163 name, track->sessionId()); 2164 } 2165 } 2166 2167 2168 int param = AudioMixer::VOLUME; 2169 if (track->mFillingUpStatus == Track::FS_FILLED) { 2170 // no ramp for the first volume setting 2171 track->mFillingUpStatus = Track::FS_ACTIVE; 2172 if (track->mState == TrackBase::RESUMING) { 2173 track->mState = TrackBase::ACTIVE; 2174 param = AudioMixer::RAMP_VOLUME; 2175 } 2176 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2177 } else if (cblk->server != 0) { 2178 // If the track is stopped before the first frame was mixed, 2179 // do not apply ramp 2180 param = AudioMixer::RAMP_VOLUME; 2181 } 2182 2183 // compute volume for this track 2184 uint32_t vl, vr, va; 2185 if (track->isMuted() || track->isPausing() || 2186 mStreamTypes[track->type()].mute) { 2187 vl = vr = va = 0; 2188 if (track->isPausing()) { 2189 track->setPaused(); 2190 } 2191 } else { 2192 2193 // read original volumes with volume control 2194 float typeVolume = mStreamTypes[track->type()].volume; 2195 float v = masterVolume * typeVolume; 2196 uint32_t vlr = cblk->volumeLR; 2197 vl = vlr & 0xFFFF; 2198 vr = vlr >> 16; 2199 // track volumes come from shared memory, so can't be trusted and must be clamped 2200 if (vl > MAX_GAIN_INT) { 2201 ALOGV("Track left volume out of range: %04X", vl); 2202 vl = MAX_GAIN_INT; 2203 } 2204 if (vr > MAX_GAIN_INT) { 2205 ALOGV("Track right volume out of range: %04X", vr); 2206 vr = MAX_GAIN_INT; 2207 } 2208 // now apply the master volume and stream type volume 2209 vl = (uint32_t)(v * vl) << 12; 2210 vr = (uint32_t)(v * vr) << 12; 2211 // assuming master volume and stream type volume each go up to 1.0, 2212 // vl and vr are now in 8.24 format 2213 2214 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2215 // send level comes from shared memory and so may be corrupt 2216 if (sendLevel >= MAX_GAIN_INT) { 2217 ALOGV("Track send level out of range: %04X", sendLevel); 2218 sendLevel = MAX_GAIN_INT; 2219 } 2220 va = (uint32_t)(v * sendLevel); 2221 } 2222 // Delegate volume control to effect in track effect chain if needed 2223 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2224 // Do not ramp volume if volume is controlled by effect 2225 param = AudioMixer::VOLUME; 2226 track->mHasVolumeController = true; 2227 } else { 2228 // force no volume ramp when volume controller was just disabled or removed 2229 // from effect chain to avoid volume spike 2230 if (track->mHasVolumeController) { 2231 param = AudioMixer::VOLUME; 2232 } 2233 track->mHasVolumeController = false; 2234 } 2235 2236 // Convert volumes from 8.24 to 4.12 format 2237 int16_t left, right, aux; 2238 // This additional clamping is needed in case chain->setVolume_l() overshot 2239 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2240 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2241 left = int16_t(v_clamped); 2242 v_clamped = (vr + (1 << 11)) >> 12; 2243 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2244 right = int16_t(v_clamped); 2245 2246 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2247 aux = int16_t(va); 2248 2249 // XXX: these things DON'T need to be done each time 2250 mAudioMixer->setBufferProvider(name, track); 2251 mAudioMixer->enable(name); 2252 2253 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2254 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2255 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2256 mAudioMixer->setParameter( 2257 name, 2258 AudioMixer::TRACK, 2259 AudioMixer::FORMAT, (void *)track->format()); 2260 mAudioMixer->setParameter( 2261 name, 2262 AudioMixer::TRACK, 2263 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2264 mAudioMixer->setParameter( 2265 name, 2266 AudioMixer::RESAMPLE, 2267 AudioMixer::SAMPLE_RATE, 2268 (void *)(cblk->sampleRate)); 2269 mAudioMixer->setParameter( 2270 name, 2271 AudioMixer::TRACK, 2272 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2273 mAudioMixer->setParameter( 2274 name, 2275 AudioMixer::TRACK, 2276 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2277 2278 // reset retry count 2279 track->mRetryCount = kMaxTrackRetries; 2280 // If one track is ready, set the mixer ready if: 2281 // - the mixer was not ready during previous round OR 2282 // - no other track is not ready 2283 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2284 mixerStatus != MIXER_TRACKS_ENABLED) { 2285 mixerStatus = MIXER_TRACKS_READY; 2286 } 2287 } else { 2288 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2289 if (track->isStopped()) { 2290 track->reset(); 2291 } 2292 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2293 // We have consumed all the buffers of this track. 2294 // Remove it from the list of active tracks. 2295 tracksToRemove->add(track); 2296 } else { 2297 // No buffers for this track. Give it a few chances to 2298 // fill a buffer, then remove it from active list. 2299 if (--(track->mRetryCount) <= 0) { 2300 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2301 tracksToRemove->add(track); 2302 // indicate to client process that the track was disabled because of underrun 2303 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2304 // If one track is not ready, mark the mixer also not ready if: 2305 // - the mixer was ready during previous round OR 2306 // - no other track is ready 2307 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2308 mixerStatus != MIXER_TRACKS_READY) { 2309 mixerStatus = MIXER_TRACKS_ENABLED; 2310 } 2311 } 2312 mAudioMixer->disable(name); 2313 } 2314 } 2315 2316 // remove all the tracks that need to be... 2317 count = tracksToRemove->size(); 2318 if (CC_UNLIKELY(count)) { 2319 for (size_t i=0 ; i<count ; i++) { 2320 const sp<Track>& track = tracksToRemove->itemAt(i); 2321 mActiveTracks.remove(track); 2322 if (track->mainBuffer() != mMixBuffer) { 2323 chain = getEffectChain_l(track->sessionId()); 2324 if (chain != 0) { 2325 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2326 chain->decActiveTrackCnt(); 2327 } 2328 } 2329 if (track->isTerminated()) { 2330 removeTrack_l(track); 2331 } 2332 } 2333 } 2334 2335 // mix buffer must be cleared if all tracks are connected to an 2336 // effect chain as in this case the mixer will not write to 2337 // mix buffer and track effects will accumulate into it 2338 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2339 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2340 } 2341 2342 mPrevMixerStatus = mixerStatus; 2343 return mixerStatus; 2344} 2345 2346void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2347{ 2348 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2349 this, streamType, mTracks.size()); 2350 Mutex::Autolock _l(mLock); 2351 2352 size_t size = mTracks.size(); 2353 for (size_t i = 0; i < size; i++) { 2354 sp<Track> t = mTracks[i]; 2355 if (t->type() == streamType) { 2356 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2357 t->mCblk->cv.signal(); 2358 } 2359 } 2360} 2361 2362void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2363{ 2364 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2365 this, streamType, valid); 2366 Mutex::Autolock _l(mLock); 2367 2368 mStreamTypes[streamType].valid = valid; 2369} 2370 2371// getTrackName_l() must be called with ThreadBase::mLock held 2372int AudioFlinger::MixerThread::getTrackName_l() 2373{ 2374 return mAudioMixer->getTrackName(); 2375} 2376 2377// deleteTrackName_l() must be called with ThreadBase::mLock held 2378void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2379{ 2380 ALOGV("remove track (%d) and delete from mixer", name); 2381 mAudioMixer->deleteTrackName(name); 2382} 2383 2384// checkForNewParameters_l() must be called with ThreadBase::mLock held 2385bool AudioFlinger::MixerThread::checkForNewParameters_l() 2386{ 2387 bool reconfig = false; 2388 2389 while (!mNewParameters.isEmpty()) { 2390 status_t status = NO_ERROR; 2391 String8 keyValuePair = mNewParameters[0]; 2392 AudioParameter param = AudioParameter(keyValuePair); 2393 int value; 2394 2395 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2396 reconfig = true; 2397 } 2398 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2399 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2400 status = BAD_VALUE; 2401 } else { 2402 reconfig = true; 2403 } 2404 } 2405 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2406 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2407 status = BAD_VALUE; 2408 } else { 2409 reconfig = true; 2410 } 2411 } 2412 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2413 // do not accept frame count changes if tracks are open as the track buffer 2414 // size depends on frame count and correct behavior would not be guaranteed 2415 // if frame count is changed after track creation 2416 if (!mTracks.isEmpty()) { 2417 status = INVALID_OPERATION; 2418 } else { 2419 reconfig = true; 2420 } 2421 } 2422 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2423 // when changing the audio output device, call addBatteryData to notify 2424 // the change 2425 if ((int)mDevice != value) { 2426 uint32_t params = 0; 2427 // check whether speaker is on 2428 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2429 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2430 } 2431 2432 int deviceWithoutSpeaker 2433 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2434 // check if any other device (except speaker) is on 2435 if (value & deviceWithoutSpeaker ) { 2436 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2437 } 2438 2439 if (params != 0) { 2440 addBatteryData(params); 2441 } 2442 } 2443 2444 // forward device change to effects that have requested to be 2445 // aware of attached audio device. 2446 mDevice = (uint32_t)value; 2447 for (size_t i = 0; i < mEffectChains.size(); i++) { 2448 mEffectChains[i]->setDevice_l(mDevice); 2449 } 2450 } 2451 2452 if (status == NO_ERROR) { 2453 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2454 keyValuePair.string()); 2455 if (!mStandby && status == INVALID_OPERATION) { 2456 mOutput->stream->common.standby(&mOutput->stream->common); 2457 mStandby = true; 2458 mBytesWritten = 0; 2459 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2460 keyValuePair.string()); 2461 } 2462 if (status == NO_ERROR && reconfig) { 2463 delete mAudioMixer; 2464 readOutputParameters(); 2465 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2466 for (size_t i = 0; i < mTracks.size() ; i++) { 2467 int name = getTrackName_l(); 2468 if (name < 0) break; 2469 mTracks[i]->mName = name; 2470 // limit track sample rate to 2 x new output sample rate 2471 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2472 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2473 } 2474 } 2475 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2476 } 2477 } 2478 2479 mNewParameters.removeAt(0); 2480 2481 mParamStatus = status; 2482 mParamCond.signal(); 2483 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2484 // already timed out waiting for the status and will never signal the condition. 2485 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2486 } 2487 return reconfig; 2488} 2489 2490status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2491{ 2492 const size_t SIZE = 256; 2493 char buffer[SIZE]; 2494 String8 result; 2495 2496 PlaybackThread::dumpInternals(fd, args); 2497 2498 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2499 result.append(buffer); 2500 write(fd, result.string(), result.size()); 2501 return NO_ERROR; 2502} 2503 2504uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2505{ 2506 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2507} 2508 2509uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2510{ 2511 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2512} 2513 2514// ---------------------------------------------------------------------------- 2515AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2516 : PlaybackThread(audioFlinger, output, id, device) 2517{ 2518 mType = ThreadBase::DIRECT; 2519} 2520 2521AudioFlinger::DirectOutputThread::~DirectOutputThread() 2522{ 2523} 2524 2525static inline 2526int32_t mul(int16_t in, int16_t v) 2527{ 2528#if defined(__arm__) && !defined(__thumb__) 2529 int32_t out; 2530 asm( "smulbb %[out], %[in], %[v] \n" 2531 : [out]"=r"(out) 2532 : [in]"%r"(in), [v]"r"(v) 2533 : ); 2534 return out; 2535#else 2536 return in * int32_t(v); 2537#endif 2538} 2539 2540void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2541{ 2542 // Do not apply volume on compressed audio 2543 if (!audio_is_linear_pcm(mFormat)) { 2544 return; 2545 } 2546 2547 // convert to signed 16 bit before volume calculation 2548 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2549 size_t count = mFrameCount * mChannelCount; 2550 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2551 int16_t *dst = mMixBuffer + count-1; 2552 while(count--) { 2553 *dst-- = (int16_t)(*src--^0x80) << 8; 2554 } 2555 } 2556 2557 size_t frameCount = mFrameCount; 2558 int16_t *out = mMixBuffer; 2559 if (ramp) { 2560 if (mChannelCount == 1) { 2561 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2562 int32_t vlInc = d / (int32_t)frameCount; 2563 int32_t vl = ((int32_t)mLeftVolShort << 16); 2564 do { 2565 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2566 out++; 2567 vl += vlInc; 2568 } while (--frameCount); 2569 2570 } else { 2571 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2572 int32_t vlInc = d / (int32_t)frameCount; 2573 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2574 int32_t vrInc = d / (int32_t)frameCount; 2575 int32_t vl = ((int32_t)mLeftVolShort << 16); 2576 int32_t vr = ((int32_t)mRightVolShort << 16); 2577 do { 2578 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2579 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2580 out += 2; 2581 vl += vlInc; 2582 vr += vrInc; 2583 } while (--frameCount); 2584 } 2585 } else { 2586 if (mChannelCount == 1) { 2587 do { 2588 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2589 out++; 2590 } while (--frameCount); 2591 } else { 2592 do { 2593 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2594 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2595 out += 2; 2596 } while (--frameCount); 2597 } 2598 } 2599 2600 // convert back to unsigned 8 bit after volume calculation 2601 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2602 size_t count = mFrameCount * mChannelCount; 2603 int16_t *src = mMixBuffer; 2604 uint8_t *dst = (uint8_t *)mMixBuffer; 2605 while(count--) { 2606 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2607 } 2608 } 2609 2610 mLeftVolShort = leftVol; 2611 mRightVolShort = rightVol; 2612} 2613 2614bool AudioFlinger::DirectOutputThread::threadLoop() 2615{ 2616 mixer_state mixerStatus = MIXER_IDLE; 2617 sp<Track> trackToRemove; 2618 sp<Track> activeTrack; 2619 nsecs_t standbyTime = systemTime(); 2620 int8_t *curBuf; 2621 size_t mixBufferSize = mFrameCount*mFrameSize; 2622 uint32_t activeSleepTime = activeSleepTimeUs(); 2623 uint32_t idleSleepTime = idleSleepTimeUs(); 2624 uint32_t sleepTime = idleSleepTime; 2625 // use shorter standby delay as on normal output to release 2626 // hardware resources as soon as possible 2627 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2628 2629 acquireWakeLock(); 2630 2631 while (!exitPending()) 2632 { 2633 bool rampVolume; 2634 uint16_t leftVol; 2635 uint16_t rightVol; 2636 Vector< sp<EffectChain> > effectChains; 2637 2638 processConfigEvents(); 2639 2640 mixerStatus = MIXER_IDLE; 2641 2642 { // scope for the mLock 2643 2644 Mutex::Autolock _l(mLock); 2645 2646 if (checkForNewParameters_l()) { 2647 mixBufferSize = mFrameCount*mFrameSize; 2648 activeSleepTime = activeSleepTimeUs(); 2649 idleSleepTime = idleSleepTimeUs(); 2650 standbyDelay = microseconds(activeSleepTime*2); 2651 } 2652 2653 // put audio hardware into standby after short delay 2654 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2655 mSuspended)) { 2656 // wait until we have something to do... 2657 if (!mStandby) { 2658 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2659 mOutput->stream->common.standby(&mOutput->stream->common); 2660 mStandby = true; 2661 mBytesWritten = 0; 2662 } 2663 2664 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2665 // we're about to wait, flush the binder command buffer 2666 IPCThreadState::self()->flushCommands(); 2667 2668 if (exitPending()) break; 2669 2670 releaseWakeLock_l(); 2671 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2672 mWaitWorkCV.wait(mLock); 2673 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2674 acquireWakeLock_l(); 2675 2676 if (!mMasterMute) { 2677 char value[PROPERTY_VALUE_MAX]; 2678 property_get("ro.audio.silent", value, "0"); 2679 if (atoi(value)) { 2680 ALOGD("Silence is golden"); 2681 setMasterMute(true); 2682 } 2683 } 2684 2685 standbyTime = systemTime() + standbyDelay; 2686 sleepTime = idleSleepTime; 2687 continue; 2688 } 2689 } 2690 2691 effectChains = mEffectChains; 2692 2693 // find out which tracks need to be processed 2694 if (mActiveTracks.size() != 0) { 2695 sp<Track> t = mActiveTracks[0].promote(); 2696 if (t == 0) continue; 2697 2698 Track* const track = t.get(); 2699 audio_track_cblk_t* cblk = track->cblk(); 2700 2701 // The first time a track is added we wait 2702 // for all its buffers to be filled before processing it 2703 if (cblk->framesReady() && track->isReady() && 2704 !track->isPaused() && !track->isTerminated()) 2705 { 2706 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2707 2708 if (track->mFillingUpStatus == Track::FS_FILLED) { 2709 track->mFillingUpStatus = Track::FS_ACTIVE; 2710 mLeftVolFloat = mRightVolFloat = 0; 2711 mLeftVolShort = mRightVolShort = 0; 2712 if (track->mState == TrackBase::RESUMING) { 2713 track->mState = TrackBase::ACTIVE; 2714 rampVolume = true; 2715 } 2716 } else if (cblk->server != 0) { 2717 // If the track is stopped before the first frame was mixed, 2718 // do not apply ramp 2719 rampVolume = true; 2720 } 2721 // compute volume for this track 2722 float left, right; 2723 if (track->isMuted() || mMasterMute || track->isPausing() || 2724 mStreamTypes[track->type()].mute) { 2725 left = right = 0; 2726 if (track->isPausing()) { 2727 track->setPaused(); 2728 } 2729 } else { 2730 float typeVolume = mStreamTypes[track->type()].volume; 2731 float v = mMasterVolume * typeVolume; 2732 uint32_t vlr = cblk->volumeLR; 2733 float v_clamped = v * (vlr & 0xFFFF); 2734 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2735 left = v_clamped/MAX_GAIN; 2736 v_clamped = v * (vlr >> 16); 2737 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2738 right = v_clamped/MAX_GAIN; 2739 } 2740 2741 if (left != mLeftVolFloat || right != mRightVolFloat) { 2742 mLeftVolFloat = left; 2743 mRightVolFloat = right; 2744 2745 // If audio HAL implements volume control, 2746 // force software volume to nominal value 2747 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2748 left = 1.0f; 2749 right = 1.0f; 2750 } 2751 2752 // Convert volumes from float to 8.24 2753 uint32_t vl = (uint32_t)(left * (1 << 24)); 2754 uint32_t vr = (uint32_t)(right * (1 << 24)); 2755 2756 // Delegate volume control to effect in track effect chain if needed 2757 // only one effect chain can be present on DirectOutputThread, so if 2758 // there is one, the track is connected to it 2759 if (!effectChains.isEmpty()) { 2760 // Do not ramp volume if volume is controlled by effect 2761 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2762 rampVolume = false; 2763 } 2764 } 2765 2766 // Convert volumes from 8.24 to 4.12 format 2767 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2768 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2769 leftVol = (uint16_t)v_clamped; 2770 v_clamped = (vr + (1 << 11)) >> 12; 2771 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2772 rightVol = (uint16_t)v_clamped; 2773 } else { 2774 leftVol = mLeftVolShort; 2775 rightVol = mRightVolShort; 2776 rampVolume = false; 2777 } 2778 2779 // reset retry count 2780 track->mRetryCount = kMaxTrackRetriesDirect; 2781 activeTrack = t; 2782 mixerStatus = MIXER_TRACKS_READY; 2783 } else { 2784 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2785 if (track->isStopped()) { 2786 track->reset(); 2787 } 2788 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2789 // We have consumed all the buffers of this track. 2790 // Remove it from the list of active tracks. 2791 trackToRemove = track; 2792 } else { 2793 // No buffers for this track. Give it a few chances to 2794 // fill a buffer, then remove it from active list. 2795 if (--(track->mRetryCount) <= 0) { 2796 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2797 trackToRemove = track; 2798 } else { 2799 mixerStatus = MIXER_TRACKS_ENABLED; 2800 } 2801 } 2802 } 2803 } 2804 2805 // remove all the tracks that need to be... 2806 if (CC_UNLIKELY(trackToRemove != 0)) { 2807 mActiveTracks.remove(trackToRemove); 2808 if (!effectChains.isEmpty()) { 2809 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2810 trackToRemove->sessionId()); 2811 effectChains[0]->decActiveTrackCnt(); 2812 } 2813 if (trackToRemove->isTerminated()) { 2814 removeTrack_l(trackToRemove); 2815 } 2816 } 2817 2818 lockEffectChains_l(effectChains); 2819 } 2820 2821 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2822 AudioBufferProvider::Buffer buffer; 2823 size_t frameCount = mFrameCount; 2824 curBuf = (int8_t *)mMixBuffer; 2825 // output audio to hardware 2826 while (frameCount) { 2827 buffer.frameCount = frameCount; 2828 activeTrack->getNextBuffer(&buffer); 2829 if (CC_UNLIKELY(buffer.raw == NULL)) { 2830 memset(curBuf, 0, frameCount * mFrameSize); 2831 break; 2832 } 2833 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2834 frameCount -= buffer.frameCount; 2835 curBuf += buffer.frameCount * mFrameSize; 2836 activeTrack->releaseBuffer(&buffer); 2837 } 2838 sleepTime = 0; 2839 standbyTime = systemTime() + standbyDelay; 2840 } else { 2841 if (sleepTime == 0) { 2842 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2843 sleepTime = activeSleepTime; 2844 } else { 2845 sleepTime = idleSleepTime; 2846 } 2847 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2848 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2849 sleepTime = 0; 2850 } 2851 } 2852 2853 if (mSuspended) { 2854 sleepTime = suspendSleepTimeUs(); 2855 } 2856 // sleepTime == 0 means we must write to audio hardware 2857 if (sleepTime == 0) { 2858 if (mixerStatus == MIXER_TRACKS_READY) { 2859 applyVolume(leftVol, rightVol, rampVolume); 2860 } 2861 for (size_t i = 0; i < effectChains.size(); i ++) { 2862 effectChains[i]->process_l(); 2863 } 2864 unlockEffectChains(effectChains); 2865 2866 mLastWriteTime = systemTime(); 2867 mInWrite = true; 2868 mBytesWritten += mixBufferSize; 2869 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2870 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2871 mNumWrites++; 2872 mInWrite = false; 2873 mStandby = false; 2874 } else { 2875 unlockEffectChains(effectChains); 2876 usleep(sleepTime); 2877 } 2878 2879 // finally let go of removed track, without the lock held 2880 // since we can't guarantee the destructors won't acquire that 2881 // same lock. 2882 trackToRemove.clear(); 2883 activeTrack.clear(); 2884 2885 // Effect chains will be actually deleted here if they were removed from 2886 // mEffectChains list during mixing or effects processing 2887 effectChains.clear(); 2888 } 2889 2890 if (!mStandby) { 2891 mOutput->stream->common.standby(&mOutput->stream->common); 2892 } 2893 2894 releaseWakeLock(); 2895 2896 ALOGV("DirectOutputThread %p exiting", this); 2897 return false; 2898} 2899 2900// getTrackName_l() must be called with ThreadBase::mLock held 2901int AudioFlinger::DirectOutputThread::getTrackName_l() 2902{ 2903 return 0; 2904} 2905 2906// deleteTrackName_l() must be called with ThreadBase::mLock held 2907void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2908{ 2909} 2910 2911// checkForNewParameters_l() must be called with ThreadBase::mLock held 2912bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2913{ 2914 bool reconfig = false; 2915 2916 while (!mNewParameters.isEmpty()) { 2917 status_t status = NO_ERROR; 2918 String8 keyValuePair = mNewParameters[0]; 2919 AudioParameter param = AudioParameter(keyValuePair); 2920 int value; 2921 2922 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2923 // do not accept frame count changes if tracks are open as the track buffer 2924 // size depends on frame count and correct behavior would not be garantied 2925 // if frame count is changed after track creation 2926 if (!mTracks.isEmpty()) { 2927 status = INVALID_OPERATION; 2928 } else { 2929 reconfig = true; 2930 } 2931 } 2932 if (status == NO_ERROR) { 2933 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2934 keyValuePair.string()); 2935 if (!mStandby && status == INVALID_OPERATION) { 2936 mOutput->stream->common.standby(&mOutput->stream->common); 2937 mStandby = true; 2938 mBytesWritten = 0; 2939 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2940 keyValuePair.string()); 2941 } 2942 if (status == NO_ERROR && reconfig) { 2943 readOutputParameters(); 2944 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2945 } 2946 } 2947 2948 mNewParameters.removeAt(0); 2949 2950 mParamStatus = status; 2951 mParamCond.signal(); 2952 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2953 // already timed out waiting for the status and will never signal the condition. 2954 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2955 } 2956 return reconfig; 2957} 2958 2959uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2960{ 2961 uint32_t time; 2962 if (audio_is_linear_pcm(mFormat)) { 2963 time = PlaybackThread::activeSleepTimeUs(); 2964 } else { 2965 time = 10000; 2966 } 2967 return time; 2968} 2969 2970uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2971{ 2972 uint32_t time; 2973 if (audio_is_linear_pcm(mFormat)) { 2974 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2975 } else { 2976 time = 10000; 2977 } 2978 return time; 2979} 2980 2981uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2982{ 2983 uint32_t time; 2984 if (audio_is_linear_pcm(mFormat)) { 2985 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2986 } else { 2987 time = 10000; 2988 } 2989 return time; 2990} 2991 2992 2993// ---------------------------------------------------------------------------- 2994 2995AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2996 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2997{ 2998 mType = ThreadBase::DUPLICATING; 2999 addOutputTrack(mainThread); 3000} 3001 3002AudioFlinger::DuplicatingThread::~DuplicatingThread() 3003{ 3004 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3005 mOutputTracks[i]->destroy(); 3006 } 3007 mOutputTracks.clear(); 3008} 3009 3010bool AudioFlinger::DuplicatingThread::threadLoop() 3011{ 3012 Vector< sp<Track> > tracksToRemove; 3013 mixer_state mixerStatus = MIXER_IDLE; 3014 nsecs_t standbyTime = systemTime(); 3015 size_t mixBufferSize = mFrameCount*mFrameSize; 3016 SortedVector< sp<OutputTrack> > outputTracks; 3017 uint32_t writeFrames = 0; 3018 uint32_t activeSleepTime = activeSleepTimeUs(); 3019 uint32_t idleSleepTime = idleSleepTimeUs(); 3020 uint32_t sleepTime = idleSleepTime; 3021 Vector< sp<EffectChain> > effectChains; 3022 3023 acquireWakeLock(); 3024 3025 while (!exitPending()) 3026 { 3027 processConfigEvents(); 3028 3029 mixerStatus = MIXER_IDLE; 3030 { // scope for the mLock 3031 3032 Mutex::Autolock _l(mLock); 3033 3034 if (checkForNewParameters_l()) { 3035 mixBufferSize = mFrameCount*mFrameSize; 3036 updateWaitTime(); 3037 activeSleepTime = activeSleepTimeUs(); 3038 idleSleepTime = idleSleepTimeUs(); 3039 } 3040 3041 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3042 3043 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3044 outputTracks.add(mOutputTracks[i]); 3045 } 3046 3047 // put audio hardware into standby after short delay 3048 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3049 mSuspended)) { 3050 if (!mStandby) { 3051 for (size_t i = 0; i < outputTracks.size(); i++) { 3052 outputTracks[i]->stop(); 3053 } 3054 mStandby = true; 3055 mBytesWritten = 0; 3056 } 3057 3058 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3059 // we're about to wait, flush the binder command buffer 3060 IPCThreadState::self()->flushCommands(); 3061 outputTracks.clear(); 3062 3063 if (exitPending()) break; 3064 3065 releaseWakeLock_l(); 3066 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3067 mWaitWorkCV.wait(mLock); 3068 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3069 acquireWakeLock_l(); 3070 3071 mPrevMixerStatus = MIXER_IDLE; 3072 if (!mMasterMute) { 3073 char value[PROPERTY_VALUE_MAX]; 3074 property_get("ro.audio.silent", value, "0"); 3075 if (atoi(value)) { 3076 ALOGD("Silence is golden"); 3077 setMasterMute(true); 3078 } 3079 } 3080 3081 standbyTime = systemTime() + kStandbyTimeInNsecs; 3082 sleepTime = idleSleepTime; 3083 continue; 3084 } 3085 } 3086 3087 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3088 3089 // prevent any changes in effect chain list and in each effect chain 3090 // during mixing and effect process as the audio buffers could be deleted 3091 // or modified if an effect is created or deleted 3092 lockEffectChains_l(effectChains); 3093 } 3094 3095 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3096 // mix buffers... 3097 if (outputsReady(outputTracks)) { 3098 mAudioMixer->process(); 3099 } else { 3100 memset(mMixBuffer, 0, mixBufferSize); 3101 } 3102 sleepTime = 0; 3103 writeFrames = mFrameCount; 3104 } else { 3105 if (sleepTime == 0) { 3106 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3107 sleepTime = activeSleepTime; 3108 } else { 3109 sleepTime = idleSleepTime; 3110 } 3111 } else if (mBytesWritten != 0) { 3112 // flush remaining overflow buffers in output tracks 3113 for (size_t i = 0; i < outputTracks.size(); i++) { 3114 if (outputTracks[i]->isActive()) { 3115 sleepTime = 0; 3116 writeFrames = 0; 3117 memset(mMixBuffer, 0, mixBufferSize); 3118 break; 3119 } 3120 } 3121 } 3122 } 3123 3124 if (mSuspended) { 3125 sleepTime = suspendSleepTimeUs(); 3126 } 3127 // sleepTime == 0 means we must write to audio hardware 3128 if (sleepTime == 0) { 3129 for (size_t i = 0; i < effectChains.size(); i ++) { 3130 effectChains[i]->process_l(); 3131 } 3132 // enable changes in effect chain 3133 unlockEffectChains(effectChains); 3134 3135 standbyTime = systemTime() + kStandbyTimeInNsecs; 3136 for (size_t i = 0; i < outputTracks.size(); i++) { 3137 outputTracks[i]->write(mMixBuffer, writeFrames); 3138 } 3139 mStandby = false; 3140 mBytesWritten += mixBufferSize; 3141 } else { 3142 // enable changes in effect chain 3143 unlockEffectChains(effectChains); 3144 usleep(sleepTime); 3145 } 3146 3147 // finally let go of all our tracks, without the lock held 3148 // since we can't guarantee the destructors won't acquire that 3149 // same lock. 3150 tracksToRemove.clear(); 3151 outputTracks.clear(); 3152 3153 // Effect chains will be actually deleted here if they were removed from 3154 // mEffectChains list during mixing or effects processing 3155 effectChains.clear(); 3156 } 3157 3158 releaseWakeLock(); 3159 3160 return false; 3161} 3162 3163void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3164{ 3165 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3166 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3167 this, 3168 mSampleRate, 3169 mFormat, 3170 mChannelMask, 3171 frameCount); 3172 if (outputTrack->cblk() != NULL) { 3173 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3174 mOutputTracks.add(outputTrack); 3175 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3176 updateWaitTime(); 3177 } 3178} 3179 3180void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3181{ 3182 Mutex::Autolock _l(mLock); 3183 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3184 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3185 mOutputTracks[i]->destroy(); 3186 mOutputTracks.removeAt(i); 3187 updateWaitTime(); 3188 return; 3189 } 3190 } 3191 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3192} 3193 3194void AudioFlinger::DuplicatingThread::updateWaitTime() 3195{ 3196 mWaitTimeMs = UINT_MAX; 3197 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3198 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3199 if (strong != NULL) { 3200 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3201 if (waitTimeMs < mWaitTimeMs) { 3202 mWaitTimeMs = waitTimeMs; 3203 } 3204 } 3205 } 3206} 3207 3208 3209bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3210{ 3211 for (size_t i = 0; i < outputTracks.size(); i++) { 3212 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3213 if (thread == 0) { 3214 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3215 return false; 3216 } 3217 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3218 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3219 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3220 return false; 3221 } 3222 } 3223 return true; 3224} 3225 3226uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3227{ 3228 return (mWaitTimeMs * 1000) / 2; 3229} 3230 3231// ---------------------------------------------------------------------------- 3232 3233// TrackBase constructor must be called with AudioFlinger::mLock held 3234AudioFlinger::ThreadBase::TrackBase::TrackBase( 3235 const wp<ThreadBase>& thread, 3236 const sp<Client>& client, 3237 uint32_t sampleRate, 3238 audio_format_t format, 3239 uint32_t channelMask, 3240 int frameCount, 3241 uint32_t flags, 3242 const sp<IMemory>& sharedBuffer, 3243 int sessionId) 3244 : RefBase(), 3245 mThread(thread), 3246 mClient(client), 3247 mCblk(0), 3248 mFrameCount(0), 3249 mState(IDLE), 3250 mClientTid(-1), 3251 mFormat(format), 3252 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3253 mSessionId(sessionId) 3254{ 3255 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3256 3257 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3258 size_t size = sizeof(audio_track_cblk_t); 3259 uint8_t channelCount = popcount(channelMask); 3260 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3261 if (sharedBuffer == 0) { 3262 size += bufferSize; 3263 } 3264 3265 if (client != NULL) { 3266 mCblkMemory = client->heap()->allocate(size); 3267 if (mCblkMemory != 0) { 3268 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3269 if (mCblk) { // construct the shared structure in-place. 3270 new(mCblk) audio_track_cblk_t(); 3271 // clear all buffers 3272 mCblk->frameCount = frameCount; 3273 mCblk->sampleRate = sampleRate; 3274 mChannelCount = channelCount; 3275 mChannelMask = channelMask; 3276 if (sharedBuffer == 0) { 3277 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3278 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3279 // Force underrun condition to avoid false underrun callback until first data is 3280 // written to buffer (other flags are cleared) 3281 mCblk->flags = CBLK_UNDERRUN_ON; 3282 } else { 3283 mBuffer = sharedBuffer->pointer(); 3284 } 3285 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3286 } 3287 } else { 3288 ALOGE("not enough memory for AudioTrack size=%u", size); 3289 client->heap()->dump("AudioTrack"); 3290 return; 3291 } 3292 } else { 3293 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3294 // construct the shared structure in-place. 3295 new(mCblk) audio_track_cblk_t(); 3296 // clear all buffers 3297 mCblk->frameCount = frameCount; 3298 mCblk->sampleRate = sampleRate; 3299 mChannelCount = channelCount; 3300 mChannelMask = channelMask; 3301 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3302 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3303 // Force underrun condition to avoid false underrun callback until first data is 3304 // written to buffer (other flags are cleared) 3305 mCblk->flags = CBLK_UNDERRUN_ON; 3306 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3307 } 3308} 3309 3310AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3311{ 3312 if (mCblk) { 3313 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3314 if (mClient == NULL) { 3315 delete mCblk; 3316 } 3317 } 3318 mCblkMemory.clear(); // and free the shared memory 3319 if (mClient != NULL) { 3320 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3321 mClient.clear(); 3322 } 3323} 3324 3325void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3326{ 3327 buffer->raw = NULL; 3328 mFrameCount = buffer->frameCount; 3329 step(); 3330 buffer->frameCount = 0; 3331} 3332 3333bool AudioFlinger::ThreadBase::TrackBase::step() { 3334 bool result; 3335 audio_track_cblk_t* cblk = this->cblk(); 3336 3337 result = cblk->stepServer(mFrameCount); 3338 if (!result) { 3339 ALOGV("stepServer failed acquiring cblk mutex"); 3340 mFlags |= STEPSERVER_FAILED; 3341 } 3342 return result; 3343} 3344 3345void AudioFlinger::ThreadBase::TrackBase::reset() { 3346 audio_track_cblk_t* cblk = this->cblk(); 3347 3348 cblk->user = 0; 3349 cblk->server = 0; 3350 cblk->userBase = 0; 3351 cblk->serverBase = 0; 3352 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3353 ALOGV("TrackBase::reset"); 3354} 3355 3356sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3357{ 3358 return mCblkMemory; 3359} 3360 3361int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3362 return (int)mCblk->sampleRate; 3363} 3364 3365int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3366 return (const int)mChannelCount; 3367} 3368 3369uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3370 return mChannelMask; 3371} 3372 3373void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3374 audio_track_cblk_t* cblk = this->cblk(); 3375 size_t frameSize = cblk->frameSize; 3376 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3377 int8_t *bufferEnd = bufferStart + frames * frameSize; 3378 3379 // Check validity of returned pointer in case the track control block would have been corrupted. 3380 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3381 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3382 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3383 server %d, serverBase %d, user %d, userBase %d", 3384 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3385 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3386 return 0; 3387 } 3388 3389 return bufferStart; 3390} 3391 3392// ---------------------------------------------------------------------------- 3393 3394// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3395AudioFlinger::PlaybackThread::Track::Track( 3396 const wp<ThreadBase>& thread, 3397 const sp<Client>& client, 3398 audio_stream_type_t streamType, 3399 uint32_t sampleRate, 3400 audio_format_t format, 3401 uint32_t channelMask, 3402 int frameCount, 3403 const sp<IMemory>& sharedBuffer, 3404 int sessionId) 3405 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3406 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3407 mAuxEffectId(0), mHasVolumeController(false) 3408{ 3409 if (mCblk != NULL) { 3410 sp<ThreadBase> baseThread = thread.promote(); 3411 if (baseThread != 0) { 3412 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3413 mName = playbackThread->getTrackName_l(); 3414 mMainBuffer = playbackThread->mixBuffer(); 3415 } 3416 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3417 if (mName < 0) { 3418 ALOGE("no more track names available"); 3419 } 3420 mStreamType = streamType; 3421 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3422 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3423 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3424 } 3425} 3426 3427AudioFlinger::PlaybackThread::Track::~Track() 3428{ 3429 ALOGV("PlaybackThread::Track destructor"); 3430 sp<ThreadBase> thread = mThread.promote(); 3431 if (thread != 0) { 3432 Mutex::Autolock _l(thread->mLock); 3433 mState = TERMINATED; 3434 } 3435} 3436 3437void AudioFlinger::PlaybackThread::Track::destroy() 3438{ 3439 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3440 // by removing it from mTracks vector, so there is a risk that this Tracks's 3441 // desctructor is called. As the destructor needs to lock mLock, 3442 // we must acquire a strong reference on this Track before locking mLock 3443 // here so that the destructor is called only when exiting this function. 3444 // On the other hand, as long as Track::destroy() is only called by 3445 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3446 // this Track with its member mTrack. 3447 sp<Track> keep(this); 3448 { // scope for mLock 3449 sp<ThreadBase> thread = mThread.promote(); 3450 if (thread != 0) { 3451 if (!isOutputTrack()) { 3452 if (mState == ACTIVE || mState == RESUMING) { 3453 AudioSystem::stopOutput(thread->id(), 3454 (audio_stream_type_t)mStreamType, 3455 mSessionId); 3456 3457 // to track the speaker usage 3458 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3459 } 3460 AudioSystem::releaseOutput(thread->id()); 3461 } 3462 Mutex::Autolock _l(thread->mLock); 3463 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3464 playbackThread->destroyTrack_l(this); 3465 } 3466 } 3467} 3468 3469void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3470{ 3471 uint32_t vlr = mCblk->volumeLR; 3472 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", 3473 mName - AudioMixer::TRACK0, 3474 (mClient == NULL) ? getpid() : mClient->pid(), 3475 mStreamType, 3476 mFormat, 3477 mChannelMask, 3478 mSessionId, 3479 mFrameCount, 3480 mState, 3481 mMute, 3482 mFillingUpStatus, 3483 mCblk->sampleRate, 3484 vlr & 0xFFFF, 3485 vlr >> 16, 3486 mCblk->server, 3487 mCblk->user, 3488 (int)mMainBuffer, 3489 (int)mAuxBuffer); 3490} 3491 3492status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3493{ 3494 audio_track_cblk_t* cblk = this->cblk(); 3495 uint32_t framesReady; 3496 uint32_t framesReq = buffer->frameCount; 3497 3498 // Check if last stepServer failed, try to step now 3499 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3500 if (!step()) goto getNextBuffer_exit; 3501 ALOGV("stepServer recovered"); 3502 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3503 } 3504 3505 framesReady = cblk->framesReady(); 3506 3507 if (CC_LIKELY(framesReady)) { 3508 uint32_t s = cblk->server; 3509 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3510 3511 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3512 if (framesReq > framesReady) { 3513 framesReq = framesReady; 3514 } 3515 if (s + framesReq > bufferEnd) { 3516 framesReq = bufferEnd - s; 3517 } 3518 3519 buffer->raw = getBuffer(s, framesReq); 3520 if (buffer->raw == NULL) goto getNextBuffer_exit; 3521 3522 buffer->frameCount = framesReq; 3523 return NO_ERROR; 3524 } 3525 3526getNextBuffer_exit: 3527 buffer->raw = NULL; 3528 buffer->frameCount = 0; 3529 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3530 return NOT_ENOUGH_DATA; 3531} 3532 3533bool AudioFlinger::PlaybackThread::Track::isReady() const { 3534 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3535 3536 if (mCblk->framesReady() >= mCblk->frameCount || 3537 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3538 mFillingUpStatus = FS_FILLED; 3539 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3540 return true; 3541 } 3542 return false; 3543} 3544 3545status_t AudioFlinger::PlaybackThread::Track::start() 3546{ 3547 status_t status = NO_ERROR; 3548 ALOGV("start(%d), calling thread %d session %d", 3549 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3550 sp<ThreadBase> thread = mThread.promote(); 3551 if (thread != 0) { 3552 Mutex::Autolock _l(thread->mLock); 3553 track_state state = mState; 3554 // here the track could be either new, or restarted 3555 // in both cases "unstop" the track 3556 if (mState == PAUSED) { 3557 mState = TrackBase::RESUMING; 3558 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3559 } else { 3560 mState = TrackBase::ACTIVE; 3561 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3562 } 3563 3564 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3565 thread->mLock.unlock(); 3566 status = AudioSystem::startOutput(thread->id(), 3567 (audio_stream_type_t)mStreamType, 3568 mSessionId); 3569 thread->mLock.lock(); 3570 3571 // to track the speaker usage 3572 if (status == NO_ERROR) { 3573 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3574 } 3575 } 3576 if (status == NO_ERROR) { 3577 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3578 playbackThread->addTrack_l(this); 3579 } else { 3580 mState = state; 3581 } 3582 } else { 3583 status = BAD_VALUE; 3584 } 3585 return status; 3586} 3587 3588void AudioFlinger::PlaybackThread::Track::stop() 3589{ 3590 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3591 sp<ThreadBase> thread = mThread.promote(); 3592 if (thread != 0) { 3593 Mutex::Autolock _l(thread->mLock); 3594 track_state state = mState; 3595 if (mState > STOPPED) { 3596 mState = STOPPED; 3597 // If the track is not active (PAUSED and buffers full), flush buffers 3598 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3599 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3600 reset(); 3601 } 3602 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3603 } 3604 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3605 thread->mLock.unlock(); 3606 AudioSystem::stopOutput(thread->id(), 3607 (audio_stream_type_t)mStreamType, 3608 mSessionId); 3609 thread->mLock.lock(); 3610 3611 // to track the speaker usage 3612 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3613 } 3614 } 3615} 3616 3617void AudioFlinger::PlaybackThread::Track::pause() 3618{ 3619 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3620 sp<ThreadBase> thread = mThread.promote(); 3621 if (thread != 0) { 3622 Mutex::Autolock _l(thread->mLock); 3623 if (mState == ACTIVE || mState == RESUMING) { 3624 mState = PAUSING; 3625 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3626 if (!isOutputTrack()) { 3627 thread->mLock.unlock(); 3628 AudioSystem::stopOutput(thread->id(), 3629 (audio_stream_type_t)mStreamType, 3630 mSessionId); 3631 thread->mLock.lock(); 3632 3633 // to track the speaker usage 3634 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3635 } 3636 } 3637 } 3638} 3639 3640void AudioFlinger::PlaybackThread::Track::flush() 3641{ 3642 ALOGV("flush(%d)", mName); 3643 sp<ThreadBase> thread = mThread.promote(); 3644 if (thread != 0) { 3645 Mutex::Autolock _l(thread->mLock); 3646 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3647 return; 3648 } 3649 // No point remaining in PAUSED state after a flush => go to 3650 // STOPPED state 3651 mState = STOPPED; 3652 3653 // do not reset the track if it is still in the process of being stopped or paused. 3654 // this will be done by prepareTracks_l() when the track is stopped. 3655 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3656 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3657 reset(); 3658 } 3659 } 3660} 3661 3662void AudioFlinger::PlaybackThread::Track::reset() 3663{ 3664 // Do not reset twice to avoid discarding data written just after a flush and before 3665 // the audioflinger thread detects the track is stopped. 3666 if (!mResetDone) { 3667 TrackBase::reset(); 3668 // Force underrun condition to avoid false underrun callback until first data is 3669 // written to buffer 3670 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3671 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3672 mFillingUpStatus = FS_FILLING; 3673 mResetDone = true; 3674 } 3675} 3676 3677void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3678{ 3679 mMute = muted; 3680} 3681 3682status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3683{ 3684 status_t status = DEAD_OBJECT; 3685 sp<ThreadBase> thread = mThread.promote(); 3686 if (thread != 0) { 3687 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3688 status = playbackThread->attachAuxEffect(this, EffectId); 3689 } 3690 return status; 3691} 3692 3693void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3694{ 3695 mAuxEffectId = EffectId; 3696 mAuxBuffer = buffer; 3697} 3698 3699// ---------------------------------------------------------------------------- 3700 3701// RecordTrack constructor must be called with AudioFlinger::mLock held 3702AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3703 const wp<ThreadBase>& thread, 3704 const sp<Client>& client, 3705 uint32_t sampleRate, 3706 audio_format_t format, 3707 uint32_t channelMask, 3708 int frameCount, 3709 uint32_t flags, 3710 int sessionId) 3711 : TrackBase(thread, client, sampleRate, format, 3712 channelMask, frameCount, flags, 0, sessionId), 3713 mOverflow(false) 3714{ 3715 if (mCblk != NULL) { 3716 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3717 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3718 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3719 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3720 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3721 } else { 3722 mCblk->frameSize = sizeof(int8_t); 3723 } 3724 } 3725} 3726 3727AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3728{ 3729 sp<ThreadBase> thread = mThread.promote(); 3730 if (thread != 0) { 3731 AudioSystem::releaseInput(thread->id()); 3732 } 3733} 3734 3735status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3736{ 3737 audio_track_cblk_t* cblk = this->cblk(); 3738 uint32_t framesAvail; 3739 uint32_t framesReq = buffer->frameCount; 3740 3741 // Check if last stepServer failed, try to step now 3742 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3743 if (!step()) goto getNextBuffer_exit; 3744 ALOGV("stepServer recovered"); 3745 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3746 } 3747 3748 framesAvail = cblk->framesAvailable_l(); 3749 3750 if (CC_LIKELY(framesAvail)) { 3751 uint32_t s = cblk->server; 3752 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3753 3754 if (framesReq > framesAvail) { 3755 framesReq = framesAvail; 3756 } 3757 if (s + framesReq > bufferEnd) { 3758 framesReq = bufferEnd - s; 3759 } 3760 3761 buffer->raw = getBuffer(s, framesReq); 3762 if (buffer->raw == NULL) goto getNextBuffer_exit; 3763 3764 buffer->frameCount = framesReq; 3765 return NO_ERROR; 3766 } 3767 3768getNextBuffer_exit: 3769 buffer->raw = NULL; 3770 buffer->frameCount = 0; 3771 return NOT_ENOUGH_DATA; 3772} 3773 3774status_t AudioFlinger::RecordThread::RecordTrack::start() 3775{ 3776 sp<ThreadBase> thread = mThread.promote(); 3777 if (thread != 0) { 3778 RecordThread *recordThread = (RecordThread *)thread.get(); 3779 return recordThread->start(this); 3780 } else { 3781 return BAD_VALUE; 3782 } 3783} 3784 3785void AudioFlinger::RecordThread::RecordTrack::stop() 3786{ 3787 sp<ThreadBase> thread = mThread.promote(); 3788 if (thread != 0) { 3789 RecordThread *recordThread = (RecordThread *)thread.get(); 3790 recordThread->stop(this); 3791 TrackBase::reset(); 3792 // Force overerrun condition to avoid false overrun callback until first data is 3793 // read from buffer 3794 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3795 } 3796} 3797 3798void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3799{ 3800 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3801 (mClient == NULL) ? getpid() : mClient->pid(), 3802 mFormat, 3803 mChannelMask, 3804 mSessionId, 3805 mFrameCount, 3806 mState, 3807 mCblk->sampleRate, 3808 mCblk->server, 3809 mCblk->user); 3810} 3811 3812 3813// ---------------------------------------------------------------------------- 3814 3815AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3816 const wp<ThreadBase>& thread, 3817 DuplicatingThread *sourceThread, 3818 uint32_t sampleRate, 3819 audio_format_t format, 3820 uint32_t channelMask, 3821 int frameCount) 3822 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3823 mActive(false), mSourceThread(sourceThread) 3824{ 3825 3826 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3827 if (mCblk != NULL) { 3828 mCblk->flags |= CBLK_DIRECTION_OUT; 3829 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3830 mCblk->volumeLR = (MAX_GAIN_INT << 16) | MAX_GAIN_INT; 3831 mOutBuffer.frameCount = 0; 3832 playbackThread->mTracks.add(this); 3833 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3834 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3835 mCblk, mBuffer, mCblk->buffers, 3836 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3837 } else { 3838 ALOGW("Error creating output track on thread %p", playbackThread); 3839 } 3840} 3841 3842AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3843{ 3844 clearBufferQueue(); 3845} 3846 3847status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3848{ 3849 status_t status = Track::start(); 3850 if (status != NO_ERROR) { 3851 return status; 3852 } 3853 3854 mActive = true; 3855 mRetryCount = 127; 3856 return status; 3857} 3858 3859void AudioFlinger::PlaybackThread::OutputTrack::stop() 3860{ 3861 Track::stop(); 3862 clearBufferQueue(); 3863 mOutBuffer.frameCount = 0; 3864 mActive = false; 3865} 3866 3867bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3868{ 3869 Buffer *pInBuffer; 3870 Buffer inBuffer; 3871 uint32_t channelCount = mChannelCount; 3872 bool outputBufferFull = false; 3873 inBuffer.frameCount = frames; 3874 inBuffer.i16 = data; 3875 3876 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3877 3878 if (!mActive && frames != 0) { 3879 start(); 3880 sp<ThreadBase> thread = mThread.promote(); 3881 if (thread != 0) { 3882 MixerThread *mixerThread = (MixerThread *)thread.get(); 3883 if (mCblk->frameCount > frames){ 3884 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3885 uint32_t startFrames = (mCblk->frameCount - frames); 3886 pInBuffer = new Buffer; 3887 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3888 pInBuffer->frameCount = startFrames; 3889 pInBuffer->i16 = pInBuffer->mBuffer; 3890 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3891 mBufferQueue.add(pInBuffer); 3892 } else { 3893 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3894 } 3895 } 3896 } 3897 } 3898 3899 while (waitTimeLeftMs) { 3900 // First write pending buffers, then new data 3901 if (mBufferQueue.size()) { 3902 pInBuffer = mBufferQueue.itemAt(0); 3903 } else { 3904 pInBuffer = &inBuffer; 3905 } 3906 3907 if (pInBuffer->frameCount == 0) { 3908 break; 3909 } 3910 3911 if (mOutBuffer.frameCount == 0) { 3912 mOutBuffer.frameCount = pInBuffer->frameCount; 3913 nsecs_t startTime = systemTime(); 3914 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3915 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3916 outputBufferFull = true; 3917 break; 3918 } 3919 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3920 if (waitTimeLeftMs >= waitTimeMs) { 3921 waitTimeLeftMs -= waitTimeMs; 3922 } else { 3923 waitTimeLeftMs = 0; 3924 } 3925 } 3926 3927 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3928 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3929 mCblk->stepUser(outFrames); 3930 pInBuffer->frameCount -= outFrames; 3931 pInBuffer->i16 += outFrames * channelCount; 3932 mOutBuffer.frameCount -= outFrames; 3933 mOutBuffer.i16 += outFrames * channelCount; 3934 3935 if (pInBuffer->frameCount == 0) { 3936 if (mBufferQueue.size()) { 3937 mBufferQueue.removeAt(0); 3938 delete [] pInBuffer->mBuffer; 3939 delete pInBuffer; 3940 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3941 } else { 3942 break; 3943 } 3944 } 3945 } 3946 3947 // If we could not write all frames, allocate a buffer and queue it for next time. 3948 if (inBuffer.frameCount) { 3949 sp<ThreadBase> thread = mThread.promote(); 3950 if (thread != 0 && !thread->standby()) { 3951 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3952 pInBuffer = new Buffer; 3953 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3954 pInBuffer->frameCount = inBuffer.frameCount; 3955 pInBuffer->i16 = pInBuffer->mBuffer; 3956 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3957 mBufferQueue.add(pInBuffer); 3958 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3959 } else { 3960 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3961 } 3962 } 3963 } 3964 3965 // Calling write() with a 0 length buffer, means that no more data will be written: 3966 // If no more buffers are pending, fill output track buffer to make sure it is started 3967 // by output mixer. 3968 if (frames == 0 && mBufferQueue.size() == 0) { 3969 if (mCblk->user < mCblk->frameCount) { 3970 frames = mCblk->frameCount - mCblk->user; 3971 pInBuffer = new Buffer; 3972 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3973 pInBuffer->frameCount = frames; 3974 pInBuffer->i16 = pInBuffer->mBuffer; 3975 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3976 mBufferQueue.add(pInBuffer); 3977 } else if (mActive) { 3978 stop(); 3979 } 3980 } 3981 3982 return outputBufferFull; 3983} 3984 3985status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3986{ 3987 int active; 3988 status_t result; 3989 audio_track_cblk_t* cblk = mCblk; 3990 uint32_t framesReq = buffer->frameCount; 3991 3992// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3993 buffer->frameCount = 0; 3994 3995 uint32_t framesAvail = cblk->framesAvailable(); 3996 3997 3998 if (framesAvail == 0) { 3999 Mutex::Autolock _l(cblk->lock); 4000 goto start_loop_here; 4001 while (framesAvail == 0) { 4002 active = mActive; 4003 if (CC_UNLIKELY(!active)) { 4004 ALOGV("Not active and NO_MORE_BUFFERS"); 4005 return NO_MORE_BUFFERS; 4006 } 4007 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4008 if (result != NO_ERROR) { 4009 return NO_MORE_BUFFERS; 4010 } 4011 // read the server count again 4012 start_loop_here: 4013 framesAvail = cblk->framesAvailable_l(); 4014 } 4015 } 4016 4017// if (framesAvail < framesReq) { 4018// return NO_MORE_BUFFERS; 4019// } 4020 4021 if (framesReq > framesAvail) { 4022 framesReq = framesAvail; 4023 } 4024 4025 uint32_t u = cblk->user; 4026 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4027 4028 if (u + framesReq > bufferEnd) { 4029 framesReq = bufferEnd - u; 4030 } 4031 4032 buffer->frameCount = framesReq; 4033 buffer->raw = (void *)cblk->buffer(u); 4034 return NO_ERROR; 4035} 4036 4037 4038void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4039{ 4040 size_t size = mBufferQueue.size(); 4041 Buffer *pBuffer; 4042 4043 for (size_t i = 0; i < size; i++) { 4044 pBuffer = mBufferQueue.itemAt(i); 4045 delete [] pBuffer->mBuffer; 4046 delete pBuffer; 4047 } 4048 mBufferQueue.clear(); 4049} 4050 4051// ---------------------------------------------------------------------------- 4052 4053AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4054 : RefBase(), 4055 mAudioFlinger(audioFlinger), 4056 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4057 mPid(pid) 4058{ 4059 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4060} 4061 4062// Client destructor must be called with AudioFlinger::mLock held 4063AudioFlinger::Client::~Client() 4064{ 4065 mAudioFlinger->removeClient_l(mPid); 4066} 4067 4068const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4069{ 4070 return mMemoryDealer; 4071} 4072 4073// ---------------------------------------------------------------------------- 4074 4075AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4076 const sp<IAudioFlingerClient>& client, 4077 pid_t pid) 4078 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4079{ 4080} 4081 4082AudioFlinger::NotificationClient::~NotificationClient() 4083{ 4084 mClient.clear(); 4085} 4086 4087void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4088{ 4089 sp<NotificationClient> keep(this); 4090 { 4091 mAudioFlinger->removeNotificationClient(mPid); 4092 } 4093} 4094 4095// ---------------------------------------------------------------------------- 4096 4097AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4098 : BnAudioTrack(), 4099 mTrack(track) 4100{ 4101} 4102 4103AudioFlinger::TrackHandle::~TrackHandle() { 4104 // just stop the track on deletion, associated resources 4105 // will be freed from the main thread once all pending buffers have 4106 // been played. Unless it's not in the active track list, in which 4107 // case we free everything now... 4108 mTrack->destroy(); 4109} 4110 4111sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4112 return mTrack->getCblk(); 4113} 4114 4115status_t AudioFlinger::TrackHandle::start() { 4116 return mTrack->start(); 4117} 4118 4119void AudioFlinger::TrackHandle::stop() { 4120 mTrack->stop(); 4121} 4122 4123void AudioFlinger::TrackHandle::flush() { 4124 mTrack->flush(); 4125} 4126 4127void AudioFlinger::TrackHandle::mute(bool e) { 4128 mTrack->mute(e); 4129} 4130 4131void AudioFlinger::TrackHandle::pause() { 4132 mTrack->pause(); 4133} 4134 4135status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4136{ 4137 return mTrack->attachAuxEffect(EffectId); 4138} 4139 4140status_t AudioFlinger::TrackHandle::onTransact( 4141 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4142{ 4143 return BnAudioTrack::onTransact(code, data, reply, flags); 4144} 4145 4146// ---------------------------------------------------------------------------- 4147 4148sp<IAudioRecord> AudioFlinger::openRecord( 4149 pid_t pid, 4150 int input, 4151 uint32_t sampleRate, 4152 audio_format_t format, 4153 uint32_t channelMask, 4154 int frameCount, 4155 uint32_t flags, 4156 int *sessionId, 4157 status_t *status) 4158{ 4159 sp<RecordThread::RecordTrack> recordTrack; 4160 sp<RecordHandle> recordHandle; 4161 sp<Client> client; 4162 wp<Client> wclient; 4163 status_t lStatus; 4164 RecordThread *thread; 4165 size_t inFrameCount; 4166 int lSessionId; 4167 4168 // check calling permissions 4169 if (!recordingAllowed()) { 4170 lStatus = PERMISSION_DENIED; 4171 goto Exit; 4172 } 4173 4174 // add client to list 4175 { // scope for mLock 4176 Mutex::Autolock _l(mLock); 4177 thread = checkRecordThread_l(input); 4178 if (thread == NULL) { 4179 lStatus = BAD_VALUE; 4180 goto Exit; 4181 } 4182 4183 wclient = mClients.valueFor(pid); 4184 if (wclient != NULL) { 4185 client = wclient.promote(); 4186 } else { 4187 client = new Client(this, pid); 4188 mClients.add(pid, client); 4189 } 4190 4191 // If no audio session id is provided, create one here 4192 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4193 lSessionId = *sessionId; 4194 } else { 4195 lSessionId = nextUniqueId(); 4196 if (sessionId != NULL) { 4197 *sessionId = lSessionId; 4198 } 4199 } 4200 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4201 recordTrack = thread->createRecordTrack_l(client, 4202 sampleRate, 4203 format, 4204 channelMask, 4205 frameCount, 4206 flags, 4207 lSessionId, 4208 &lStatus); 4209 } 4210 if (lStatus != NO_ERROR) { 4211 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4212 // destructor is called by the TrackBase destructor with mLock held 4213 client.clear(); 4214 recordTrack.clear(); 4215 goto Exit; 4216 } 4217 4218 // return to handle to client 4219 recordHandle = new RecordHandle(recordTrack); 4220 lStatus = NO_ERROR; 4221 4222Exit: 4223 if (status) { 4224 *status = lStatus; 4225 } 4226 return recordHandle; 4227} 4228 4229// ---------------------------------------------------------------------------- 4230 4231AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4232 : BnAudioRecord(), 4233 mRecordTrack(recordTrack) 4234{ 4235} 4236 4237AudioFlinger::RecordHandle::~RecordHandle() { 4238 stop(); 4239} 4240 4241sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4242 return mRecordTrack->getCblk(); 4243} 4244 4245status_t AudioFlinger::RecordHandle::start() { 4246 ALOGV("RecordHandle::start()"); 4247 return mRecordTrack->start(); 4248} 4249 4250void AudioFlinger::RecordHandle::stop() { 4251 ALOGV("RecordHandle::stop()"); 4252 mRecordTrack->stop(); 4253} 4254 4255status_t AudioFlinger::RecordHandle::onTransact( 4256 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4257{ 4258 return BnAudioRecord::onTransact(code, data, reply, flags); 4259} 4260 4261// ---------------------------------------------------------------------------- 4262 4263AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4264 AudioStreamIn *input, 4265 uint32_t sampleRate, 4266 uint32_t channels, 4267 int id, 4268 uint32_t device) : 4269 ThreadBase(audioFlinger, id, device), 4270 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4271{ 4272 mType = ThreadBase::RECORD; 4273 4274 snprintf(mName, kNameLength, "AudioIn_%d", id); 4275 4276 mReqChannelCount = popcount(channels); 4277 mReqSampleRate = sampleRate; 4278 readInputParameters(); 4279} 4280 4281 4282AudioFlinger::RecordThread::~RecordThread() 4283{ 4284 delete[] mRsmpInBuffer; 4285 if (mResampler != NULL) { 4286 delete mResampler; 4287 delete[] mRsmpOutBuffer; 4288 } 4289} 4290 4291void AudioFlinger::RecordThread::onFirstRef() 4292{ 4293 run(mName, PRIORITY_URGENT_AUDIO); 4294} 4295 4296status_t AudioFlinger::RecordThread::readyToRun() 4297{ 4298 status_t status = initCheck(); 4299 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4300 return status; 4301} 4302 4303bool AudioFlinger::RecordThread::threadLoop() 4304{ 4305 AudioBufferProvider::Buffer buffer; 4306 sp<RecordTrack> activeTrack; 4307 Vector< sp<EffectChain> > effectChains; 4308 4309 nsecs_t lastWarning = 0; 4310 4311 acquireWakeLock(); 4312 4313 // start recording 4314 while (!exitPending()) { 4315 4316 processConfigEvents(); 4317 4318 { // scope for mLock 4319 Mutex::Autolock _l(mLock); 4320 checkForNewParameters_l(); 4321 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4322 if (!mStandby) { 4323 mInput->stream->common.standby(&mInput->stream->common); 4324 mStandby = true; 4325 } 4326 4327 if (exitPending()) break; 4328 4329 releaseWakeLock_l(); 4330 ALOGV("RecordThread: loop stopping"); 4331 // go to sleep 4332 mWaitWorkCV.wait(mLock); 4333 ALOGV("RecordThread: loop starting"); 4334 acquireWakeLock_l(); 4335 continue; 4336 } 4337 if (mActiveTrack != 0) { 4338 if (mActiveTrack->mState == TrackBase::PAUSING) { 4339 if (!mStandby) { 4340 mInput->stream->common.standby(&mInput->stream->common); 4341 mStandby = true; 4342 } 4343 mActiveTrack.clear(); 4344 mStartStopCond.broadcast(); 4345 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4346 if (mReqChannelCount != mActiveTrack->channelCount()) { 4347 mActiveTrack.clear(); 4348 mStartStopCond.broadcast(); 4349 } else if (mBytesRead != 0) { 4350 // record start succeeds only if first read from audio input 4351 // succeeds 4352 if (mBytesRead > 0) { 4353 mActiveTrack->mState = TrackBase::ACTIVE; 4354 } else { 4355 mActiveTrack.clear(); 4356 } 4357 mStartStopCond.broadcast(); 4358 } 4359 mStandby = false; 4360 } 4361 } 4362 lockEffectChains_l(effectChains); 4363 } 4364 4365 if (mActiveTrack != 0) { 4366 if (mActiveTrack->mState != TrackBase::ACTIVE && 4367 mActiveTrack->mState != TrackBase::RESUMING) { 4368 unlockEffectChains(effectChains); 4369 usleep(kRecordThreadSleepUs); 4370 continue; 4371 } 4372 for (size_t i = 0; i < effectChains.size(); i ++) { 4373 effectChains[i]->process_l(); 4374 } 4375 4376 buffer.frameCount = mFrameCount; 4377 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4378 size_t framesOut = buffer.frameCount; 4379 if (mResampler == NULL) { 4380 // no resampling 4381 while (framesOut) { 4382 size_t framesIn = mFrameCount - mRsmpInIndex; 4383 if (framesIn) { 4384 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4385 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4386 if (framesIn > framesOut) 4387 framesIn = framesOut; 4388 mRsmpInIndex += framesIn; 4389 framesOut -= framesIn; 4390 if ((int)mChannelCount == mReqChannelCount || 4391 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4392 memcpy(dst, src, framesIn * mFrameSize); 4393 } else { 4394 int16_t *src16 = (int16_t *)src; 4395 int16_t *dst16 = (int16_t *)dst; 4396 if (mChannelCount == 1) { 4397 while (framesIn--) { 4398 *dst16++ = *src16; 4399 *dst16++ = *src16++; 4400 } 4401 } else { 4402 while (framesIn--) { 4403 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4404 src16 += 2; 4405 } 4406 } 4407 } 4408 } 4409 if (framesOut && mFrameCount == mRsmpInIndex) { 4410 if (framesOut == mFrameCount && 4411 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4412 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4413 framesOut = 0; 4414 } else { 4415 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4416 mRsmpInIndex = 0; 4417 } 4418 if (mBytesRead < 0) { 4419 ALOGE("Error reading audio input"); 4420 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4421 // Force input into standby so that it tries to 4422 // recover at next read attempt 4423 mInput->stream->common.standby(&mInput->stream->common); 4424 usleep(kRecordThreadSleepUs); 4425 } 4426 mRsmpInIndex = mFrameCount; 4427 framesOut = 0; 4428 buffer.frameCount = 0; 4429 } 4430 } 4431 } 4432 } else { 4433 // resampling 4434 4435 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4436 // alter output frame count as if we were expecting stereo samples 4437 if (mChannelCount == 1 && mReqChannelCount == 1) { 4438 framesOut >>= 1; 4439 } 4440 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4441 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4442 // are 32 bit aligned which should be always true. 4443 if (mChannelCount == 2 && mReqChannelCount == 1) { 4444 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4445 // the resampler always outputs stereo samples: do post stereo to mono conversion 4446 int16_t *src = (int16_t *)mRsmpOutBuffer; 4447 int16_t *dst = buffer.i16; 4448 while (framesOut--) { 4449 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4450 src += 2; 4451 } 4452 } else { 4453 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4454 } 4455 4456 } 4457 mActiveTrack->releaseBuffer(&buffer); 4458 mActiveTrack->overflow(); 4459 } 4460 // client isn't retrieving buffers fast enough 4461 else { 4462 if (!mActiveTrack->setOverflow()) { 4463 nsecs_t now = systemTime(); 4464 if ((now - lastWarning) > kWarningThrottleNs) { 4465 ALOGW("RecordThread: buffer overflow"); 4466 lastWarning = now; 4467 } 4468 } 4469 // Release the processor for a while before asking for a new buffer. 4470 // This will give the application more chance to read from the buffer and 4471 // clear the overflow. 4472 usleep(kRecordThreadSleepUs); 4473 } 4474 } 4475 // enable changes in effect chain 4476 unlockEffectChains(effectChains); 4477 effectChains.clear(); 4478 } 4479 4480 if (!mStandby) { 4481 mInput->stream->common.standby(&mInput->stream->common); 4482 } 4483 mActiveTrack.clear(); 4484 4485 mStartStopCond.broadcast(); 4486 4487 releaseWakeLock(); 4488 4489 ALOGV("RecordThread %p exiting", this); 4490 return false; 4491} 4492 4493 4494sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4495 const sp<AudioFlinger::Client>& client, 4496 uint32_t sampleRate, 4497 audio_format_t format, 4498 int channelMask, 4499 int frameCount, 4500 uint32_t flags, 4501 int sessionId, 4502 status_t *status) 4503{ 4504 sp<RecordTrack> track; 4505 status_t lStatus; 4506 4507 lStatus = initCheck(); 4508 if (lStatus != NO_ERROR) { 4509 ALOGE("Audio driver not initialized."); 4510 goto Exit; 4511 } 4512 4513 { // scope for mLock 4514 Mutex::Autolock _l(mLock); 4515 4516 track = new RecordTrack(this, client, sampleRate, 4517 format, channelMask, frameCount, flags, sessionId); 4518 4519 if (track->getCblk() == NULL) { 4520 lStatus = NO_MEMORY; 4521 goto Exit; 4522 } 4523 4524 mTrack = track.get(); 4525 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4526 bool suspend = audio_is_bluetooth_sco_device( 4527 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4528 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4529 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4530 } 4531 lStatus = NO_ERROR; 4532 4533Exit: 4534 if (status) { 4535 *status = lStatus; 4536 } 4537 return track; 4538} 4539 4540status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4541{ 4542 ALOGV("RecordThread::start"); 4543 sp <ThreadBase> strongMe = this; 4544 status_t status = NO_ERROR; 4545 { 4546 AutoMutex lock(mLock); 4547 if (mActiveTrack != 0) { 4548 if (recordTrack != mActiveTrack.get()) { 4549 status = -EBUSY; 4550 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4551 mActiveTrack->mState = TrackBase::ACTIVE; 4552 } 4553 return status; 4554 } 4555 4556 recordTrack->mState = TrackBase::IDLE; 4557 mActiveTrack = recordTrack; 4558 mLock.unlock(); 4559 status_t status = AudioSystem::startInput(mId); 4560 mLock.lock(); 4561 if (status != NO_ERROR) { 4562 mActiveTrack.clear(); 4563 return status; 4564 } 4565 mRsmpInIndex = mFrameCount; 4566 mBytesRead = 0; 4567 if (mResampler != NULL) { 4568 mResampler->reset(); 4569 } 4570 mActiveTrack->mState = TrackBase::RESUMING; 4571 // signal thread to start 4572 ALOGV("Signal record thread"); 4573 mWaitWorkCV.signal(); 4574 // do not wait for mStartStopCond if exiting 4575 if (mExiting) { 4576 mActiveTrack.clear(); 4577 status = INVALID_OPERATION; 4578 goto startError; 4579 } 4580 mStartStopCond.wait(mLock); 4581 if (mActiveTrack == 0) { 4582 ALOGV("Record failed to start"); 4583 status = BAD_VALUE; 4584 goto startError; 4585 } 4586 ALOGV("Record started OK"); 4587 return status; 4588 } 4589startError: 4590 AudioSystem::stopInput(mId); 4591 return status; 4592} 4593 4594void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4595 ALOGV("RecordThread::stop"); 4596 sp <ThreadBase> strongMe = this; 4597 { 4598 AutoMutex lock(mLock); 4599 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4600 mActiveTrack->mState = TrackBase::PAUSING; 4601 // do not wait for mStartStopCond if exiting 4602 if (mExiting) { 4603 return; 4604 } 4605 mStartStopCond.wait(mLock); 4606 // if we have been restarted, recordTrack == mActiveTrack.get() here 4607 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4608 mLock.unlock(); 4609 AudioSystem::stopInput(mId); 4610 mLock.lock(); 4611 ALOGV("Record stopped OK"); 4612 } 4613 } 4614 } 4615} 4616 4617status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4618{ 4619 const size_t SIZE = 256; 4620 char buffer[SIZE]; 4621 String8 result; 4622 pid_t pid = 0; 4623 4624 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4625 result.append(buffer); 4626 4627 if (mActiveTrack != 0) { 4628 result.append("Active Track:\n"); 4629 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4630 mActiveTrack->dump(buffer, SIZE); 4631 result.append(buffer); 4632 4633 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4634 result.append(buffer); 4635 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4636 result.append(buffer); 4637 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4638 result.append(buffer); 4639 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4640 result.append(buffer); 4641 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4642 result.append(buffer); 4643 4644 4645 } else { 4646 result.append("No record client\n"); 4647 } 4648 write(fd, result.string(), result.size()); 4649 4650 dumpBase(fd, args); 4651 dumpEffectChains(fd, args); 4652 4653 return NO_ERROR; 4654} 4655 4656status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4657{ 4658 size_t framesReq = buffer->frameCount; 4659 size_t framesReady = mFrameCount - mRsmpInIndex; 4660 int channelCount; 4661 4662 if (framesReady == 0) { 4663 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4664 if (mBytesRead < 0) { 4665 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4666 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4667 // Force input into standby so that it tries to 4668 // recover at next read attempt 4669 mInput->stream->common.standby(&mInput->stream->common); 4670 usleep(kRecordThreadSleepUs); 4671 } 4672 buffer->raw = NULL; 4673 buffer->frameCount = 0; 4674 return NOT_ENOUGH_DATA; 4675 } 4676 mRsmpInIndex = 0; 4677 framesReady = mFrameCount; 4678 } 4679 4680 if (framesReq > framesReady) { 4681 framesReq = framesReady; 4682 } 4683 4684 if (mChannelCount == 1 && mReqChannelCount == 2) { 4685 channelCount = 1; 4686 } else { 4687 channelCount = 2; 4688 } 4689 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4690 buffer->frameCount = framesReq; 4691 return NO_ERROR; 4692} 4693 4694void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4695{ 4696 mRsmpInIndex += buffer->frameCount; 4697 buffer->frameCount = 0; 4698} 4699 4700bool AudioFlinger::RecordThread::checkForNewParameters_l() 4701{ 4702 bool reconfig = false; 4703 4704 while (!mNewParameters.isEmpty()) { 4705 status_t status = NO_ERROR; 4706 String8 keyValuePair = mNewParameters[0]; 4707 AudioParameter param = AudioParameter(keyValuePair); 4708 int value; 4709 audio_format_t reqFormat = mFormat; 4710 int reqSamplingRate = mReqSampleRate; 4711 int reqChannelCount = mReqChannelCount; 4712 4713 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4714 reqSamplingRate = value; 4715 reconfig = true; 4716 } 4717 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4718 reqFormat = (audio_format_t) value; 4719 reconfig = true; 4720 } 4721 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4722 reqChannelCount = popcount(value); 4723 reconfig = true; 4724 } 4725 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4726 // do not accept frame count changes if tracks are open as the track buffer 4727 // size depends on frame count and correct behavior would not be garantied 4728 // if frame count is changed after track creation 4729 if (mActiveTrack != 0) { 4730 status = INVALID_OPERATION; 4731 } else { 4732 reconfig = true; 4733 } 4734 } 4735 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4736 // forward device change to effects that have requested to be 4737 // aware of attached audio device. 4738 for (size_t i = 0; i < mEffectChains.size(); i++) { 4739 mEffectChains[i]->setDevice_l(value); 4740 } 4741 // store input device and output device but do not forward output device to audio HAL. 4742 // Note that status is ignored by the caller for output device 4743 // (see AudioFlinger::setParameters() 4744 if (value & AUDIO_DEVICE_OUT_ALL) { 4745 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4746 status = BAD_VALUE; 4747 } else { 4748 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4749 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4750 if (mTrack != NULL) { 4751 bool suspend = audio_is_bluetooth_sco_device( 4752 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4753 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4754 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4755 } 4756 } 4757 mDevice |= (uint32_t)value; 4758 } 4759 if (status == NO_ERROR) { 4760 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4761 if (status == INVALID_OPERATION) { 4762 mInput->stream->common.standby(&mInput->stream->common); 4763 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4764 } 4765 if (reconfig) { 4766 if (status == BAD_VALUE && 4767 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4768 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4769 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4770 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4771 (reqChannelCount < 3)) { 4772 status = NO_ERROR; 4773 } 4774 if (status == NO_ERROR) { 4775 readInputParameters(); 4776 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4777 } 4778 } 4779 } 4780 4781 mNewParameters.removeAt(0); 4782 4783 mParamStatus = status; 4784 mParamCond.signal(); 4785 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4786 // already timed out waiting for the status and will never signal the condition. 4787 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4788 } 4789 return reconfig; 4790} 4791 4792String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4793{ 4794 char *s; 4795 String8 out_s8 = String8(); 4796 4797 Mutex::Autolock _l(mLock); 4798 if (initCheck() != NO_ERROR) { 4799 return out_s8; 4800 } 4801 4802 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4803 out_s8 = String8(s); 4804 free(s); 4805 return out_s8; 4806} 4807 4808void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4809 AudioSystem::OutputDescriptor desc; 4810 void *param2 = 0; 4811 4812 switch (event) { 4813 case AudioSystem::INPUT_OPENED: 4814 case AudioSystem::INPUT_CONFIG_CHANGED: 4815 desc.channels = mChannelMask; 4816 desc.samplingRate = mSampleRate; 4817 desc.format = mFormat; 4818 desc.frameCount = mFrameCount; 4819 desc.latency = 0; 4820 param2 = &desc; 4821 break; 4822 4823 case AudioSystem::INPUT_CLOSED: 4824 default: 4825 break; 4826 } 4827 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4828} 4829 4830void AudioFlinger::RecordThread::readInputParameters() 4831{ 4832 if (mRsmpInBuffer) delete mRsmpInBuffer; 4833 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4834 if (mResampler) delete mResampler; 4835 mResampler = NULL; 4836 4837 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4838 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4839 mChannelCount = (uint16_t)popcount(mChannelMask); 4840 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4841 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4842 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4843 mFrameCount = mInputBytes / mFrameSize; 4844 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4845 4846 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4847 { 4848 int channelCount; 4849 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4850 // stereo to mono post process as the resampler always outputs stereo. 4851 if (mChannelCount == 1 && mReqChannelCount == 2) { 4852 channelCount = 1; 4853 } else { 4854 channelCount = 2; 4855 } 4856 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4857 mResampler->setSampleRate(mSampleRate); 4858 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4859 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4860 4861 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4862 if (mChannelCount == 1 && mReqChannelCount == 1) { 4863 mFrameCount >>= 1; 4864 } 4865 4866 } 4867 mRsmpInIndex = mFrameCount; 4868} 4869 4870unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4871{ 4872 Mutex::Autolock _l(mLock); 4873 if (initCheck() != NO_ERROR) { 4874 return 0; 4875 } 4876 4877 return mInput->stream->get_input_frames_lost(mInput->stream); 4878} 4879 4880uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4881{ 4882 Mutex::Autolock _l(mLock); 4883 uint32_t result = 0; 4884 if (getEffectChain_l(sessionId) != 0) { 4885 result = EFFECT_SESSION; 4886 } 4887 4888 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4889 result |= TRACK_SESSION; 4890 } 4891 4892 return result; 4893} 4894 4895AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4896{ 4897 Mutex::Autolock _l(mLock); 4898 return mTrack; 4899} 4900 4901AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4902{ 4903 Mutex::Autolock _l(mLock); 4904 return mInput; 4905} 4906 4907AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4908{ 4909 Mutex::Autolock _l(mLock); 4910 AudioStreamIn *input = mInput; 4911 mInput = NULL; 4912 return input; 4913} 4914 4915// this method must always be called either with ThreadBase mLock held or inside the thread loop 4916audio_stream_t* AudioFlinger::RecordThread::stream() 4917{ 4918 if (mInput == NULL) { 4919 return NULL; 4920 } 4921 return &mInput->stream->common; 4922} 4923 4924 4925// ---------------------------------------------------------------------------- 4926 4927int AudioFlinger::openOutput(uint32_t *pDevices, 4928 uint32_t *pSamplingRate, 4929 audio_format_t *pFormat, 4930 uint32_t *pChannels, 4931 uint32_t *pLatencyMs, 4932 uint32_t flags) 4933{ 4934 status_t status; 4935 PlaybackThread *thread = NULL; 4936 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4937 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4938 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4939 uint32_t channels = pChannels ? *pChannels : 0; 4940 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4941 audio_stream_out_t *outStream; 4942 audio_hw_device_t *outHwDev; 4943 4944 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4945 pDevices ? *pDevices : 0, 4946 samplingRate, 4947 format, 4948 channels, 4949 flags); 4950 4951 if (pDevices == NULL || *pDevices == 0) { 4952 return 0; 4953 } 4954 4955 Mutex::Autolock _l(mLock); 4956 4957 outHwDev = findSuitableHwDev_l(*pDevices); 4958 if (outHwDev == NULL) 4959 return 0; 4960 4961 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4962 &channels, &samplingRate, &outStream); 4963 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4964 outStream, 4965 samplingRate, 4966 format, 4967 channels, 4968 status); 4969 4970 mHardwareStatus = AUDIO_HW_IDLE; 4971 if (outStream != NULL) { 4972 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4973 int id = nextUniqueId(); 4974 4975 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4976 (format != AUDIO_FORMAT_PCM_16_BIT) || 4977 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4978 thread = new DirectOutputThread(this, output, id, *pDevices); 4979 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4980 } else { 4981 thread = new MixerThread(this, output, id, *pDevices); 4982 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4983 } 4984 mPlaybackThreads.add(id, thread); 4985 4986 if (pSamplingRate) *pSamplingRate = samplingRate; 4987 if (pFormat) *pFormat = format; 4988 if (pChannels) *pChannels = channels; 4989 if (pLatencyMs) *pLatencyMs = thread->latency(); 4990 4991 // notify client processes of the new output creation 4992 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4993 return id; 4994 } 4995 4996 return 0; 4997} 4998 4999int AudioFlinger::openDuplicateOutput(int output1, int output2) 5000{ 5001 Mutex::Autolock _l(mLock); 5002 MixerThread *thread1 = checkMixerThread_l(output1); 5003 MixerThread *thread2 = checkMixerThread_l(output2); 5004 5005 if (thread1 == NULL || thread2 == NULL) { 5006 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5007 return 0; 5008 } 5009 5010 int id = nextUniqueId(); 5011 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5012 thread->addOutputTrack(thread2); 5013 mPlaybackThreads.add(id, thread); 5014 // notify client processes of the new output creation 5015 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5016 return id; 5017} 5018 5019status_t AudioFlinger::closeOutput(int output) 5020{ 5021 // keep strong reference on the playback thread so that 5022 // it is not destroyed while exit() is executed 5023 sp <PlaybackThread> thread; 5024 { 5025 Mutex::Autolock _l(mLock); 5026 thread = checkPlaybackThread_l(output); 5027 if (thread == NULL) { 5028 return BAD_VALUE; 5029 } 5030 5031 ALOGV("closeOutput() %d", output); 5032 5033 if (thread->type() == ThreadBase::MIXER) { 5034 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5035 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5036 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5037 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5038 } 5039 } 5040 } 5041 void *param2 = 0; 5042 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5043 mPlaybackThreads.removeItem(output); 5044 } 5045 thread->exit(); 5046 5047 if (thread->type() != ThreadBase::DUPLICATING) { 5048 AudioStreamOut *out = thread->clearOutput(); 5049 assert(out != NULL); 5050 // from now on thread->mOutput is NULL 5051 out->hwDev->close_output_stream(out->hwDev, out->stream); 5052 delete out; 5053 } 5054 return NO_ERROR; 5055} 5056 5057status_t AudioFlinger::suspendOutput(int output) 5058{ 5059 Mutex::Autolock _l(mLock); 5060 PlaybackThread *thread = checkPlaybackThread_l(output); 5061 5062 if (thread == NULL) { 5063 return BAD_VALUE; 5064 } 5065 5066 ALOGV("suspendOutput() %d", output); 5067 thread->suspend(); 5068 5069 return NO_ERROR; 5070} 5071 5072status_t AudioFlinger::restoreOutput(int output) 5073{ 5074 Mutex::Autolock _l(mLock); 5075 PlaybackThread *thread = checkPlaybackThread_l(output); 5076 5077 if (thread == NULL) { 5078 return BAD_VALUE; 5079 } 5080 5081 ALOGV("restoreOutput() %d", output); 5082 5083 thread->restore(); 5084 5085 return NO_ERROR; 5086} 5087 5088int AudioFlinger::openInput(uint32_t *pDevices, 5089 uint32_t *pSamplingRate, 5090 audio_format_t *pFormat, 5091 uint32_t *pChannels, 5092 uint32_t acoustics) 5093{ 5094 status_t status; 5095 RecordThread *thread = NULL; 5096 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5097 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5098 uint32_t channels = pChannels ? *pChannels : 0; 5099 uint32_t reqSamplingRate = samplingRate; 5100 audio_format_t reqFormat = format; 5101 uint32_t reqChannels = channels; 5102 audio_stream_in_t *inStream; 5103 audio_hw_device_t *inHwDev; 5104 5105 if (pDevices == NULL || *pDevices == 0) { 5106 return 0; 5107 } 5108 5109 Mutex::Autolock _l(mLock); 5110 5111 inHwDev = findSuitableHwDev_l(*pDevices); 5112 if (inHwDev == NULL) 5113 return 0; 5114 5115 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5116 &channels, &samplingRate, 5117 (audio_in_acoustics_t)acoustics, 5118 &inStream); 5119 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5120 inStream, 5121 samplingRate, 5122 format, 5123 channels, 5124 acoustics, 5125 status); 5126 5127 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5128 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5129 // or stereo to mono conversions on 16 bit PCM inputs. 5130 if (inStream == NULL && status == BAD_VALUE && 5131 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5132 (samplingRate <= 2 * reqSamplingRate) && 5133 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5134 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5135 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5136 &channels, &samplingRate, 5137 (audio_in_acoustics_t)acoustics, 5138 &inStream); 5139 } 5140 5141 if (inStream != NULL) { 5142 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5143 5144 int id = nextUniqueId(); 5145 // Start record thread 5146 // RecorThread require both input and output device indication to forward to audio 5147 // pre processing modules 5148 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5149 thread = new RecordThread(this, 5150 input, 5151 reqSamplingRate, 5152 reqChannels, 5153 id, 5154 device); 5155 mRecordThreads.add(id, thread); 5156 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5157 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5158 if (pFormat) *pFormat = format; 5159 if (pChannels) *pChannels = reqChannels; 5160 5161 input->stream->common.standby(&input->stream->common); 5162 5163 // notify client processes of the new input creation 5164 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5165 return id; 5166 } 5167 5168 return 0; 5169} 5170 5171status_t AudioFlinger::closeInput(int input) 5172{ 5173 // keep strong reference on the record thread so that 5174 // it is not destroyed while exit() is executed 5175 sp <RecordThread> thread; 5176 { 5177 Mutex::Autolock _l(mLock); 5178 thread = checkRecordThread_l(input); 5179 if (thread == NULL) { 5180 return BAD_VALUE; 5181 } 5182 5183 ALOGV("closeInput() %d", input); 5184 void *param2 = 0; 5185 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5186 mRecordThreads.removeItem(input); 5187 } 5188 thread->exit(); 5189 5190 AudioStreamIn *in = thread->clearInput(); 5191 assert(in != NULL); 5192 // from now on thread->mInput is NULL 5193 in->hwDev->close_input_stream(in->hwDev, in->stream); 5194 delete in; 5195 5196 return NO_ERROR; 5197} 5198 5199status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5200{ 5201 Mutex::Autolock _l(mLock); 5202 MixerThread *dstThread = checkMixerThread_l(output); 5203 if (dstThread == NULL) { 5204 ALOGW("setStreamOutput() bad output id %d", output); 5205 return BAD_VALUE; 5206 } 5207 5208 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5209 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5210 5211 dstThread->setStreamValid(stream, true); 5212 5213 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5214 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5215 if (thread != dstThread && 5216 thread->type() != ThreadBase::DIRECT) { 5217 MixerThread *srcThread = (MixerThread *)thread; 5218 srcThread->setStreamValid(stream, false); 5219 srcThread->invalidateTracks(stream); 5220 } 5221 } 5222 5223 return NO_ERROR; 5224} 5225 5226 5227int AudioFlinger::newAudioSessionId() 5228{ 5229 return nextUniqueId(); 5230} 5231 5232void AudioFlinger::acquireAudioSessionId(int audioSession) 5233{ 5234 Mutex::Autolock _l(mLock); 5235 int caller = IPCThreadState::self()->getCallingPid(); 5236 ALOGV("acquiring %d from %d", audioSession, caller); 5237 int num = mAudioSessionRefs.size(); 5238 for (int i = 0; i< num; i++) { 5239 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5240 if (ref->sessionid == audioSession && ref->pid == caller) { 5241 ref->cnt++; 5242 ALOGV(" incremented refcount to %d", ref->cnt); 5243 return; 5244 } 5245 } 5246 AudioSessionRef *ref = new AudioSessionRef(); 5247 ref->sessionid = audioSession; 5248 ref->pid = caller; 5249 ref->cnt = 1; 5250 mAudioSessionRefs.push(ref); 5251 ALOGV(" added new entry for %d", ref->sessionid); 5252} 5253 5254void AudioFlinger::releaseAudioSessionId(int audioSession) 5255{ 5256 Mutex::Autolock _l(mLock); 5257 int caller = IPCThreadState::self()->getCallingPid(); 5258 ALOGV("releasing %d from %d", audioSession, caller); 5259 int num = mAudioSessionRefs.size(); 5260 for (int i = 0; i< num; i++) { 5261 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5262 if (ref->sessionid == audioSession && ref->pid == caller) { 5263 ref->cnt--; 5264 ALOGV(" decremented refcount to %d", ref->cnt); 5265 if (ref->cnt == 0) { 5266 mAudioSessionRefs.removeAt(i); 5267 delete ref; 5268 purgeStaleEffects_l(); 5269 } 5270 return; 5271 } 5272 } 5273 ALOGW("session id %d not found for pid %d", audioSession, caller); 5274} 5275 5276void AudioFlinger::purgeStaleEffects_l() { 5277 5278 ALOGV("purging stale effects"); 5279 5280 Vector< sp<EffectChain> > chains; 5281 5282 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5283 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5284 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5285 sp<EffectChain> ec = t->mEffectChains[j]; 5286 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5287 chains.push(ec); 5288 } 5289 } 5290 } 5291 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5292 sp<RecordThread> t = mRecordThreads.valueAt(i); 5293 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5294 sp<EffectChain> ec = t->mEffectChains[j]; 5295 chains.push(ec); 5296 } 5297 } 5298 5299 for (size_t i = 0; i < chains.size(); i++) { 5300 sp<EffectChain> ec = chains[i]; 5301 int sessionid = ec->sessionId(); 5302 sp<ThreadBase> t = ec->mThread.promote(); 5303 if (t == 0) { 5304 continue; 5305 } 5306 size_t numsessionrefs = mAudioSessionRefs.size(); 5307 bool found = false; 5308 for (size_t k = 0; k < numsessionrefs; k++) { 5309 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5310 if (ref->sessionid == sessionid) { 5311 ALOGV(" session %d still exists for %d with %d refs", 5312 sessionid, ref->pid, ref->cnt); 5313 found = true; 5314 break; 5315 } 5316 } 5317 if (!found) { 5318 // remove all effects from the chain 5319 while (ec->mEffects.size()) { 5320 sp<EffectModule> effect = ec->mEffects[0]; 5321 effect->unPin(); 5322 Mutex::Autolock _l (t->mLock); 5323 t->removeEffect_l(effect); 5324 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5325 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5326 if (handle != 0) { 5327 handle->mEffect.clear(); 5328 if (handle->mHasControl && handle->mEnabled) { 5329 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5330 } 5331 } 5332 } 5333 AudioSystem::unregisterEffect(effect->id()); 5334 } 5335 } 5336 } 5337 return; 5338} 5339 5340// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5341AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5342{ 5343 PlaybackThread *thread = NULL; 5344 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5345 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5346 } 5347 return thread; 5348} 5349 5350// checkMixerThread_l() must be called with AudioFlinger::mLock held 5351AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5352{ 5353 PlaybackThread *thread = checkPlaybackThread_l(output); 5354 if (thread != NULL) { 5355 if (thread->type() == ThreadBase::DIRECT) { 5356 thread = NULL; 5357 } 5358 } 5359 return (MixerThread *)thread; 5360} 5361 5362// checkRecordThread_l() must be called with AudioFlinger::mLock held 5363AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5364{ 5365 RecordThread *thread = NULL; 5366 if (mRecordThreads.indexOfKey(input) >= 0) { 5367 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5368 } 5369 return thread; 5370} 5371 5372uint32_t AudioFlinger::nextUniqueId() 5373{ 5374 return android_atomic_inc(&mNextUniqueId); 5375} 5376 5377AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5378{ 5379 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5380 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5381 AudioStreamOut *output = thread->getOutput(); 5382 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5383 return thread; 5384 } 5385 } 5386 return NULL; 5387} 5388 5389uint32_t AudioFlinger::primaryOutputDevice_l() 5390{ 5391 PlaybackThread *thread = primaryPlaybackThread_l(); 5392 5393 if (thread == NULL) { 5394 return 0; 5395 } 5396 5397 return thread->device(); 5398} 5399 5400 5401// ---------------------------------------------------------------------------- 5402// Effect management 5403// ---------------------------------------------------------------------------- 5404 5405 5406status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5407{ 5408 Mutex::Autolock _l(mLock); 5409 return EffectQueryNumberEffects(numEffects); 5410} 5411 5412status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5413{ 5414 Mutex::Autolock _l(mLock); 5415 return EffectQueryEffect(index, descriptor); 5416} 5417 5418status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5419{ 5420 Mutex::Autolock _l(mLock); 5421 return EffectGetDescriptor(pUuid, descriptor); 5422} 5423 5424 5425sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5426 effect_descriptor_t *pDesc, 5427 const sp<IEffectClient>& effectClient, 5428 int32_t priority, 5429 int io, 5430 int sessionId, 5431 status_t *status, 5432 int *id, 5433 int *enabled) 5434{ 5435 status_t lStatus = NO_ERROR; 5436 sp<EffectHandle> handle; 5437 effect_descriptor_t desc; 5438 sp<Client> client; 5439 wp<Client> wclient; 5440 5441 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5442 pid, effectClient.get(), priority, sessionId, io); 5443 5444 if (pDesc == NULL) { 5445 lStatus = BAD_VALUE; 5446 goto Exit; 5447 } 5448 5449 // check audio settings permission for global effects 5450 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5451 lStatus = PERMISSION_DENIED; 5452 goto Exit; 5453 } 5454 5455 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5456 // that can only be created by audio policy manager (running in same process) 5457 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5458 lStatus = PERMISSION_DENIED; 5459 goto Exit; 5460 } 5461 5462 if (io == 0) { 5463 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5464 // output must be specified by AudioPolicyManager when using session 5465 // AUDIO_SESSION_OUTPUT_STAGE 5466 lStatus = BAD_VALUE; 5467 goto Exit; 5468 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5469 // if the output returned by getOutputForEffect() is removed before we lock the 5470 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5471 // and we will exit safely 5472 io = AudioSystem::getOutputForEffect(&desc); 5473 } 5474 } 5475 5476 { 5477 Mutex::Autolock _l(mLock); 5478 5479 5480 if (!EffectIsNullUuid(&pDesc->uuid)) { 5481 // if uuid is specified, request effect descriptor 5482 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5483 if (lStatus < 0) { 5484 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5485 goto Exit; 5486 } 5487 } else { 5488 // if uuid is not specified, look for an available implementation 5489 // of the required type in effect factory 5490 if (EffectIsNullUuid(&pDesc->type)) { 5491 ALOGW("createEffect() no effect type"); 5492 lStatus = BAD_VALUE; 5493 goto Exit; 5494 } 5495 uint32_t numEffects = 0; 5496 effect_descriptor_t d; 5497 d.flags = 0; // prevent compiler warning 5498 bool found = false; 5499 5500 lStatus = EffectQueryNumberEffects(&numEffects); 5501 if (lStatus < 0) { 5502 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5503 goto Exit; 5504 } 5505 for (uint32_t i = 0; i < numEffects; i++) { 5506 lStatus = EffectQueryEffect(i, &desc); 5507 if (lStatus < 0) { 5508 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5509 continue; 5510 } 5511 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5512 // If matching type found save effect descriptor. If the session is 5513 // 0 and the effect is not auxiliary, continue enumeration in case 5514 // an auxiliary version of this effect type is available 5515 found = true; 5516 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5517 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5518 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5519 break; 5520 } 5521 } 5522 } 5523 if (!found) { 5524 lStatus = BAD_VALUE; 5525 ALOGW("createEffect() effect not found"); 5526 goto Exit; 5527 } 5528 // For same effect type, chose auxiliary version over insert version if 5529 // connect to output mix (Compliance to OpenSL ES) 5530 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5531 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5532 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5533 } 5534 } 5535 5536 // Do not allow auxiliary effects on a session different from 0 (output mix) 5537 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5538 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5539 lStatus = INVALID_OPERATION; 5540 goto Exit; 5541 } 5542 5543 // check recording permission for visualizer 5544 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5545 !recordingAllowed()) { 5546 lStatus = PERMISSION_DENIED; 5547 goto Exit; 5548 } 5549 5550 // return effect descriptor 5551 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5552 5553 // If output is not specified try to find a matching audio session ID in one of the 5554 // output threads. 5555 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5556 // because of code checking output when entering the function. 5557 // Note: io is never 0 when creating an effect on an input 5558 if (io == 0) { 5559 // look for the thread where the specified audio session is present 5560 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5561 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5562 io = mPlaybackThreads.keyAt(i); 5563 break; 5564 } 5565 } 5566 if (io == 0) { 5567 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5568 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5569 io = mRecordThreads.keyAt(i); 5570 break; 5571 } 5572 } 5573 } 5574 // If no output thread contains the requested session ID, default to 5575 // first output. The effect chain will be moved to the correct output 5576 // thread when a track with the same session ID is created 5577 if (io == 0 && mPlaybackThreads.size()) { 5578 io = mPlaybackThreads.keyAt(0); 5579 } 5580 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5581 } 5582 ThreadBase *thread = checkRecordThread_l(io); 5583 if (thread == NULL) { 5584 thread = checkPlaybackThread_l(io); 5585 if (thread == NULL) { 5586 ALOGE("createEffect() unknown output thread"); 5587 lStatus = BAD_VALUE; 5588 goto Exit; 5589 } 5590 } 5591 5592 wclient = mClients.valueFor(pid); 5593 5594 if (wclient != NULL) { 5595 client = wclient.promote(); 5596 } else { 5597 client = new Client(this, pid); 5598 mClients.add(pid, client); 5599 } 5600 5601 // create effect on selected output thread 5602 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5603 &desc, enabled, &lStatus); 5604 if (handle != 0 && id != NULL) { 5605 *id = handle->id(); 5606 } 5607 } 5608 5609Exit: 5610 if(status) { 5611 *status = lStatus; 5612 } 5613 return handle; 5614} 5615 5616status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5617{ 5618 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5619 sessionId, srcOutput, dstOutput); 5620 Mutex::Autolock _l(mLock); 5621 if (srcOutput == dstOutput) { 5622 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5623 return NO_ERROR; 5624 } 5625 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5626 if (srcThread == NULL) { 5627 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5628 return BAD_VALUE; 5629 } 5630 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5631 if (dstThread == NULL) { 5632 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5633 return BAD_VALUE; 5634 } 5635 5636 Mutex::Autolock _dl(dstThread->mLock); 5637 Mutex::Autolock _sl(srcThread->mLock); 5638 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5639 5640 return NO_ERROR; 5641} 5642 5643// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5644status_t AudioFlinger::moveEffectChain_l(int sessionId, 5645 AudioFlinger::PlaybackThread *srcThread, 5646 AudioFlinger::PlaybackThread *dstThread, 5647 bool reRegister) 5648{ 5649 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5650 sessionId, srcThread, dstThread); 5651 5652 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5653 if (chain == 0) { 5654 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5655 sessionId, srcThread); 5656 return INVALID_OPERATION; 5657 } 5658 5659 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5660 // so that a new chain is created with correct parameters when first effect is added. This is 5661 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5662 // removed. 5663 srcThread->removeEffectChain_l(chain); 5664 5665 // transfer all effects one by one so that new effect chain is created on new thread with 5666 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5667 int dstOutput = dstThread->id(); 5668 sp<EffectChain> dstChain; 5669 uint32_t strategy = 0; // prevent compiler warning 5670 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5671 while (effect != 0) { 5672 srcThread->removeEffect_l(effect); 5673 dstThread->addEffect_l(effect); 5674 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5675 if (effect->state() == EffectModule::ACTIVE || 5676 effect->state() == EffectModule::STOPPING) { 5677 effect->start(); 5678 } 5679 // if the move request is not received from audio policy manager, the effect must be 5680 // re-registered with the new strategy and output 5681 if (dstChain == 0) { 5682 dstChain = effect->chain().promote(); 5683 if (dstChain == 0) { 5684 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5685 srcThread->addEffect_l(effect); 5686 return NO_INIT; 5687 } 5688 strategy = dstChain->strategy(); 5689 } 5690 if (reRegister) { 5691 AudioSystem::unregisterEffect(effect->id()); 5692 AudioSystem::registerEffect(&effect->desc(), 5693 dstOutput, 5694 strategy, 5695 sessionId, 5696 effect->id()); 5697 } 5698 effect = chain->getEffectFromId_l(0); 5699 } 5700 5701 return NO_ERROR; 5702} 5703 5704 5705// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5706sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5707 const sp<AudioFlinger::Client>& client, 5708 const sp<IEffectClient>& effectClient, 5709 int32_t priority, 5710 int sessionId, 5711 effect_descriptor_t *desc, 5712 int *enabled, 5713 status_t *status 5714 ) 5715{ 5716 sp<EffectModule> effect; 5717 sp<EffectHandle> handle; 5718 status_t lStatus; 5719 sp<EffectChain> chain; 5720 bool chainCreated = false; 5721 bool effectCreated = false; 5722 bool effectRegistered = false; 5723 5724 lStatus = initCheck(); 5725 if (lStatus != NO_ERROR) { 5726 ALOGW("createEffect_l() Audio driver not initialized."); 5727 goto Exit; 5728 } 5729 5730 // Do not allow effects with session ID 0 on direct output or duplicating threads 5731 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5732 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5733 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5734 desc->name, sessionId); 5735 lStatus = BAD_VALUE; 5736 goto Exit; 5737 } 5738 // Only Pre processor effects are allowed on input threads and only on input threads 5739 if ((mType == RECORD && 5740 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5741 (mType != RECORD && 5742 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5743 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5744 desc->name, desc->flags, mType); 5745 lStatus = BAD_VALUE; 5746 goto Exit; 5747 } 5748 5749 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5750 5751 { // scope for mLock 5752 Mutex::Autolock _l(mLock); 5753 5754 // check for existing effect chain with the requested audio session 5755 chain = getEffectChain_l(sessionId); 5756 if (chain == 0) { 5757 // create a new chain for this session 5758 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5759 chain = new EffectChain(this, sessionId); 5760 addEffectChain_l(chain); 5761 chain->setStrategy(getStrategyForSession_l(sessionId)); 5762 chainCreated = true; 5763 } else { 5764 effect = chain->getEffectFromDesc_l(desc); 5765 } 5766 5767 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5768 5769 if (effect == 0) { 5770 int id = mAudioFlinger->nextUniqueId(); 5771 // Check CPU and memory usage 5772 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5773 if (lStatus != NO_ERROR) { 5774 goto Exit; 5775 } 5776 effectRegistered = true; 5777 // create a new effect module if none present in the chain 5778 effect = new EffectModule(this, chain, desc, id, sessionId); 5779 lStatus = effect->status(); 5780 if (lStatus != NO_ERROR) { 5781 goto Exit; 5782 } 5783 lStatus = chain->addEffect_l(effect); 5784 if (lStatus != NO_ERROR) { 5785 goto Exit; 5786 } 5787 effectCreated = true; 5788 5789 effect->setDevice(mDevice); 5790 effect->setMode(mAudioFlinger->getMode()); 5791 } 5792 // create effect handle and connect it to effect module 5793 handle = new EffectHandle(effect, client, effectClient, priority); 5794 lStatus = effect->addHandle(handle); 5795 if (enabled) { 5796 *enabled = (int)effect->isEnabled(); 5797 } 5798 } 5799 5800Exit: 5801 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5802 Mutex::Autolock _l(mLock); 5803 if (effectCreated) { 5804 chain->removeEffect_l(effect); 5805 } 5806 if (effectRegistered) { 5807 AudioSystem::unregisterEffect(effect->id()); 5808 } 5809 if (chainCreated) { 5810 removeEffectChain_l(chain); 5811 } 5812 handle.clear(); 5813 } 5814 5815 if(status) { 5816 *status = lStatus; 5817 } 5818 return handle; 5819} 5820 5821sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5822{ 5823 sp<EffectModule> effect; 5824 5825 sp<EffectChain> chain = getEffectChain_l(sessionId); 5826 if (chain != 0) { 5827 effect = chain->getEffectFromId_l(effectId); 5828 } 5829 return effect; 5830} 5831 5832// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5833// PlaybackThread::mLock held 5834status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5835{ 5836 // check for existing effect chain with the requested audio session 5837 int sessionId = effect->sessionId(); 5838 sp<EffectChain> chain = getEffectChain_l(sessionId); 5839 bool chainCreated = false; 5840 5841 if (chain == 0) { 5842 // create a new chain for this session 5843 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5844 chain = new EffectChain(this, sessionId); 5845 addEffectChain_l(chain); 5846 chain->setStrategy(getStrategyForSession_l(sessionId)); 5847 chainCreated = true; 5848 } 5849 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5850 5851 if (chain->getEffectFromId_l(effect->id()) != 0) { 5852 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5853 this, effect->desc().name, chain.get()); 5854 return BAD_VALUE; 5855 } 5856 5857 status_t status = chain->addEffect_l(effect); 5858 if (status != NO_ERROR) { 5859 if (chainCreated) { 5860 removeEffectChain_l(chain); 5861 } 5862 return status; 5863 } 5864 5865 effect->setDevice(mDevice); 5866 effect->setMode(mAudioFlinger->getMode()); 5867 return NO_ERROR; 5868} 5869 5870void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5871 5872 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5873 effect_descriptor_t desc = effect->desc(); 5874 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5875 detachAuxEffect_l(effect->id()); 5876 } 5877 5878 sp<EffectChain> chain = effect->chain().promote(); 5879 if (chain != 0) { 5880 // remove effect chain if removing last effect 5881 if (chain->removeEffect_l(effect) == 0) { 5882 removeEffectChain_l(chain); 5883 } 5884 } else { 5885 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5886 } 5887} 5888 5889void AudioFlinger::ThreadBase::lockEffectChains_l( 5890 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5891{ 5892 effectChains = mEffectChains; 5893 for (size_t i = 0; i < mEffectChains.size(); i++) { 5894 mEffectChains[i]->lock(); 5895 } 5896} 5897 5898void AudioFlinger::ThreadBase::unlockEffectChains( 5899 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5900{ 5901 for (size_t i = 0; i < effectChains.size(); i++) { 5902 effectChains[i]->unlock(); 5903 } 5904} 5905 5906sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5907{ 5908 Mutex::Autolock _l(mLock); 5909 return getEffectChain_l(sessionId); 5910} 5911 5912sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5913{ 5914 sp<EffectChain> chain; 5915 5916 size_t size = mEffectChains.size(); 5917 for (size_t i = 0; i < size; i++) { 5918 if (mEffectChains[i]->sessionId() == sessionId) { 5919 chain = mEffectChains[i]; 5920 break; 5921 } 5922 } 5923 return chain; 5924} 5925 5926void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5927{ 5928 Mutex::Autolock _l(mLock); 5929 size_t size = mEffectChains.size(); 5930 for (size_t i = 0; i < size; i++) { 5931 mEffectChains[i]->setMode_l(mode); 5932 } 5933} 5934 5935void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5936 const wp<EffectHandle>& handle, 5937 bool unpiniflast) { 5938 5939 Mutex::Autolock _l(mLock); 5940 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5941 // delete the effect module if removing last handle on it 5942 if (effect->removeHandle(handle) == 0) { 5943 if (!effect->isPinned() || unpiniflast) { 5944 removeEffect_l(effect); 5945 AudioSystem::unregisterEffect(effect->id()); 5946 } 5947 } 5948} 5949 5950status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5951{ 5952 int session = chain->sessionId(); 5953 int16_t *buffer = mMixBuffer; 5954 bool ownsBuffer = false; 5955 5956 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5957 if (session > 0) { 5958 // Only one effect chain can be present in direct output thread and it uses 5959 // the mix buffer as input 5960 if (mType != DIRECT) { 5961 size_t numSamples = mFrameCount * mChannelCount; 5962 buffer = new int16_t[numSamples]; 5963 memset(buffer, 0, numSamples * sizeof(int16_t)); 5964 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5965 ownsBuffer = true; 5966 } 5967 5968 // Attach all tracks with same session ID to this chain. 5969 for (size_t i = 0; i < mTracks.size(); ++i) { 5970 sp<Track> track = mTracks[i]; 5971 if (session == track->sessionId()) { 5972 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5973 track->setMainBuffer(buffer); 5974 chain->incTrackCnt(); 5975 } 5976 } 5977 5978 // indicate all active tracks in the chain 5979 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5980 sp<Track> track = mActiveTracks[i].promote(); 5981 if (track == 0) continue; 5982 if (session == track->sessionId()) { 5983 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5984 chain->incActiveTrackCnt(); 5985 } 5986 } 5987 } 5988 5989 chain->setInBuffer(buffer, ownsBuffer); 5990 chain->setOutBuffer(mMixBuffer); 5991 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5992 // chains list in order to be processed last as it contains output stage effects 5993 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5994 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5995 // after track specific effects and before output stage 5996 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5997 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5998 // Effect chain for other sessions are inserted at beginning of effect 5999 // chains list to be processed before output mix effects. Relative order between other 6000 // sessions is not important 6001 size_t size = mEffectChains.size(); 6002 size_t i = 0; 6003 for (i = 0; i < size; i++) { 6004 if (mEffectChains[i]->sessionId() < session) break; 6005 } 6006 mEffectChains.insertAt(chain, i); 6007 checkSuspendOnAddEffectChain_l(chain); 6008 6009 return NO_ERROR; 6010} 6011 6012size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6013{ 6014 int session = chain->sessionId(); 6015 6016 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6017 6018 for (size_t i = 0; i < mEffectChains.size(); i++) { 6019 if (chain == mEffectChains[i]) { 6020 mEffectChains.removeAt(i); 6021 // detach all active tracks from the chain 6022 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6023 sp<Track> track = mActiveTracks[i].promote(); 6024 if (track == 0) continue; 6025 if (session == track->sessionId()) { 6026 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6027 chain.get(), session); 6028 chain->decActiveTrackCnt(); 6029 } 6030 } 6031 6032 // detach all tracks with same session ID from this chain 6033 for (size_t i = 0; i < mTracks.size(); ++i) { 6034 sp<Track> track = mTracks[i]; 6035 if (session == track->sessionId()) { 6036 track->setMainBuffer(mMixBuffer); 6037 chain->decTrackCnt(); 6038 } 6039 } 6040 break; 6041 } 6042 } 6043 return mEffectChains.size(); 6044} 6045 6046status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6047 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6048{ 6049 Mutex::Autolock _l(mLock); 6050 return attachAuxEffect_l(track, EffectId); 6051} 6052 6053status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6054 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6055{ 6056 status_t status = NO_ERROR; 6057 6058 if (EffectId == 0) { 6059 track->setAuxBuffer(0, NULL); 6060 } else { 6061 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6062 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6063 if (effect != 0) { 6064 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6065 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6066 } else { 6067 status = INVALID_OPERATION; 6068 } 6069 } else { 6070 status = BAD_VALUE; 6071 } 6072 } 6073 return status; 6074} 6075 6076void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6077{ 6078 for (size_t i = 0; i < mTracks.size(); ++i) { 6079 sp<Track> track = mTracks[i]; 6080 if (track->auxEffectId() == effectId) { 6081 attachAuxEffect_l(track, 0); 6082 } 6083 } 6084} 6085 6086status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6087{ 6088 // only one chain per input thread 6089 if (mEffectChains.size() != 0) { 6090 return INVALID_OPERATION; 6091 } 6092 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6093 6094 chain->setInBuffer(NULL); 6095 chain->setOutBuffer(NULL); 6096 6097 checkSuspendOnAddEffectChain_l(chain); 6098 6099 mEffectChains.add(chain); 6100 6101 return NO_ERROR; 6102} 6103 6104size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6105{ 6106 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6107 ALOGW_IF(mEffectChains.size() != 1, 6108 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6109 chain.get(), mEffectChains.size(), this); 6110 if (mEffectChains.size() == 1) { 6111 mEffectChains.removeAt(0); 6112 } 6113 return 0; 6114} 6115 6116// ---------------------------------------------------------------------------- 6117// EffectModule implementation 6118// ---------------------------------------------------------------------------- 6119 6120#undef LOG_TAG 6121#define LOG_TAG "AudioFlinger::EffectModule" 6122 6123AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6124 const wp<AudioFlinger::EffectChain>& chain, 6125 effect_descriptor_t *desc, 6126 int id, 6127 int sessionId) 6128 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6129 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6130{ 6131 ALOGV("Constructor %p", this); 6132 int lStatus; 6133 sp<ThreadBase> thread = mThread.promote(); 6134 if (thread == 0) { 6135 return; 6136 } 6137 6138 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6139 6140 // create effect engine from effect factory 6141 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6142 6143 if (mStatus != NO_ERROR) { 6144 return; 6145 } 6146 lStatus = init(); 6147 if (lStatus < 0) { 6148 mStatus = lStatus; 6149 goto Error; 6150 } 6151 6152 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6153 mPinned = true; 6154 } 6155 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6156 return; 6157Error: 6158 EffectRelease(mEffectInterface); 6159 mEffectInterface = NULL; 6160 ALOGV("Constructor Error %d", mStatus); 6161} 6162 6163AudioFlinger::EffectModule::~EffectModule() 6164{ 6165 ALOGV("Destructor %p", this); 6166 if (mEffectInterface != NULL) { 6167 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6168 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6169 sp<ThreadBase> thread = mThread.promote(); 6170 if (thread != 0) { 6171 audio_stream_t *stream = thread->stream(); 6172 if (stream != NULL) { 6173 stream->remove_audio_effect(stream, mEffectInterface); 6174 } 6175 } 6176 } 6177 // release effect engine 6178 EffectRelease(mEffectInterface); 6179 } 6180} 6181 6182status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6183{ 6184 status_t status; 6185 6186 Mutex::Autolock _l(mLock); 6187 // First handle in mHandles has highest priority and controls the effect module 6188 int priority = handle->priority(); 6189 size_t size = mHandles.size(); 6190 sp<EffectHandle> h; 6191 size_t i; 6192 for (i = 0; i < size; i++) { 6193 h = mHandles[i].promote(); 6194 if (h == 0) continue; 6195 if (h->priority() <= priority) break; 6196 } 6197 // if inserted in first place, move effect control from previous owner to this handle 6198 if (i == 0) { 6199 bool enabled = false; 6200 if (h != 0) { 6201 enabled = h->enabled(); 6202 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6203 } 6204 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6205 status = NO_ERROR; 6206 } else { 6207 status = ALREADY_EXISTS; 6208 } 6209 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6210 mHandles.insertAt(handle, i); 6211 return status; 6212} 6213 6214size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6215{ 6216 Mutex::Autolock _l(mLock); 6217 size_t size = mHandles.size(); 6218 size_t i; 6219 for (i = 0; i < size; i++) { 6220 if (mHandles[i] == handle) break; 6221 } 6222 if (i == size) { 6223 return size; 6224 } 6225 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6226 6227 bool enabled = false; 6228 EffectHandle *hdl = handle.unsafe_get(); 6229 if (hdl) { 6230 ALOGV("removeHandle() unsafe_get OK"); 6231 enabled = hdl->enabled(); 6232 } 6233 mHandles.removeAt(i); 6234 size = mHandles.size(); 6235 // if removed from first place, move effect control from this handle to next in line 6236 if (i == 0 && size != 0) { 6237 sp<EffectHandle> h = mHandles[0].promote(); 6238 if (h != 0) { 6239 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6240 } 6241 } 6242 6243 // Prevent calls to process() and other functions on effect interface from now on. 6244 // The effect engine will be released by the destructor when the last strong reference on 6245 // this object is released which can happen after next process is called. 6246 if (size == 0 && !mPinned) { 6247 mState = DESTROYED; 6248 } 6249 6250 return size; 6251} 6252 6253sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6254{ 6255 Mutex::Autolock _l(mLock); 6256 sp<EffectHandle> handle; 6257 if (mHandles.size() != 0) { 6258 handle = mHandles[0].promote(); 6259 } 6260 return handle; 6261} 6262 6263void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6264{ 6265 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6266 // keep a strong reference on this EffectModule to avoid calling the 6267 // destructor before we exit 6268 sp<EffectModule> keep(this); 6269 { 6270 sp<ThreadBase> thread = mThread.promote(); 6271 if (thread != 0) { 6272 thread->disconnectEffect(keep, handle, unpiniflast); 6273 } 6274 } 6275} 6276 6277void AudioFlinger::EffectModule::updateState() { 6278 Mutex::Autolock _l(mLock); 6279 6280 switch (mState) { 6281 case RESTART: 6282 reset_l(); 6283 // FALL THROUGH 6284 6285 case STARTING: 6286 // clear auxiliary effect input buffer for next accumulation 6287 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6288 memset(mConfig.inputCfg.buffer.raw, 6289 0, 6290 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6291 } 6292 start_l(); 6293 mState = ACTIVE; 6294 break; 6295 case STOPPING: 6296 stop_l(); 6297 mDisableWaitCnt = mMaxDisableWaitCnt; 6298 mState = STOPPED; 6299 break; 6300 case STOPPED: 6301 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6302 // turn off sequence. 6303 if (--mDisableWaitCnt == 0) { 6304 reset_l(); 6305 mState = IDLE; 6306 } 6307 break; 6308 default: //IDLE , ACTIVE, DESTROYED 6309 break; 6310 } 6311} 6312 6313void AudioFlinger::EffectModule::process() 6314{ 6315 Mutex::Autolock _l(mLock); 6316 6317 if (mState == DESTROYED || mEffectInterface == NULL || 6318 mConfig.inputCfg.buffer.raw == NULL || 6319 mConfig.outputCfg.buffer.raw == NULL) { 6320 return; 6321 } 6322 6323 if (isProcessEnabled()) { 6324 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6325 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6326 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6327 mConfig.inputCfg.buffer.s32, 6328 mConfig.inputCfg.buffer.frameCount/2); 6329 } 6330 6331 // do the actual processing in the effect engine 6332 int ret = (*mEffectInterface)->process(mEffectInterface, 6333 &mConfig.inputCfg.buffer, 6334 &mConfig.outputCfg.buffer); 6335 6336 // force transition to IDLE state when engine is ready 6337 if (mState == STOPPED && ret == -ENODATA) { 6338 mDisableWaitCnt = 1; 6339 } 6340 6341 // clear auxiliary effect input buffer for next accumulation 6342 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6343 memset(mConfig.inputCfg.buffer.raw, 0, 6344 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6345 } 6346 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6347 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6348 // If an insert effect is idle and input buffer is different from output buffer, 6349 // accumulate input onto output 6350 sp<EffectChain> chain = mChain.promote(); 6351 if (chain != 0 && chain->activeTrackCnt() != 0) { 6352 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6353 int16_t *in = mConfig.inputCfg.buffer.s16; 6354 int16_t *out = mConfig.outputCfg.buffer.s16; 6355 for (size_t i = 0; i < frameCnt; i++) { 6356 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6357 } 6358 } 6359 } 6360} 6361 6362void AudioFlinger::EffectModule::reset_l() 6363{ 6364 if (mEffectInterface == NULL) { 6365 return; 6366 } 6367 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6368} 6369 6370status_t AudioFlinger::EffectModule::configure() 6371{ 6372 uint32_t channels; 6373 if (mEffectInterface == NULL) { 6374 return NO_INIT; 6375 } 6376 6377 sp<ThreadBase> thread = mThread.promote(); 6378 if (thread == 0) { 6379 return DEAD_OBJECT; 6380 } 6381 6382 // TODO: handle configuration of effects replacing track process 6383 if (thread->channelCount() == 1) { 6384 channels = AUDIO_CHANNEL_OUT_MONO; 6385 } else { 6386 channels = AUDIO_CHANNEL_OUT_STEREO; 6387 } 6388 6389 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6390 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6391 } else { 6392 mConfig.inputCfg.channels = channels; 6393 } 6394 mConfig.outputCfg.channels = channels; 6395 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6396 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6397 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6398 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6399 mConfig.inputCfg.bufferProvider.cookie = NULL; 6400 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6401 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6402 mConfig.outputCfg.bufferProvider.cookie = NULL; 6403 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6404 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6405 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6406 // Insert effect: 6407 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6408 // always overwrites output buffer: input buffer == output buffer 6409 // - in other sessions: 6410 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6411 // other effect: overwrites output buffer: input buffer == output buffer 6412 // Auxiliary effect: 6413 // accumulates in output buffer: input buffer != output buffer 6414 // Therefore: accumulate <=> input buffer != output buffer 6415 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6416 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6417 } else { 6418 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6419 } 6420 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6421 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6422 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6423 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6424 6425 ALOGV("configure() %p thread %p buffer %p framecount %d", 6426 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6427 6428 status_t cmdStatus; 6429 uint32_t size = sizeof(int); 6430 status_t status = (*mEffectInterface)->command(mEffectInterface, 6431 EFFECT_CMD_SET_CONFIG, 6432 sizeof(effect_config_t), 6433 &mConfig, 6434 &size, 6435 &cmdStatus); 6436 if (status == 0) { 6437 status = cmdStatus; 6438 } 6439 6440 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6441 (1000 * mConfig.outputCfg.buffer.frameCount); 6442 6443 return status; 6444} 6445 6446status_t AudioFlinger::EffectModule::init() 6447{ 6448 Mutex::Autolock _l(mLock); 6449 if (mEffectInterface == NULL) { 6450 return NO_INIT; 6451 } 6452 status_t cmdStatus; 6453 uint32_t size = sizeof(status_t); 6454 status_t status = (*mEffectInterface)->command(mEffectInterface, 6455 EFFECT_CMD_INIT, 6456 0, 6457 NULL, 6458 &size, 6459 &cmdStatus); 6460 if (status == 0) { 6461 status = cmdStatus; 6462 } 6463 return status; 6464} 6465 6466status_t AudioFlinger::EffectModule::start() 6467{ 6468 Mutex::Autolock _l(mLock); 6469 return start_l(); 6470} 6471 6472status_t AudioFlinger::EffectModule::start_l() 6473{ 6474 if (mEffectInterface == NULL) { 6475 return NO_INIT; 6476 } 6477 status_t cmdStatus; 6478 uint32_t size = sizeof(status_t); 6479 status_t status = (*mEffectInterface)->command(mEffectInterface, 6480 EFFECT_CMD_ENABLE, 6481 0, 6482 NULL, 6483 &size, 6484 &cmdStatus); 6485 if (status == 0) { 6486 status = cmdStatus; 6487 } 6488 if (status == 0 && 6489 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6490 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6491 sp<ThreadBase> thread = mThread.promote(); 6492 if (thread != 0) { 6493 audio_stream_t *stream = thread->stream(); 6494 if (stream != NULL) { 6495 stream->add_audio_effect(stream, mEffectInterface); 6496 } 6497 } 6498 } 6499 return status; 6500} 6501 6502status_t AudioFlinger::EffectModule::stop() 6503{ 6504 Mutex::Autolock _l(mLock); 6505 return stop_l(); 6506} 6507 6508status_t AudioFlinger::EffectModule::stop_l() 6509{ 6510 if (mEffectInterface == NULL) { 6511 return NO_INIT; 6512 } 6513 status_t cmdStatus; 6514 uint32_t size = sizeof(status_t); 6515 status_t status = (*mEffectInterface)->command(mEffectInterface, 6516 EFFECT_CMD_DISABLE, 6517 0, 6518 NULL, 6519 &size, 6520 &cmdStatus); 6521 if (status == 0) { 6522 status = cmdStatus; 6523 } 6524 if (status == 0 && 6525 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6526 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6527 sp<ThreadBase> thread = mThread.promote(); 6528 if (thread != 0) { 6529 audio_stream_t *stream = thread->stream(); 6530 if (stream != NULL) { 6531 stream->remove_audio_effect(stream, mEffectInterface); 6532 } 6533 } 6534 } 6535 return status; 6536} 6537 6538status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6539 uint32_t cmdSize, 6540 void *pCmdData, 6541 uint32_t *replySize, 6542 void *pReplyData) 6543{ 6544 Mutex::Autolock _l(mLock); 6545// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6546 6547 if (mState == DESTROYED || mEffectInterface == NULL) { 6548 return NO_INIT; 6549 } 6550 status_t status = (*mEffectInterface)->command(mEffectInterface, 6551 cmdCode, 6552 cmdSize, 6553 pCmdData, 6554 replySize, 6555 pReplyData); 6556 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6557 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6558 for (size_t i = 1; i < mHandles.size(); i++) { 6559 sp<EffectHandle> h = mHandles[i].promote(); 6560 if (h != 0) { 6561 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6562 } 6563 } 6564 } 6565 return status; 6566} 6567 6568status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6569{ 6570 6571 Mutex::Autolock _l(mLock); 6572 ALOGV("setEnabled %p enabled %d", this, enabled); 6573 6574 if (enabled != isEnabled()) { 6575 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6576 if (enabled && status != NO_ERROR) { 6577 return status; 6578 } 6579 6580 switch (mState) { 6581 // going from disabled to enabled 6582 case IDLE: 6583 mState = STARTING; 6584 break; 6585 case STOPPED: 6586 mState = RESTART; 6587 break; 6588 case STOPPING: 6589 mState = ACTIVE; 6590 break; 6591 6592 // going from enabled to disabled 6593 case RESTART: 6594 mState = STOPPED; 6595 break; 6596 case STARTING: 6597 mState = IDLE; 6598 break; 6599 case ACTIVE: 6600 mState = STOPPING; 6601 break; 6602 case DESTROYED: 6603 return NO_ERROR; // simply ignore as we are being destroyed 6604 } 6605 for (size_t i = 1; i < mHandles.size(); i++) { 6606 sp<EffectHandle> h = mHandles[i].promote(); 6607 if (h != 0) { 6608 h->setEnabled(enabled); 6609 } 6610 } 6611 } 6612 return NO_ERROR; 6613} 6614 6615bool AudioFlinger::EffectModule::isEnabled() 6616{ 6617 switch (mState) { 6618 case RESTART: 6619 case STARTING: 6620 case ACTIVE: 6621 return true; 6622 case IDLE: 6623 case STOPPING: 6624 case STOPPED: 6625 case DESTROYED: 6626 default: 6627 return false; 6628 } 6629} 6630 6631bool AudioFlinger::EffectModule::isProcessEnabled() 6632{ 6633 switch (mState) { 6634 case RESTART: 6635 case ACTIVE: 6636 case STOPPING: 6637 case STOPPED: 6638 return true; 6639 case IDLE: 6640 case STARTING: 6641 case DESTROYED: 6642 default: 6643 return false; 6644 } 6645} 6646 6647status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6648{ 6649 Mutex::Autolock _l(mLock); 6650 status_t status = NO_ERROR; 6651 6652 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6653 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6654 if (isProcessEnabled() && 6655 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6656 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6657 status_t cmdStatus; 6658 uint32_t volume[2]; 6659 uint32_t *pVolume = NULL; 6660 uint32_t size = sizeof(volume); 6661 volume[0] = *left; 6662 volume[1] = *right; 6663 if (controller) { 6664 pVolume = volume; 6665 } 6666 status = (*mEffectInterface)->command(mEffectInterface, 6667 EFFECT_CMD_SET_VOLUME, 6668 size, 6669 volume, 6670 &size, 6671 pVolume); 6672 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6673 *left = volume[0]; 6674 *right = volume[1]; 6675 } 6676 } 6677 return status; 6678} 6679 6680status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6681{ 6682 Mutex::Autolock _l(mLock); 6683 status_t status = NO_ERROR; 6684 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6685 // audio pre processing modules on RecordThread can receive both output and 6686 // input device indication in the same call 6687 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6688 if (dev) { 6689 status_t cmdStatus; 6690 uint32_t size = sizeof(status_t); 6691 6692 status = (*mEffectInterface)->command(mEffectInterface, 6693 EFFECT_CMD_SET_DEVICE, 6694 sizeof(uint32_t), 6695 &dev, 6696 &size, 6697 &cmdStatus); 6698 if (status == NO_ERROR) { 6699 status = cmdStatus; 6700 } 6701 } 6702 dev = device & AUDIO_DEVICE_IN_ALL; 6703 if (dev) { 6704 status_t cmdStatus; 6705 uint32_t size = sizeof(status_t); 6706 6707 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6708 EFFECT_CMD_SET_INPUT_DEVICE, 6709 sizeof(uint32_t), 6710 &dev, 6711 &size, 6712 &cmdStatus); 6713 if (status2 == NO_ERROR) { 6714 status2 = cmdStatus; 6715 } 6716 if (status == NO_ERROR) { 6717 status = status2; 6718 } 6719 } 6720 } 6721 return status; 6722} 6723 6724status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6725{ 6726 Mutex::Autolock _l(mLock); 6727 status_t status = NO_ERROR; 6728 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6729 status_t cmdStatus; 6730 uint32_t size = sizeof(status_t); 6731 status = (*mEffectInterface)->command(mEffectInterface, 6732 EFFECT_CMD_SET_AUDIO_MODE, 6733 sizeof(audio_mode_t), 6734 &mode, 6735 &size, 6736 &cmdStatus); 6737 if (status == NO_ERROR) { 6738 status = cmdStatus; 6739 } 6740 } 6741 return status; 6742} 6743 6744void AudioFlinger::EffectModule::setSuspended(bool suspended) 6745{ 6746 Mutex::Autolock _l(mLock); 6747 mSuspended = suspended; 6748} 6749 6750bool AudioFlinger::EffectModule::suspended() const 6751{ 6752 Mutex::Autolock _l(mLock); 6753 return mSuspended; 6754} 6755 6756status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6757{ 6758 const size_t SIZE = 256; 6759 char buffer[SIZE]; 6760 String8 result; 6761 6762 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6763 result.append(buffer); 6764 6765 bool locked = tryLock(mLock); 6766 // failed to lock - AudioFlinger is probably deadlocked 6767 if (!locked) { 6768 result.append("\t\tCould not lock Fx mutex:\n"); 6769 } 6770 6771 result.append("\t\tSession Status State Engine:\n"); 6772 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6773 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6774 result.append(buffer); 6775 6776 result.append("\t\tDescriptor:\n"); 6777 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6778 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6779 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6780 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6781 result.append(buffer); 6782 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6783 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6784 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6785 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6786 result.append(buffer); 6787 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6788 mDescriptor.apiVersion, 6789 mDescriptor.flags); 6790 result.append(buffer); 6791 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6792 mDescriptor.name); 6793 result.append(buffer); 6794 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6795 mDescriptor.implementor); 6796 result.append(buffer); 6797 6798 result.append("\t\t- Input configuration:\n"); 6799 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6800 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6801 (uint32_t)mConfig.inputCfg.buffer.raw, 6802 mConfig.inputCfg.buffer.frameCount, 6803 mConfig.inputCfg.samplingRate, 6804 mConfig.inputCfg.channels, 6805 mConfig.inputCfg.format); 6806 result.append(buffer); 6807 6808 result.append("\t\t- Output configuration:\n"); 6809 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6810 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6811 (uint32_t)mConfig.outputCfg.buffer.raw, 6812 mConfig.outputCfg.buffer.frameCount, 6813 mConfig.outputCfg.samplingRate, 6814 mConfig.outputCfg.channels, 6815 mConfig.outputCfg.format); 6816 result.append(buffer); 6817 6818 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6819 result.append(buffer); 6820 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6821 for (size_t i = 0; i < mHandles.size(); ++i) { 6822 sp<EffectHandle> handle = mHandles[i].promote(); 6823 if (handle != 0) { 6824 handle->dump(buffer, SIZE); 6825 result.append(buffer); 6826 } 6827 } 6828 6829 result.append("\n"); 6830 6831 write(fd, result.string(), result.length()); 6832 6833 if (locked) { 6834 mLock.unlock(); 6835 } 6836 6837 return NO_ERROR; 6838} 6839 6840// ---------------------------------------------------------------------------- 6841// EffectHandle implementation 6842// ---------------------------------------------------------------------------- 6843 6844#undef LOG_TAG 6845#define LOG_TAG "AudioFlinger::EffectHandle" 6846 6847AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6848 const sp<AudioFlinger::Client>& client, 6849 const sp<IEffectClient>& effectClient, 6850 int32_t priority) 6851 : BnEffect(), 6852 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6853 mPriority(priority), mHasControl(false), mEnabled(false) 6854{ 6855 ALOGV("constructor %p", this); 6856 6857 if (client == 0) { 6858 return; 6859 } 6860 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6861 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6862 if (mCblkMemory != 0) { 6863 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6864 6865 if (mCblk) { 6866 new(mCblk) effect_param_cblk_t(); 6867 mBuffer = (uint8_t *)mCblk + bufOffset; 6868 } 6869 } else { 6870 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6871 return; 6872 } 6873} 6874 6875AudioFlinger::EffectHandle::~EffectHandle() 6876{ 6877 ALOGV("Destructor %p", this); 6878 disconnect(false); 6879 ALOGV("Destructor DONE %p", this); 6880} 6881 6882status_t AudioFlinger::EffectHandle::enable() 6883{ 6884 ALOGV("enable %p", this); 6885 if (!mHasControl) return INVALID_OPERATION; 6886 if (mEffect == 0) return DEAD_OBJECT; 6887 6888 if (mEnabled) { 6889 return NO_ERROR; 6890 } 6891 6892 mEnabled = true; 6893 6894 sp<ThreadBase> thread = mEffect->thread().promote(); 6895 if (thread != 0) { 6896 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6897 } 6898 6899 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6900 if (mEffect->suspended()) { 6901 return NO_ERROR; 6902 } 6903 6904 status_t status = mEffect->setEnabled(true); 6905 if (status != NO_ERROR) { 6906 if (thread != 0) { 6907 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6908 } 6909 mEnabled = false; 6910 } 6911 return status; 6912} 6913 6914status_t AudioFlinger::EffectHandle::disable() 6915{ 6916 ALOGV("disable %p", this); 6917 if (!mHasControl) return INVALID_OPERATION; 6918 if (mEffect == 0) return DEAD_OBJECT; 6919 6920 if (!mEnabled) { 6921 return NO_ERROR; 6922 } 6923 mEnabled = false; 6924 6925 if (mEffect->suspended()) { 6926 return NO_ERROR; 6927 } 6928 6929 status_t status = mEffect->setEnabled(false); 6930 6931 sp<ThreadBase> thread = mEffect->thread().promote(); 6932 if (thread != 0) { 6933 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6934 } 6935 6936 return status; 6937} 6938 6939void AudioFlinger::EffectHandle::disconnect() 6940{ 6941 disconnect(true); 6942} 6943 6944void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6945{ 6946 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6947 if (mEffect == 0) { 6948 return; 6949 } 6950 mEffect->disconnect(this, unpiniflast); 6951 6952 if (mHasControl && mEnabled) { 6953 sp<ThreadBase> thread = mEffect->thread().promote(); 6954 if (thread != 0) { 6955 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6956 } 6957 } 6958 6959 // release sp on module => module destructor can be called now 6960 mEffect.clear(); 6961 if (mClient != 0) { 6962 if (mCblk) { 6963 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6964 } 6965 mCblkMemory.clear(); // and free the shared memory 6966 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6967 mClient.clear(); 6968 } 6969} 6970 6971status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6972 uint32_t cmdSize, 6973 void *pCmdData, 6974 uint32_t *replySize, 6975 void *pReplyData) 6976{ 6977// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6978// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6979 6980 // only get parameter command is permitted for applications not controlling the effect 6981 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6982 return INVALID_OPERATION; 6983 } 6984 if (mEffect == 0) return DEAD_OBJECT; 6985 if (mClient == 0) return INVALID_OPERATION; 6986 6987 // handle commands that are not forwarded transparently to effect engine 6988 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6989 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6990 // no risk to block the whole media server process or mixer threads is we are stuck here 6991 Mutex::Autolock _l(mCblk->lock); 6992 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6993 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6994 mCblk->serverIndex = 0; 6995 mCblk->clientIndex = 0; 6996 return BAD_VALUE; 6997 } 6998 status_t status = NO_ERROR; 6999 while (mCblk->serverIndex < mCblk->clientIndex) { 7000 int reply; 7001 uint32_t rsize = sizeof(int); 7002 int *p = (int *)(mBuffer + mCblk->serverIndex); 7003 int size = *p++; 7004 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7005 ALOGW("command(): invalid parameter block size"); 7006 break; 7007 } 7008 effect_param_t *param = (effect_param_t *)p; 7009 if (param->psize == 0 || param->vsize == 0) { 7010 ALOGW("command(): null parameter or value size"); 7011 mCblk->serverIndex += size; 7012 continue; 7013 } 7014 uint32_t psize = sizeof(effect_param_t) + 7015 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7016 param->vsize; 7017 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7018 psize, 7019 p, 7020 &rsize, 7021 &reply); 7022 // stop at first error encountered 7023 if (ret != NO_ERROR) { 7024 status = ret; 7025 *(int *)pReplyData = reply; 7026 break; 7027 } else if (reply != NO_ERROR) { 7028 *(int *)pReplyData = reply; 7029 break; 7030 } 7031 mCblk->serverIndex += size; 7032 } 7033 mCblk->serverIndex = 0; 7034 mCblk->clientIndex = 0; 7035 return status; 7036 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7037 *(int *)pReplyData = NO_ERROR; 7038 return enable(); 7039 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7040 *(int *)pReplyData = NO_ERROR; 7041 return disable(); 7042 } 7043 7044 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7045} 7046 7047sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7048 return mCblkMemory; 7049} 7050 7051void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7052{ 7053 ALOGV("setControl %p control %d", this, hasControl); 7054 7055 mHasControl = hasControl; 7056 mEnabled = enabled; 7057 7058 if (signal && mEffectClient != 0) { 7059 mEffectClient->controlStatusChanged(hasControl); 7060 } 7061} 7062 7063void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7064 uint32_t cmdSize, 7065 void *pCmdData, 7066 uint32_t replySize, 7067 void *pReplyData) 7068{ 7069 if (mEffectClient != 0) { 7070 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7071 } 7072} 7073 7074 7075 7076void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7077{ 7078 if (mEffectClient != 0) { 7079 mEffectClient->enableStatusChanged(enabled); 7080 } 7081} 7082 7083status_t AudioFlinger::EffectHandle::onTransact( 7084 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7085{ 7086 return BnEffect::onTransact(code, data, reply, flags); 7087} 7088 7089 7090void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7091{ 7092 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7093 7094 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7095 (mClient == NULL) ? getpid() : mClient->pid(), 7096 mPriority, 7097 mHasControl, 7098 !locked, 7099 mCblk ? mCblk->clientIndex : 0, 7100 mCblk ? mCblk->serverIndex : 0 7101 ); 7102 7103 if (locked) { 7104 mCblk->lock.unlock(); 7105 } 7106} 7107 7108#undef LOG_TAG 7109#define LOG_TAG "AudioFlinger::EffectChain" 7110 7111AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7112 int sessionId) 7113 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7114 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7115 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7116{ 7117 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7118 sp<ThreadBase> thread = mThread.promote(); 7119 if (thread == 0) { 7120 return; 7121 } 7122 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7123 thread->frameCount(); 7124} 7125 7126AudioFlinger::EffectChain::~EffectChain() 7127{ 7128 if (mOwnInBuffer) { 7129 delete mInBuffer; 7130 } 7131 7132} 7133 7134// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7135sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7136{ 7137 sp<EffectModule> effect; 7138 size_t size = mEffects.size(); 7139 7140 for (size_t i = 0; i < size; i++) { 7141 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7142 effect = mEffects[i]; 7143 break; 7144 } 7145 } 7146 return effect; 7147} 7148 7149// getEffectFromId_l() must be called with ThreadBase::mLock held 7150sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7151{ 7152 sp<EffectModule> effect; 7153 size_t size = mEffects.size(); 7154 7155 for (size_t i = 0; i < size; i++) { 7156 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7157 if (id == 0 || mEffects[i]->id() == id) { 7158 effect = mEffects[i]; 7159 break; 7160 } 7161 } 7162 return effect; 7163} 7164 7165// getEffectFromType_l() must be called with ThreadBase::mLock held 7166sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7167 const effect_uuid_t *type) 7168{ 7169 sp<EffectModule> effect; 7170 size_t size = mEffects.size(); 7171 7172 for (size_t i = 0; i < size; i++) { 7173 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7174 effect = mEffects[i]; 7175 break; 7176 } 7177 } 7178 return effect; 7179} 7180 7181// Must be called with EffectChain::mLock locked 7182void AudioFlinger::EffectChain::process_l() 7183{ 7184 sp<ThreadBase> thread = mThread.promote(); 7185 if (thread == 0) { 7186 ALOGW("process_l(): cannot promote mixer thread"); 7187 return; 7188 } 7189 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7190 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7191 // always process effects unless no more tracks are on the session and the effect tail 7192 // has been rendered 7193 bool doProcess = true; 7194 if (!isGlobalSession) { 7195 bool tracksOnSession = (trackCnt() != 0); 7196 7197 if (!tracksOnSession && mTailBufferCount == 0) { 7198 doProcess = false; 7199 } 7200 7201 if (activeTrackCnt() == 0) { 7202 // if no track is active and the effect tail has not been rendered, 7203 // the input buffer must be cleared here as the mixer process will not do it 7204 if (tracksOnSession || mTailBufferCount > 0) { 7205 size_t numSamples = thread->frameCount() * thread->channelCount(); 7206 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7207 if (mTailBufferCount > 0) { 7208 mTailBufferCount--; 7209 } 7210 } 7211 } 7212 } 7213 7214 size_t size = mEffects.size(); 7215 if (doProcess) { 7216 for (size_t i = 0; i < size; i++) { 7217 mEffects[i]->process(); 7218 } 7219 } 7220 for (size_t i = 0; i < size; i++) { 7221 mEffects[i]->updateState(); 7222 } 7223} 7224 7225// addEffect_l() must be called with PlaybackThread::mLock held 7226status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7227{ 7228 effect_descriptor_t desc = effect->desc(); 7229 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7230 7231 Mutex::Autolock _l(mLock); 7232 effect->setChain(this); 7233 sp<ThreadBase> thread = mThread.promote(); 7234 if (thread == 0) { 7235 return NO_INIT; 7236 } 7237 effect->setThread(thread); 7238 7239 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7240 // Auxiliary effects are inserted at the beginning of mEffects vector as 7241 // they are processed first and accumulated in chain input buffer 7242 mEffects.insertAt(effect, 0); 7243 7244 // the input buffer for auxiliary effect contains mono samples in 7245 // 32 bit format. This is to avoid saturation in AudoMixer 7246 // accumulation stage. Saturation is done in EffectModule::process() before 7247 // calling the process in effect engine 7248 size_t numSamples = thread->frameCount(); 7249 int32_t *buffer = new int32_t[numSamples]; 7250 memset(buffer, 0, numSamples * sizeof(int32_t)); 7251 effect->setInBuffer((int16_t *)buffer); 7252 // auxiliary effects output samples to chain input buffer for further processing 7253 // by insert effects 7254 effect->setOutBuffer(mInBuffer); 7255 } else { 7256 // Insert effects are inserted at the end of mEffects vector as they are processed 7257 // after track and auxiliary effects. 7258 // Insert effect order as a function of indicated preference: 7259 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7260 // another effect is present 7261 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7262 // last effect claiming first position 7263 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7264 // first effect claiming last position 7265 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7266 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7267 // already present 7268 7269 int size = (int)mEffects.size(); 7270 int idx_insert = size; 7271 int idx_insert_first = -1; 7272 int idx_insert_last = -1; 7273 7274 for (int i = 0; i < size; i++) { 7275 effect_descriptor_t d = mEffects[i]->desc(); 7276 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7277 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7278 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7279 // check invalid effect chaining combinations 7280 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7281 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7282 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7283 return INVALID_OPERATION; 7284 } 7285 // remember position of first insert effect and by default 7286 // select this as insert position for new effect 7287 if (idx_insert == size) { 7288 idx_insert = i; 7289 } 7290 // remember position of last insert effect claiming 7291 // first position 7292 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7293 idx_insert_first = i; 7294 } 7295 // remember position of first insert effect claiming 7296 // last position 7297 if (iPref == EFFECT_FLAG_INSERT_LAST && 7298 idx_insert_last == -1) { 7299 idx_insert_last = i; 7300 } 7301 } 7302 } 7303 7304 // modify idx_insert from first position if needed 7305 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7306 if (idx_insert_last != -1) { 7307 idx_insert = idx_insert_last; 7308 } else { 7309 idx_insert = size; 7310 } 7311 } else { 7312 if (idx_insert_first != -1) { 7313 idx_insert = idx_insert_first + 1; 7314 } 7315 } 7316 7317 // always read samples from chain input buffer 7318 effect->setInBuffer(mInBuffer); 7319 7320 // if last effect in the chain, output samples to chain 7321 // output buffer, otherwise to chain input buffer 7322 if (idx_insert == size) { 7323 if (idx_insert != 0) { 7324 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7325 mEffects[idx_insert-1]->configure(); 7326 } 7327 effect->setOutBuffer(mOutBuffer); 7328 } else { 7329 effect->setOutBuffer(mInBuffer); 7330 } 7331 mEffects.insertAt(effect, idx_insert); 7332 7333 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7334 } 7335 effect->configure(); 7336 return NO_ERROR; 7337} 7338 7339// removeEffect_l() must be called with PlaybackThread::mLock held 7340size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7341{ 7342 Mutex::Autolock _l(mLock); 7343 int size = (int)mEffects.size(); 7344 int i; 7345 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7346 7347 for (i = 0; i < size; i++) { 7348 if (effect == mEffects[i]) { 7349 // calling stop here will remove pre-processing effect from the audio HAL. 7350 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7351 // the middle of a read from audio HAL 7352 if (mEffects[i]->state() == EffectModule::ACTIVE || 7353 mEffects[i]->state() == EffectModule::STOPPING) { 7354 mEffects[i]->stop(); 7355 } 7356 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7357 delete[] effect->inBuffer(); 7358 } else { 7359 if (i == size - 1 && i != 0) { 7360 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7361 mEffects[i - 1]->configure(); 7362 } 7363 } 7364 mEffects.removeAt(i); 7365 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7366 break; 7367 } 7368 } 7369 7370 return mEffects.size(); 7371} 7372 7373// setDevice_l() must be called with PlaybackThread::mLock held 7374void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7375{ 7376 size_t size = mEffects.size(); 7377 for (size_t i = 0; i < size; i++) { 7378 mEffects[i]->setDevice(device); 7379 } 7380} 7381 7382// setMode_l() must be called with PlaybackThread::mLock held 7383void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7384{ 7385 size_t size = mEffects.size(); 7386 for (size_t i = 0; i < size; i++) { 7387 mEffects[i]->setMode(mode); 7388 } 7389} 7390 7391// setVolume_l() must be called with PlaybackThread::mLock held 7392bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7393{ 7394 uint32_t newLeft = *left; 7395 uint32_t newRight = *right; 7396 bool hasControl = false; 7397 int ctrlIdx = -1; 7398 size_t size = mEffects.size(); 7399 7400 // first update volume controller 7401 for (size_t i = size; i > 0; i--) { 7402 if (mEffects[i - 1]->isProcessEnabled() && 7403 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7404 ctrlIdx = i - 1; 7405 hasControl = true; 7406 break; 7407 } 7408 } 7409 7410 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7411 if (hasControl) { 7412 *left = mNewLeftVolume; 7413 *right = mNewRightVolume; 7414 } 7415 return hasControl; 7416 } 7417 7418 mVolumeCtrlIdx = ctrlIdx; 7419 mLeftVolume = newLeft; 7420 mRightVolume = newRight; 7421 7422 // second get volume update from volume controller 7423 if (ctrlIdx >= 0) { 7424 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7425 mNewLeftVolume = newLeft; 7426 mNewRightVolume = newRight; 7427 } 7428 // then indicate volume to all other effects in chain. 7429 // Pass altered volume to effects before volume controller 7430 // and requested volume to effects after controller 7431 uint32_t lVol = newLeft; 7432 uint32_t rVol = newRight; 7433 7434 for (size_t i = 0; i < size; i++) { 7435 if ((int)i == ctrlIdx) continue; 7436 // this also works for ctrlIdx == -1 when there is no volume controller 7437 if ((int)i > ctrlIdx) { 7438 lVol = *left; 7439 rVol = *right; 7440 } 7441 mEffects[i]->setVolume(&lVol, &rVol, false); 7442 } 7443 *left = newLeft; 7444 *right = newRight; 7445 7446 return hasControl; 7447} 7448 7449status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7450{ 7451 const size_t SIZE = 256; 7452 char buffer[SIZE]; 7453 String8 result; 7454 7455 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7456 result.append(buffer); 7457 7458 bool locked = tryLock(mLock); 7459 // failed to lock - AudioFlinger is probably deadlocked 7460 if (!locked) { 7461 result.append("\tCould not lock mutex:\n"); 7462 } 7463 7464 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7465 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7466 mEffects.size(), 7467 (uint32_t)mInBuffer, 7468 (uint32_t)mOutBuffer, 7469 mActiveTrackCnt); 7470 result.append(buffer); 7471 write(fd, result.string(), result.size()); 7472 7473 for (size_t i = 0; i < mEffects.size(); ++i) { 7474 sp<EffectModule> effect = mEffects[i]; 7475 if (effect != 0) { 7476 effect->dump(fd, args); 7477 } 7478 } 7479 7480 if (locked) { 7481 mLock.unlock(); 7482 } 7483 7484 return NO_ERROR; 7485} 7486 7487// must be called with ThreadBase::mLock held 7488void AudioFlinger::EffectChain::setEffectSuspended_l( 7489 const effect_uuid_t *type, bool suspend) 7490{ 7491 sp<SuspendedEffectDesc> desc; 7492 // use effect type UUID timelow as key as there is no real risk of identical 7493 // timeLow fields among effect type UUIDs. 7494 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7495 if (suspend) { 7496 if (index >= 0) { 7497 desc = mSuspendedEffects.valueAt(index); 7498 } else { 7499 desc = new SuspendedEffectDesc(); 7500 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7501 mSuspendedEffects.add(type->timeLow, desc); 7502 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7503 } 7504 if (desc->mRefCount++ == 0) { 7505 sp<EffectModule> effect = getEffectIfEnabled(type); 7506 if (effect != 0) { 7507 desc->mEffect = effect; 7508 effect->setSuspended(true); 7509 effect->setEnabled(false); 7510 } 7511 } 7512 } else { 7513 if (index < 0) { 7514 return; 7515 } 7516 desc = mSuspendedEffects.valueAt(index); 7517 if (desc->mRefCount <= 0) { 7518 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7519 desc->mRefCount = 1; 7520 } 7521 if (--desc->mRefCount == 0) { 7522 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7523 if (desc->mEffect != 0) { 7524 sp<EffectModule> effect = desc->mEffect.promote(); 7525 if (effect != 0) { 7526 effect->setSuspended(false); 7527 sp<EffectHandle> handle = effect->controlHandle(); 7528 if (handle != 0) { 7529 effect->setEnabled(handle->enabled()); 7530 } 7531 } 7532 desc->mEffect.clear(); 7533 } 7534 mSuspendedEffects.removeItemsAt(index); 7535 } 7536 } 7537} 7538 7539// must be called with ThreadBase::mLock held 7540void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7541{ 7542 sp<SuspendedEffectDesc> desc; 7543 7544 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7545 if (suspend) { 7546 if (index >= 0) { 7547 desc = mSuspendedEffects.valueAt(index); 7548 } else { 7549 desc = new SuspendedEffectDesc(); 7550 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7551 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7552 } 7553 if (desc->mRefCount++ == 0) { 7554 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7555 for (size_t i = 0; i < effects.size(); i++) { 7556 setEffectSuspended_l(&effects[i]->desc().type, true); 7557 } 7558 } 7559 } else { 7560 if (index < 0) { 7561 return; 7562 } 7563 desc = mSuspendedEffects.valueAt(index); 7564 if (desc->mRefCount <= 0) { 7565 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7566 desc->mRefCount = 1; 7567 } 7568 if (--desc->mRefCount == 0) { 7569 Vector<const effect_uuid_t *> types; 7570 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7571 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7572 continue; 7573 } 7574 types.add(&mSuspendedEffects.valueAt(i)->mType); 7575 } 7576 for (size_t i = 0; i < types.size(); i++) { 7577 setEffectSuspended_l(types[i], false); 7578 } 7579 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7580 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7581 } 7582 } 7583} 7584 7585 7586// The volume effect is used for automated tests only 7587#ifndef OPENSL_ES_H_ 7588static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7589 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7590const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7591#endif //OPENSL_ES_H_ 7592 7593bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7594{ 7595 // auxiliary effects and visualizer are never suspended on output mix 7596 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7597 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7598 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7599 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7600 return false; 7601 } 7602 return true; 7603} 7604 7605Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7606{ 7607 Vector< sp<EffectModule> > effects; 7608 for (size_t i = 0; i < mEffects.size(); i++) { 7609 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7610 continue; 7611 } 7612 effects.add(mEffects[i]); 7613 } 7614 return effects; 7615} 7616 7617sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7618 const effect_uuid_t *type) 7619{ 7620 sp<EffectModule> effect; 7621 effect = getEffectFromType_l(type); 7622 if (effect != 0 && !effect->isEnabled()) { 7623 effect.clear(); 7624 } 7625 return effect; 7626} 7627 7628void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7629 bool enabled) 7630{ 7631 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7632 if (enabled) { 7633 if (index < 0) { 7634 // if the effect is not suspend check if all effects are suspended 7635 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7636 if (index < 0) { 7637 return; 7638 } 7639 if (!isEffectEligibleForSuspend(effect->desc())) { 7640 return; 7641 } 7642 setEffectSuspended_l(&effect->desc().type, enabled); 7643 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7644 if (index < 0) { 7645 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7646 return; 7647 } 7648 } 7649 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7650 effect->desc().type.timeLow); 7651 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7652 // if effect is requested to suspended but was not yet enabled, supend it now. 7653 if (desc->mEffect == 0) { 7654 desc->mEffect = effect; 7655 effect->setEnabled(false); 7656 effect->setSuspended(true); 7657 } 7658 } else { 7659 if (index < 0) { 7660 return; 7661 } 7662 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7663 effect->desc().type.timeLow); 7664 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7665 desc->mEffect.clear(); 7666 effect->setSuspended(false); 7667 } 7668} 7669 7670#undef LOG_TAG 7671#define LOG_TAG "AudioFlinger" 7672 7673// ---------------------------------------------------------------------------- 7674 7675status_t AudioFlinger::onTransact( 7676 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7677{ 7678 return BnAudioFlinger::onTransact(code, data, reply, flags); 7679} 7680 7681}; // namespace android 7682