AudioFlinger.cpp revision f78aee70d15daf4690de7e7b4983ee68b0d1381d
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/AudioTrack.h> 41#include <media/AudioRecord.h> 42#include <media/IMediaPlayerService.h> 43#include <media/IMediaDeathNotifier.h> 44 45#include <private/media/AudioTrackShared.h> 46#include <private/media/AudioEffectShared.h> 47 48#include <system/audio.h> 49#include <hardware/audio.h> 50 51#include "AudioMixer.h" 52#include "AudioFlinger.h" 53 54#include <media/EffectsFactoryApi.h> 55#include <audio_effects/effect_visualizer.h> 56#include <audio_effects/effect_ns.h> 57#include <audio_effects/effect_aec.h> 58 59#include <audio_utils/primitives.h> 60 61#include <cpustats/ThreadCpuUsage.h> 62#include <powermanager/PowerManager.h> 63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 64 65// ---------------------------------------------------------------------------- 66 67 68namespace android { 69 70static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 71static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 72 73//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 74static const float MAX_GAIN = 4096.0f; 75static const float MAX_GAIN_INT = 0x1000; 76 77// retry counts for buffer fill timeout 78// 50 * ~20msecs = 1 second 79static const int8_t kMaxTrackRetries = 50; 80static const int8_t kMaxTrackStartupRetries = 50; 81// allow less retry attempts on direct output thread. 82// direct outputs can be a scarce resource in audio hardware and should 83// be released as quickly as possible. 84static const int8_t kMaxTrackRetriesDirect = 2; 85 86static const int kDumpLockRetries = 50; 87static const int kDumpLockSleepUs = 20000; 88 89// don't warn about blocked writes or record buffer overflows more often than this 90static const nsecs_t kWarningThrottleNs = seconds(5); 91 92// RecordThread loop sleep time upon application overrun or audio HAL read error 93static const int kRecordThreadSleepUs = 5000; 94 95// maximum time to wait for setParameters to complete 96static const nsecs_t kSetParametersTimeoutNs = seconds(2); 97 98// minimum sleep time for the mixer thread loop when tracks are active but in underrun 99static const uint32_t kMinThreadSleepTimeUs = 5000; 100// maximum divider applied to the active sleep time in the mixer thread loop 101static const uint32_t kMaxThreadSleepTimeShift = 2; 102 103 104// ---------------------------------------------------------------------------- 105 106static bool recordingAllowed() { 107 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 108 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 109 if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO"); 110 return ok; 111} 112 113static bool settingsAllowed() { 114 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 115 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 116 if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 117 return ok; 118} 119 120// To collect the amplifier usage 121static void addBatteryData(uint32_t params) { 122 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 123 if (service == NULL) { 124 // it already logged 125 return; 126 } 127 128 service->addBatteryData(params); 129} 130 131static int load_audio_interface(const char *if_name, const hw_module_t **mod, 132 audio_hw_device_t **dev) 133{ 134 int rc; 135 136 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 137 if (rc) 138 goto out; 139 140 rc = audio_hw_device_open(*mod, dev); 141 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 142 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 143 if (rc) 144 goto out; 145 146 return 0; 147 148out: 149 *mod = NULL; 150 *dev = NULL; 151 return rc; 152} 153 154static const char * const audio_interfaces[] = { 155 "primary", 156 "a2dp", 157 "usb", 158}; 159#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 160 161// ---------------------------------------------------------------------------- 162 163AudioFlinger::AudioFlinger() 164 : BnAudioFlinger(), 165 mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 166 mBtNrecIsOff(false) 167{ 168} 169 170void AudioFlinger::onFirstRef() 171{ 172 int rc = 0; 173 174 Mutex::Autolock _l(mLock); 175 176 /* TODO: move all this work into an Init() function */ 177 mHardwareStatus = AUDIO_HW_IDLE; 178 179 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 180 const hw_module_t *mod; 181 audio_hw_device_t *dev; 182 183 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 184 if (rc) 185 continue; 186 187 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 188 mod->name, mod->id); 189 mAudioHwDevs.push(dev); 190 191 if (!mPrimaryHardwareDev) { 192 mPrimaryHardwareDev = dev; 193 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 194 mod->name, mod->id, audio_interfaces[i]); 195 } 196 } 197 198 mHardwareStatus = AUDIO_HW_INIT; 199 200 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 201 ALOGE("Primary audio interface not found"); 202 return; 203 } 204 205 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 206 audio_hw_device_t *dev = mAudioHwDevs[i]; 207 208 mHardwareStatus = AUDIO_HW_INIT; 209 rc = dev->init_check(dev); 210 if (rc == 0) { 211 AutoMutex lock(mHardwareLock); 212 213 mMode = AUDIO_MODE_NORMAL; 214 mHardwareStatus = AUDIO_HW_SET_MODE; 215 dev->set_mode(dev, mMode); 216 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 217 dev->set_master_volume(dev, 1.0f); 218 mHardwareStatus = AUDIO_HW_IDLE; 219 } 220 } 221} 222 223status_t AudioFlinger::initCheck() const 224{ 225 Mutex::Autolock _l(mLock); 226 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 227 return NO_INIT; 228 return NO_ERROR; 229} 230 231AudioFlinger::~AudioFlinger() 232{ 233 int num_devs = mAudioHwDevs.size(); 234 235 while (!mRecordThreads.isEmpty()) { 236 // closeInput() will remove first entry from mRecordThreads 237 closeInput(mRecordThreads.keyAt(0)); 238 } 239 while (!mPlaybackThreads.isEmpty()) { 240 // closeOutput() will remove first entry from mPlaybackThreads 241 closeOutput(mPlaybackThreads.keyAt(0)); 242 } 243 244 for (int i = 0; i < num_devs; i++) { 245 audio_hw_device_t *dev = mAudioHwDevs[i]; 246 audio_hw_device_close(dev); 247 } 248 mAudioHwDevs.clear(); 249} 250 251audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 252{ 253 /* first matching HW device is returned */ 254 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 255 audio_hw_device_t *dev = mAudioHwDevs[i]; 256 if ((dev->get_supported_devices(dev) & devices) == devices) 257 return dev; 258 } 259 return NULL; 260} 261 262status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 263{ 264 const size_t SIZE = 256; 265 char buffer[SIZE]; 266 String8 result; 267 268 result.append("Clients:\n"); 269 for (size_t i = 0; i < mClients.size(); ++i) { 270 wp<Client> wClient = mClients.valueAt(i); 271 if (wClient != 0) { 272 sp<Client> client = wClient.promote(); 273 if (client != 0) { 274 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 275 result.append(buffer); 276 } 277 } 278 } 279 280 result.append("Global session refs:\n"); 281 result.append(" session pid cnt\n"); 282 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 283 AudioSessionRef *r = mAudioSessionRefs[i]; 284 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 285 result.append(buffer); 286 } 287 write(fd, result.string(), result.size()); 288 return NO_ERROR; 289} 290 291 292status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 293{ 294 const size_t SIZE = 256; 295 char buffer[SIZE]; 296 String8 result; 297 hardware_call_state hardwareStatus = mHardwareStatus; 298 299 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 300 result.append(buffer); 301 write(fd, result.string(), result.size()); 302 return NO_ERROR; 303} 304 305status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 306{ 307 const size_t SIZE = 256; 308 char buffer[SIZE]; 309 String8 result; 310 snprintf(buffer, SIZE, "Permission Denial: " 311 "can't dump AudioFlinger from pid=%d, uid=%d\n", 312 IPCThreadState::self()->getCallingPid(), 313 IPCThreadState::self()->getCallingUid()); 314 result.append(buffer); 315 write(fd, result.string(), result.size()); 316 return NO_ERROR; 317} 318 319static bool tryLock(Mutex& mutex) 320{ 321 bool locked = false; 322 for (int i = 0; i < kDumpLockRetries; ++i) { 323 if (mutex.tryLock() == NO_ERROR) { 324 locked = true; 325 break; 326 } 327 usleep(kDumpLockSleepUs); 328 } 329 return locked; 330} 331 332status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 333{ 334 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 335 dumpPermissionDenial(fd, args); 336 } else { 337 // get state of hardware lock 338 bool hardwareLocked = tryLock(mHardwareLock); 339 if (!hardwareLocked) { 340 String8 result(kHardwareLockedString); 341 write(fd, result.string(), result.size()); 342 } else { 343 mHardwareLock.unlock(); 344 } 345 346 bool locked = tryLock(mLock); 347 348 // failed to lock - AudioFlinger is probably deadlocked 349 if (!locked) { 350 String8 result(kDeadlockedString); 351 write(fd, result.string(), result.size()); 352 } 353 354 dumpClients(fd, args); 355 dumpInternals(fd, args); 356 357 // dump playback threads 358 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 359 mPlaybackThreads.valueAt(i)->dump(fd, args); 360 } 361 362 // dump record threads 363 for (size_t i = 0; i < mRecordThreads.size(); i++) { 364 mRecordThreads.valueAt(i)->dump(fd, args); 365 } 366 367 // dump all hardware devs 368 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 369 audio_hw_device_t *dev = mAudioHwDevs[i]; 370 dev->dump(dev, fd); 371 } 372 if (locked) mLock.unlock(); 373 } 374 return NO_ERROR; 375} 376 377 378// IAudioFlinger interface 379 380 381sp<IAudioTrack> AudioFlinger::createTrack( 382 pid_t pid, 383 int streamType, 384 uint32_t sampleRate, 385 uint32_t format, 386 uint32_t channelMask, 387 int frameCount, 388 uint32_t flags, 389 const sp<IMemory>& sharedBuffer, 390 int output, 391 int *sessionId, 392 status_t *status) 393{ 394 sp<PlaybackThread::Track> track; 395 sp<TrackHandle> trackHandle; 396 sp<Client> client; 397 wp<Client> wclient; 398 status_t lStatus; 399 int lSessionId; 400 401 if (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 503uint32_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 0; 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 return mMasterVolume; 650} 651 652bool AudioFlinger::masterMute() const 653{ 654 return mMasterMute; 655} 656 657status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 658{ 659 // check calling permissions 660 if (!settingsAllowed()) { 661 return PERMISSION_DENIED; 662 } 663 664 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 665 ALOGE("setStreamVolume() invalid stream %d", stream); 666 return BAD_VALUE; 667 } 668 669 AutoMutex lock(mLock); 670 PlaybackThread *thread = NULL; 671 if (output) { 672 thread = checkPlaybackThread_l(output); 673 if (thread == NULL) { 674 return BAD_VALUE; 675 } 676 } 677 678 mStreamTypes[stream].volume = value; 679 680 if (thread == NULL) { 681 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 682 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 683 } 684 } else { 685 thread->setStreamVolume(stream, value); 686 } 687 688 return NO_ERROR; 689} 690 691status_t AudioFlinger::setStreamMute(int stream, bool muted) 692{ 693 // check calling permissions 694 if (!settingsAllowed()) { 695 return PERMISSION_DENIED; 696 } 697 698 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || 699 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 700 ALOGE("setStreamMute() invalid stream %d", stream); 701 return BAD_VALUE; 702 } 703 704 AutoMutex lock(mLock); 705 mStreamTypes[stream].mute = muted; 706 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 707 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 708 709 return NO_ERROR; 710} 711 712float AudioFlinger::streamVolume(int stream, int output) const 713{ 714 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 715 return 0.0f; 716 } 717 718 AutoMutex lock(mLock); 719 float volume; 720 if (output) { 721 PlaybackThread *thread = checkPlaybackThread_l(output); 722 if (thread == NULL) { 723 return 0.0f; 724 } 725 volume = thread->streamVolume(stream); 726 } else { 727 volume = mStreamTypes[stream].volume; 728 } 729 730 return volume; 731} 732 733bool AudioFlinger::streamMute(int stream) const 734{ 735 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 736 return true; 737 } 738 739 return mStreamTypes[stream].mute; 740} 741 742status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 743{ 744 status_t result; 745 746 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 747 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 748 // check calling permissions 749 if (!settingsAllowed()) { 750 return PERMISSION_DENIED; 751 } 752 753 // ioHandle == 0 means the parameters are global to the audio hardware interface 754 if (ioHandle == 0) { 755 AutoMutex lock(mHardwareLock); 756 mHardwareStatus = AUDIO_SET_PARAMETER; 757 status_t final_result = NO_ERROR; 758 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 759 audio_hw_device_t *dev = mAudioHwDevs[i]; 760 result = dev->set_parameters(dev, keyValuePairs.string()); 761 final_result = result ?: final_result; 762 } 763 mHardwareStatus = AUDIO_HW_IDLE; 764 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 765 AudioParameter param = AudioParameter(keyValuePairs); 766 String8 value; 767 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 768 Mutex::Autolock _l(mLock); 769 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 770 if (mBtNrecIsOff != btNrecIsOff) { 771 for (size_t i = 0; i < mRecordThreads.size(); i++) { 772 sp<RecordThread> thread = mRecordThreads.valueAt(i); 773 RecordThread::RecordTrack *track = thread->track(); 774 if (track != NULL) { 775 audio_devices_t device = (audio_devices_t)( 776 thread->device() & AUDIO_DEVICE_IN_ALL); 777 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 778 thread->setEffectSuspended(FX_IID_AEC, 779 suspend, 780 track->sessionId()); 781 thread->setEffectSuspended(FX_IID_NS, 782 suspend, 783 track->sessionId()); 784 } 785 } 786 mBtNrecIsOff = btNrecIsOff; 787 } 788 } 789 return final_result; 790 } 791 792 // hold a strong ref on thread in case closeOutput() or closeInput() is called 793 // and the thread is exited once the lock is released 794 sp<ThreadBase> thread; 795 { 796 Mutex::Autolock _l(mLock); 797 thread = checkPlaybackThread_l(ioHandle); 798 if (thread == NULL) { 799 thread = checkRecordThread_l(ioHandle); 800 } else if (thread.get() == primaryPlaybackThread_l()) { 801 // indicate output device change to all input threads for pre processing 802 AudioParameter param = AudioParameter(keyValuePairs); 803 int value; 804 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 805 for (size_t i = 0; i < mRecordThreads.size(); i++) { 806 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 807 } 808 } 809 } 810 } 811 if (thread != NULL) { 812 result = thread->setParameters(keyValuePairs); 813 return result; 814 } 815 return BAD_VALUE; 816} 817 818String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 819{ 820// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 821// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 822 823 if (ioHandle == 0) { 824 String8 out_s8; 825 826 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 827 audio_hw_device_t *dev = mAudioHwDevs[i]; 828 char *s = dev->get_parameters(dev, keys.string()); 829 out_s8 += String8(s); 830 free(s); 831 } 832 return out_s8; 833 } 834 835 Mutex::Autolock _l(mLock); 836 837 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 838 if (playbackThread != NULL) { 839 return playbackThread->getParameters(keys); 840 } 841 RecordThread *recordThread = checkRecordThread_l(ioHandle); 842 if (recordThread != NULL) { 843 return recordThread->getParameters(keys); 844 } 845 return String8(""); 846} 847 848size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 849{ 850 status_t ret = initCheck(); 851 if (ret != NO_ERROR) { 852 return 0; 853 } 854 855 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 856} 857 858unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 859{ 860 if (ioHandle == 0) { 861 return 0; 862 } 863 864 Mutex::Autolock _l(mLock); 865 866 RecordThread *recordThread = checkRecordThread_l(ioHandle); 867 if (recordThread != NULL) { 868 return recordThread->getInputFramesLost(); 869 } 870 return 0; 871} 872 873status_t AudioFlinger::setVoiceVolume(float value) 874{ 875 status_t ret = initCheck(); 876 if (ret != NO_ERROR) { 877 return ret; 878 } 879 880 // check calling permissions 881 if (!settingsAllowed()) { 882 return PERMISSION_DENIED; 883 } 884 885 AutoMutex lock(mHardwareLock); 886 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 887 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 888 mHardwareStatus = AUDIO_HW_IDLE; 889 890 return ret; 891} 892 893status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 894{ 895 status_t status; 896 897 Mutex::Autolock _l(mLock); 898 899 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 900 if (playbackThread != NULL) { 901 return playbackThread->getRenderPosition(halFrames, dspFrames); 902 } 903 904 return BAD_VALUE; 905} 906 907void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 908{ 909 910 Mutex::Autolock _l(mLock); 911 912 int pid = IPCThreadState::self()->getCallingPid(); 913 if (mNotificationClients.indexOfKey(pid) < 0) { 914 sp<NotificationClient> notificationClient = new NotificationClient(this, 915 client, 916 pid); 917 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 918 919 mNotificationClients.add(pid, notificationClient); 920 921 sp<IBinder> binder = client->asBinder(); 922 binder->linkToDeath(notificationClient); 923 924 // the config change is always sent from playback or record threads to avoid deadlock 925 // with AudioSystem::gLock 926 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 927 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 928 } 929 930 for (size_t i = 0; i < mRecordThreads.size(); i++) { 931 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 932 } 933 } 934} 935 936void AudioFlinger::removeNotificationClient(pid_t pid) 937{ 938 Mutex::Autolock _l(mLock); 939 940 int index = mNotificationClients.indexOfKey(pid); 941 if (index >= 0) { 942 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 943 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 944 mNotificationClients.removeItem(pid); 945 } 946 947 ALOGV("%d died, releasing its sessions", pid); 948 int num = mAudioSessionRefs.size(); 949 bool removed = false; 950 for (int i = 0; i< num; i++) { 951 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 952 ALOGV(" pid %d @ %d", ref->pid, i); 953 if (ref->pid == pid) { 954 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 955 mAudioSessionRefs.removeAt(i); 956 delete ref; 957 removed = true; 958 i--; 959 num--; 960 } 961 } 962 if (removed) { 963 purgeStaleEffects_l(); 964 } 965} 966 967// audioConfigChanged_l() must be called with AudioFlinger::mLock held 968void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 969{ 970 size_t size = mNotificationClients.size(); 971 for (size_t i = 0; i < size; i++) { 972 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 973 } 974} 975 976// removeClient_l() must be called with AudioFlinger::mLock held 977void AudioFlinger::removeClient_l(pid_t pid) 978{ 979 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 980 mClients.removeItem(pid); 981} 982 983 984// ---------------------------------------------------------------------------- 985 986AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 987 : Thread(false), 988 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 989 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), 990 mDevice(device) 991{ 992 mDeathRecipient = new PMDeathRecipient(this); 993} 994 995AudioFlinger::ThreadBase::~ThreadBase() 996{ 997 mParamCond.broadcast(); 998 // do not lock the mutex in destructor 999 releaseWakeLock_l(); 1000 if (mPowerManager != 0) { 1001 sp<IBinder> binder = mPowerManager->asBinder(); 1002 binder->unlinkToDeath(mDeathRecipient); 1003 } 1004} 1005 1006void AudioFlinger::ThreadBase::exit() 1007{ 1008 // keep a strong ref on ourself so that we won't get 1009 // destroyed in the middle of requestExitAndWait() 1010 sp <ThreadBase> strongMe = this; 1011 1012 ALOGV("ThreadBase::exit"); 1013 { 1014 AutoMutex lock(mLock); 1015 mExiting = true; 1016 requestExit(); 1017 mWaitWorkCV.signal(); 1018 } 1019 requestExitAndWait(); 1020} 1021 1022uint32_t AudioFlinger::ThreadBase::sampleRate() const 1023{ 1024 return mSampleRate; 1025} 1026 1027int AudioFlinger::ThreadBase::channelCount() const 1028{ 1029 return (int)mChannelCount; 1030} 1031 1032uint32_t AudioFlinger::ThreadBase::format() const 1033{ 1034 return mFormat; 1035} 1036 1037size_t AudioFlinger::ThreadBase::frameCount() const 1038{ 1039 return mFrameCount; 1040} 1041 1042status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1043{ 1044 status_t status; 1045 1046 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1047 Mutex::Autolock _l(mLock); 1048 1049 mNewParameters.add(keyValuePairs); 1050 mWaitWorkCV.signal(); 1051 // wait condition with timeout in case the thread loop has exited 1052 // before the request could be processed 1053 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1054 status = mParamStatus; 1055 mWaitWorkCV.signal(); 1056 } else { 1057 status = TIMED_OUT; 1058 } 1059 return status; 1060} 1061 1062void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1063{ 1064 Mutex::Autolock _l(mLock); 1065 sendConfigEvent_l(event, param); 1066} 1067 1068// sendConfigEvent_l() must be called with ThreadBase::mLock held 1069void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1070{ 1071 ConfigEvent configEvent; 1072 configEvent.mEvent = event; 1073 configEvent.mParam = param; 1074 mConfigEvents.add(configEvent); 1075 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1076 mWaitWorkCV.signal(); 1077} 1078 1079void AudioFlinger::ThreadBase::processConfigEvents() 1080{ 1081 mLock.lock(); 1082 while(!mConfigEvents.isEmpty()) { 1083 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1084 ConfigEvent configEvent = mConfigEvents[0]; 1085 mConfigEvents.removeAt(0); 1086 // release mLock before locking AudioFlinger mLock: lock order is always 1087 // AudioFlinger then ThreadBase to avoid cross deadlock 1088 mLock.unlock(); 1089 mAudioFlinger->mLock.lock(); 1090 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1091 mAudioFlinger->mLock.unlock(); 1092 mLock.lock(); 1093 } 1094 mLock.unlock(); 1095} 1096 1097status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1098{ 1099 const size_t SIZE = 256; 1100 char buffer[SIZE]; 1101 String8 result; 1102 1103 bool locked = tryLock(mLock); 1104 if (!locked) { 1105 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1106 write(fd, buffer, strlen(buffer)); 1107 } 1108 1109 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1110 result.append(buffer); 1111 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1112 result.append(buffer); 1113 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1114 result.append(buffer); 1115 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1116 result.append(buffer); 1117 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 1122 result.append(buffer); 1123 1124 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1125 result.append(buffer); 1126 result.append(" Index Command"); 1127 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1128 snprintf(buffer, SIZE, "\n %02d ", i); 1129 result.append(buffer); 1130 result.append(mNewParameters[i]); 1131 } 1132 1133 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1134 result.append(buffer); 1135 snprintf(buffer, SIZE, " Index event param\n"); 1136 result.append(buffer); 1137 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1138 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1139 result.append(buffer); 1140 } 1141 result.append("\n"); 1142 1143 write(fd, result.string(), result.size()); 1144 1145 if (locked) { 1146 mLock.unlock(); 1147 } 1148 return NO_ERROR; 1149} 1150 1151status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1152{ 1153 const size_t SIZE = 256; 1154 char buffer[SIZE]; 1155 String8 result; 1156 1157 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1158 write(fd, buffer, strlen(buffer)); 1159 1160 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1161 sp<EffectChain> chain = mEffectChains[i]; 1162 if (chain != 0) { 1163 chain->dump(fd, args); 1164 } 1165 } 1166 return NO_ERROR; 1167} 1168 1169void AudioFlinger::ThreadBase::acquireWakeLock() 1170{ 1171 Mutex::Autolock _l(mLock); 1172 acquireWakeLock_l(); 1173} 1174 1175void AudioFlinger::ThreadBase::acquireWakeLock_l() 1176{ 1177 if (mPowerManager == 0) { 1178 // use checkService() to avoid blocking if power service is not up yet 1179 sp<IBinder> binder = 1180 defaultServiceManager()->checkService(String16("power")); 1181 if (binder == 0) { 1182 ALOGW("Thread %s cannot connect to the power manager service", mName); 1183 } else { 1184 mPowerManager = interface_cast<IPowerManager>(binder); 1185 binder->linkToDeath(mDeathRecipient); 1186 } 1187 } 1188 if (mPowerManager != 0) { 1189 sp<IBinder> binder = new BBinder(); 1190 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1191 binder, 1192 String16(mName)); 1193 if (status == NO_ERROR) { 1194 mWakeLockToken = binder; 1195 } 1196 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1197 } 1198} 1199 1200void AudioFlinger::ThreadBase::releaseWakeLock() 1201{ 1202 Mutex::Autolock _l(mLock); 1203 releaseWakeLock_l(); 1204} 1205 1206void AudioFlinger::ThreadBase::releaseWakeLock_l() 1207{ 1208 if (mWakeLockToken != 0) { 1209 ALOGV("releaseWakeLock_l() %s", mName); 1210 if (mPowerManager != 0) { 1211 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1212 } 1213 mWakeLockToken.clear(); 1214 } 1215} 1216 1217void AudioFlinger::ThreadBase::clearPowerManager() 1218{ 1219 Mutex::Autolock _l(mLock); 1220 releaseWakeLock_l(); 1221 mPowerManager.clear(); 1222} 1223 1224void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1225{ 1226 sp<ThreadBase> thread = mThread.promote(); 1227 if (thread != 0) { 1228 thread->clearPowerManager(); 1229 } 1230 ALOGW("power manager service died !!!"); 1231} 1232 1233void AudioFlinger::ThreadBase::setEffectSuspended( 1234 const effect_uuid_t *type, bool suspend, int sessionId) 1235{ 1236 Mutex::Autolock _l(mLock); 1237 setEffectSuspended_l(type, suspend, sessionId); 1238} 1239 1240void AudioFlinger::ThreadBase::setEffectSuspended_l( 1241 const effect_uuid_t *type, bool suspend, int sessionId) 1242{ 1243 sp<EffectChain> chain; 1244 chain = getEffectChain_l(sessionId); 1245 if (chain != 0) { 1246 if (type != NULL) { 1247 chain->setEffectSuspended_l(type, suspend); 1248 } else { 1249 chain->setEffectSuspendedAll_l(suspend); 1250 } 1251 } 1252 1253 updateSuspendedSessions_l(type, suspend, sessionId); 1254} 1255 1256void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1257{ 1258 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1259 if (index < 0) { 1260 return; 1261 } 1262 1263 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1264 mSuspendedSessions.editValueAt(index); 1265 1266 for (size_t i = 0; i < sessionEffects.size(); i++) { 1267 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1268 for (int j = 0; j < desc->mRefCount; j++) { 1269 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1270 chain->setEffectSuspendedAll_l(true); 1271 } else { 1272 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1273 desc->mType.timeLow); 1274 chain->setEffectSuspended_l(&desc->mType, true); 1275 } 1276 } 1277 } 1278} 1279 1280void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1281 bool suspend, 1282 int sessionId) 1283{ 1284 int index = mSuspendedSessions.indexOfKey(sessionId); 1285 1286 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1287 1288 if (suspend) { 1289 if (index >= 0) { 1290 sessionEffects = mSuspendedSessions.editValueAt(index); 1291 } else { 1292 mSuspendedSessions.add(sessionId, sessionEffects); 1293 } 1294 } else { 1295 if (index < 0) { 1296 return; 1297 } 1298 sessionEffects = mSuspendedSessions.editValueAt(index); 1299 } 1300 1301 1302 int key = EffectChain::kKeyForSuspendAll; 1303 if (type != NULL) { 1304 key = type->timeLow; 1305 } 1306 index = sessionEffects.indexOfKey(key); 1307 1308 sp <SuspendedSessionDesc> desc; 1309 if (suspend) { 1310 if (index >= 0) { 1311 desc = sessionEffects.valueAt(index); 1312 } else { 1313 desc = new SuspendedSessionDesc(); 1314 if (type != NULL) { 1315 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1316 } 1317 sessionEffects.add(key, desc); 1318 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1319 } 1320 desc->mRefCount++; 1321 } else { 1322 if (index < 0) { 1323 return; 1324 } 1325 desc = sessionEffects.valueAt(index); 1326 if (--desc->mRefCount == 0) { 1327 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1328 sessionEffects.removeItemsAt(index); 1329 if (sessionEffects.isEmpty()) { 1330 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1331 sessionId); 1332 mSuspendedSessions.removeItem(sessionId); 1333 } 1334 } 1335 } 1336 if (!sessionEffects.isEmpty()) { 1337 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1338 } 1339} 1340 1341void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1342 bool enabled, 1343 int sessionId) 1344{ 1345 Mutex::Autolock _l(mLock); 1346 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1347} 1348 1349void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1350 bool enabled, 1351 int sessionId) 1352{ 1353 if (mType != RECORD) { 1354 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1355 // another session. This gives the priority to well behaved effect control panels 1356 // and applications not using global effects. 1357 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1358 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1359 } 1360 } 1361 1362 sp<EffectChain> chain = getEffectChain_l(sessionId); 1363 if (chain != 0) { 1364 chain->checkSuspendOnEffectEnabled(effect, enabled); 1365 } 1366} 1367 1368// ---------------------------------------------------------------------------- 1369 1370AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1371 AudioStreamOut* output, 1372 int id, 1373 uint32_t device) 1374 : ThreadBase(audioFlinger, id, device), 1375 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1376 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1377{ 1378 snprintf(mName, kNameLength, "AudioOut_%d", id); 1379 1380 readOutputParameters(); 1381 1382 mMasterVolume = mAudioFlinger->masterVolume(); 1383 mMasterMute = mAudioFlinger->masterMute(); 1384 1385 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 1386 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1387 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1388 mStreamTypes[stream].valid = true; 1389 } 1390} 1391 1392AudioFlinger::PlaybackThread::~PlaybackThread() 1393{ 1394 delete [] mMixBuffer; 1395} 1396 1397status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1398{ 1399 dumpInternals(fd, args); 1400 dumpTracks(fd, args); 1401 dumpEffectChains(fd, args); 1402 return NO_ERROR; 1403} 1404 1405status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1406{ 1407 const size_t SIZE = 256; 1408 char buffer[SIZE]; 1409 String8 result; 1410 1411 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1412 result.append(buffer); 1413 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1414 for (size_t i = 0; i < mTracks.size(); ++i) { 1415 sp<Track> track = mTracks[i]; 1416 if (track != 0) { 1417 track->dump(buffer, SIZE); 1418 result.append(buffer); 1419 } 1420 } 1421 1422 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1423 result.append(buffer); 1424 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1425 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1426 wp<Track> wTrack = mActiveTracks[i]; 1427 if (wTrack != 0) { 1428 sp<Track> track = wTrack.promote(); 1429 if (track != 0) { 1430 track->dump(buffer, SIZE); 1431 result.append(buffer); 1432 } 1433 } 1434 } 1435 write(fd, result.string(), result.size()); 1436 return NO_ERROR; 1437} 1438 1439status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1440{ 1441 const size_t SIZE = 256; 1442 char buffer[SIZE]; 1443 String8 result; 1444 1445 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1446 result.append(buffer); 1447 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1448 result.append(buffer); 1449 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1450 result.append(buffer); 1451 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1452 result.append(buffer); 1453 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1454 result.append(buffer); 1455 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1456 result.append(buffer); 1457 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1458 result.append(buffer); 1459 write(fd, result.string(), result.size()); 1460 1461 dumpBase(fd, args); 1462 1463 return NO_ERROR; 1464} 1465 1466// Thread virtuals 1467status_t AudioFlinger::PlaybackThread::readyToRun() 1468{ 1469 status_t status = initCheck(); 1470 if (status == NO_ERROR) { 1471 ALOGI("AudioFlinger's thread %p ready to run", this); 1472 } else { 1473 ALOGE("No working audio driver found."); 1474 } 1475 return status; 1476} 1477 1478void AudioFlinger::PlaybackThread::onFirstRef() 1479{ 1480 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1481} 1482 1483// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1484sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1485 const sp<AudioFlinger::Client>& client, 1486 int streamType, 1487 uint32_t sampleRate, 1488 uint32_t format, 1489 uint32_t channelMask, 1490 int frameCount, 1491 const sp<IMemory>& sharedBuffer, 1492 int sessionId, 1493 status_t *status) 1494{ 1495 sp<Track> track; 1496 status_t lStatus; 1497 1498 if (mType == DIRECT) { 1499 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1500 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1501 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1502 "for output %p with format %d", 1503 sampleRate, format, channelMask, mOutput, mFormat); 1504 lStatus = BAD_VALUE; 1505 goto Exit; 1506 } 1507 } 1508 } else { 1509 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1510 if (sampleRate > mSampleRate*2) { 1511 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1512 lStatus = BAD_VALUE; 1513 goto Exit; 1514 } 1515 } 1516 1517 lStatus = initCheck(); 1518 if (lStatus != NO_ERROR) { 1519 ALOGE("Audio driver not initialized."); 1520 goto Exit; 1521 } 1522 1523 { // scope for mLock 1524 Mutex::Autolock _l(mLock); 1525 1526 // all tracks in same audio session must share the same routing strategy otherwise 1527 // conflicts will happen when tracks are moved from one output to another by audio policy 1528 // manager 1529 uint32_t strategy = 1530 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1531 for (size_t i = 0; i < mTracks.size(); ++i) { 1532 sp<Track> t = mTracks[i]; 1533 if (t != 0) { 1534 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1535 if (sessionId == t->sessionId() && strategy != actual) { 1536 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1537 strategy, actual); 1538 lStatus = BAD_VALUE; 1539 goto Exit; 1540 } 1541 } 1542 } 1543 1544 track = new Track(this, client, streamType, sampleRate, format, 1545 channelMask, frameCount, sharedBuffer, sessionId); 1546 if (track->getCblk() == NULL || track->name() < 0) { 1547 lStatus = NO_MEMORY; 1548 goto Exit; 1549 } 1550 mTracks.add(track); 1551 1552 sp<EffectChain> chain = getEffectChain_l(sessionId); 1553 if (chain != 0) { 1554 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1555 track->setMainBuffer(chain->inBuffer()); 1556 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1557 chain->incTrackCnt(); 1558 } 1559 1560 // invalidate track immediately if the stream type was moved to another thread since 1561 // createTrack() was called by the client process. 1562 if (!mStreamTypes[streamType].valid) { 1563 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1564 this, streamType); 1565 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1566 } 1567 } 1568 lStatus = NO_ERROR; 1569 1570Exit: 1571 if(status) { 1572 *status = lStatus; 1573 } 1574 return track; 1575} 1576 1577uint32_t AudioFlinger::PlaybackThread::latency() const 1578{ 1579 Mutex::Autolock _l(mLock); 1580 if (initCheck() == NO_ERROR) { 1581 return mOutput->stream->get_latency(mOutput->stream); 1582 } else { 1583 return 0; 1584 } 1585} 1586 1587status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1588{ 1589 mMasterVolume = value; 1590 return NO_ERROR; 1591} 1592 1593status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1594{ 1595 mMasterMute = muted; 1596 return NO_ERROR; 1597} 1598 1599float AudioFlinger::PlaybackThread::masterVolume() const 1600{ 1601 return mMasterVolume; 1602} 1603 1604bool AudioFlinger::PlaybackThread::masterMute() const 1605{ 1606 return mMasterMute; 1607} 1608 1609status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1610{ 1611 mStreamTypes[stream].volume = value; 1612 return NO_ERROR; 1613} 1614 1615status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1616{ 1617 mStreamTypes[stream].mute = muted; 1618 return NO_ERROR; 1619} 1620 1621float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1622{ 1623 return mStreamTypes[stream].volume; 1624} 1625 1626bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1627{ 1628 return mStreamTypes[stream].mute; 1629} 1630 1631// addTrack_l() must be called with ThreadBase::mLock held 1632status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1633{ 1634 status_t status = ALREADY_EXISTS; 1635 1636 // set retry count for buffer fill 1637 track->mRetryCount = kMaxTrackStartupRetries; 1638 if (mActiveTracks.indexOf(track) < 0) { 1639 // the track is newly added, make sure it fills up all its 1640 // buffers before playing. This is to ensure the client will 1641 // effectively get the latency it requested. 1642 track->mFillingUpStatus = Track::FS_FILLING; 1643 track->mResetDone = false; 1644 mActiveTracks.add(track); 1645 if (track->mainBuffer() != mMixBuffer) { 1646 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1647 if (chain != 0) { 1648 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1649 chain->incActiveTrackCnt(); 1650 } 1651 } 1652 1653 status = NO_ERROR; 1654 } 1655 1656 ALOGV("mWaitWorkCV.broadcast"); 1657 mWaitWorkCV.broadcast(); 1658 1659 return status; 1660} 1661 1662// destroyTrack_l() must be called with ThreadBase::mLock held 1663void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1664{ 1665 track->mState = TrackBase::TERMINATED; 1666 if (mActiveTracks.indexOf(track) < 0) { 1667 removeTrack_l(track); 1668 } 1669} 1670 1671void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1672{ 1673 mTracks.remove(track); 1674 deleteTrackName_l(track->name()); 1675 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1676 if (chain != 0) { 1677 chain->decTrackCnt(); 1678 } 1679} 1680 1681String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1682{ 1683 String8 out_s8 = String8(""); 1684 char *s; 1685 1686 Mutex::Autolock _l(mLock); 1687 if (initCheck() != NO_ERROR) { 1688 return out_s8; 1689 } 1690 1691 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1692 out_s8 = String8(s); 1693 free(s); 1694 return out_s8; 1695} 1696 1697// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1698void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1699 AudioSystem::OutputDescriptor desc; 1700 void *param2 = 0; 1701 1702 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1703 1704 switch (event) { 1705 case AudioSystem::OUTPUT_OPENED: 1706 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1707 desc.channels = mChannelMask; 1708 desc.samplingRate = mSampleRate; 1709 desc.format = mFormat; 1710 desc.frameCount = mFrameCount; 1711 desc.latency = latency(); 1712 param2 = &desc; 1713 break; 1714 1715 case AudioSystem::STREAM_CONFIG_CHANGED: 1716 param2 = ¶m; 1717 case AudioSystem::OUTPUT_CLOSED: 1718 default: 1719 break; 1720 } 1721 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1722} 1723 1724void AudioFlinger::PlaybackThread::readOutputParameters() 1725{ 1726 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1727 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1728 mChannelCount = (uint16_t)popcount(mChannelMask); 1729 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1730 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); 1731 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1732 1733 // FIXME - Current mixer implementation only supports stereo output: Always 1734 // Allocate a stereo buffer even if HW output is mono. 1735 if (mMixBuffer != NULL) delete[] mMixBuffer; 1736 mMixBuffer = new int16_t[mFrameCount * 2]; 1737 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1738 1739 // force reconfiguration of effect chains and engines to take new buffer size and audio 1740 // parameters into account 1741 // Note that mLock is not held when readOutputParameters() is called from the constructor 1742 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1743 // matter. 1744 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1745 Vector< sp<EffectChain> > effectChains = mEffectChains; 1746 for (size_t i = 0; i < effectChains.size(); i ++) { 1747 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1748 } 1749} 1750 1751status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1752{ 1753 if (halFrames == 0 || dspFrames == 0) { 1754 return BAD_VALUE; 1755 } 1756 Mutex::Autolock _l(mLock); 1757 if (initCheck() != NO_ERROR) { 1758 return INVALID_OPERATION; 1759 } 1760 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1761 1762 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1763} 1764 1765uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1766{ 1767 Mutex::Autolock _l(mLock); 1768 uint32_t result = 0; 1769 if (getEffectChain_l(sessionId) != 0) { 1770 result = EFFECT_SESSION; 1771 } 1772 1773 for (size_t i = 0; i < mTracks.size(); ++i) { 1774 sp<Track> track = mTracks[i]; 1775 if (sessionId == track->sessionId() && 1776 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1777 result |= TRACK_SESSION; 1778 break; 1779 } 1780 } 1781 1782 return result; 1783} 1784 1785uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1786{ 1787 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1788 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1789 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1790 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1791 } 1792 for (size_t i = 0; i < mTracks.size(); i++) { 1793 sp<Track> track = mTracks[i]; 1794 if (sessionId == track->sessionId() && 1795 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1796 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1797 } 1798 } 1799 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1800} 1801 1802 1803AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() 1804{ 1805 Mutex::Autolock _l(mLock); 1806 return mOutput; 1807} 1808 1809AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1810{ 1811 Mutex::Autolock _l(mLock); 1812 AudioStreamOut *output = mOutput; 1813 mOutput = NULL; 1814 return output; 1815} 1816 1817// this method must always be called either with ThreadBase mLock held or inside the thread loop 1818audio_stream_t* AudioFlinger::PlaybackThread::stream() 1819{ 1820 if (mOutput == NULL) { 1821 return NULL; 1822 } 1823 return &mOutput->stream->common; 1824} 1825 1826uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1827{ 1828 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1829 // decoding and transfer time. So sleeping for half of the latency would likely cause 1830 // underruns 1831 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1832 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1833 } else { 1834 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1835 } 1836} 1837 1838// ---------------------------------------------------------------------------- 1839 1840AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1841 : PlaybackThread(audioFlinger, output, id, device), 1842 mAudioMixer(NULL) 1843{ 1844 mType = ThreadBase::MIXER; 1845 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1846 1847 // FIXME - Current mixer implementation only supports stereo output 1848 if (mChannelCount == 1) { 1849 ALOGE("Invalid audio hardware channel count"); 1850 } 1851} 1852 1853AudioFlinger::MixerThread::~MixerThread() 1854{ 1855 delete mAudioMixer; 1856} 1857 1858bool AudioFlinger::MixerThread::threadLoop() 1859{ 1860 Vector< sp<Track> > tracksToRemove; 1861 uint32_t mixerStatus = MIXER_IDLE; 1862 nsecs_t standbyTime = systemTime(); 1863 size_t mixBufferSize = mFrameCount * mFrameSize; 1864 // FIXME: Relaxed timing because of a certain device that can't meet latency 1865 // Should be reduced to 2x after the vendor fixes the driver issue 1866 // increase threshold again due to low power audio mode. The way this warning threshold is 1867 // calculated and its usefulness should be reconsidered anyway. 1868 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1869 nsecs_t lastWarning = 0; 1870 bool longStandbyExit = false; 1871 uint32_t activeSleepTime = activeSleepTimeUs(); 1872 uint32_t idleSleepTime = idleSleepTimeUs(); 1873 uint32_t sleepTime = idleSleepTime; 1874 uint32_t sleepTimeShift = 0; 1875 Vector< sp<EffectChain> > effectChains; 1876#ifdef DEBUG_CPU_USAGE 1877 ThreadCpuUsage cpu; 1878 const CentralTendencyStatistics& stats = cpu.statistics(); 1879#endif 1880 1881 acquireWakeLock(); 1882 1883 while (!exitPending()) 1884 { 1885#ifdef DEBUG_CPU_USAGE 1886 cpu.sampleAndEnable(); 1887 unsigned n = stats.n(); 1888 // cpu.elapsed() is expensive, so don't call it every loop 1889 if ((n & 127) == 1) { 1890 long long elapsed = cpu.elapsed(); 1891 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1892 double perLoop = elapsed / (double) n; 1893 double perLoop100 = perLoop * 0.01; 1894 double mean = stats.mean(); 1895 double stddev = stats.stddev(); 1896 double minimum = stats.minimum(); 1897 double maximum = stats.maximum(); 1898 cpu.resetStatistics(); 1899 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", 1900 elapsed * .000000001, n, perLoop * .000001, 1901 mean * .001, 1902 stddev * .001, 1903 minimum * .001, 1904 maximum * .001, 1905 mean / perLoop100, 1906 stddev / perLoop100, 1907 minimum / perLoop100, 1908 maximum / perLoop100); 1909 } 1910 } 1911#endif 1912 processConfigEvents(); 1913 1914 mixerStatus = MIXER_IDLE; 1915 { // scope for mLock 1916 1917 Mutex::Autolock _l(mLock); 1918 1919 if (checkForNewParameters_l()) { 1920 mixBufferSize = mFrameCount * mFrameSize; 1921 // FIXME: Relaxed timing because of a certain device that can't meet latency 1922 // Should be reduced to 2x after the vendor fixes the driver issue 1923 // increase threshold again due to low power audio mode. The way this warning 1924 // threshold is calculated and its usefulness should be reconsidered anyway. 1925 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1926 activeSleepTime = activeSleepTimeUs(); 1927 idleSleepTime = idleSleepTimeUs(); 1928 } 1929 1930 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1931 1932 // put audio hardware into standby after short delay 1933 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1934 mSuspended)) { 1935 if (!mStandby) { 1936 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1937 mOutput->stream->common.standby(&mOutput->stream->common); 1938 mStandby = true; 1939 mBytesWritten = 0; 1940 } 1941 1942 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1943 // we're about to wait, flush the binder command buffer 1944 IPCThreadState::self()->flushCommands(); 1945 1946 if (exitPending()) break; 1947 1948 releaseWakeLock_l(); 1949 // wait until we have something to do... 1950 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1951 mWaitWorkCV.wait(mLock); 1952 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1953 acquireWakeLock_l(); 1954 1955 if (mMasterMute == false) { 1956 char value[PROPERTY_VALUE_MAX]; 1957 property_get("ro.audio.silent", value, "0"); 1958 if (atoi(value)) { 1959 ALOGD("Silence is golden"); 1960 setMasterMute(true); 1961 } 1962 } 1963 1964 standbyTime = systemTime() + kStandbyTimeInNsecs; 1965 sleepTime = idleSleepTime; 1966 sleepTimeShift = 0; 1967 continue; 1968 } 1969 } 1970 1971 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1972 1973 // prevent any changes in effect chain list and in each effect chain 1974 // during mixing and effect process as the audio buffers could be deleted 1975 // or modified if an effect is created or deleted 1976 lockEffectChains_l(effectChains); 1977 } 1978 1979 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1980 // mix buffers... 1981 mAudioMixer->process(); 1982 sleepTime = 0; 1983 // increase sleep time progressively when application underrun condition clears 1984 if (sleepTimeShift > 0) { 1985 sleepTimeShift--; 1986 } 1987 standbyTime = systemTime() + kStandbyTimeInNsecs; 1988 //TODO: delay standby when effects have a tail 1989 } else { 1990 // If no tracks are ready, sleep once for the duration of an output 1991 // buffer size, then write 0s to the output 1992 if (sleepTime == 0) { 1993 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1994 sleepTime = activeSleepTime >> sleepTimeShift; 1995 if (sleepTime < kMinThreadSleepTimeUs) { 1996 sleepTime = kMinThreadSleepTimeUs; 1997 } 1998 // reduce sleep time in case of consecutive application underruns to avoid 1999 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2000 // duration we would end up writing less data than needed by the audio HAL if 2001 // the condition persists. 2002 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2003 sleepTimeShift++; 2004 } 2005 } else { 2006 sleepTime = idleSleepTime; 2007 } 2008 } else if (mBytesWritten != 0 || 2009 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2010 memset (mMixBuffer, 0, mixBufferSize); 2011 sleepTime = 0; 2012 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2013 } 2014 // TODO add standby time extension fct of effect tail 2015 } 2016 2017 if (mSuspended) { 2018 sleepTime = suspendSleepTimeUs(); 2019 } 2020 // sleepTime == 0 means we must write to audio hardware 2021 if (sleepTime == 0) { 2022 for (size_t i = 0; i < effectChains.size(); i ++) { 2023 effectChains[i]->process_l(); 2024 } 2025 // enable changes in effect chain 2026 unlockEffectChains(effectChains); 2027 mLastWriteTime = systemTime(); 2028 mInWrite = true; 2029 mBytesWritten += mixBufferSize; 2030 2031 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2032 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2033 mNumWrites++; 2034 mInWrite = false; 2035 nsecs_t now = systemTime(); 2036 nsecs_t delta = now - mLastWriteTime; 2037 if (!mStandby && delta > maxPeriod) { 2038 mNumDelayedWrites++; 2039 if ((now - lastWarning) > kWarningThrottleNs) { 2040 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2041 ns2ms(delta), mNumDelayedWrites, this); 2042 lastWarning = now; 2043 } 2044 if (mStandby) { 2045 longStandbyExit = true; 2046 } 2047 } 2048 mStandby = false; 2049 } else { 2050 // enable changes in effect chain 2051 unlockEffectChains(effectChains); 2052 usleep(sleepTime); 2053 } 2054 2055 // finally let go of all our tracks, without the lock held 2056 // since we can't guarantee the destructors won't acquire that 2057 // same lock. 2058 tracksToRemove.clear(); 2059 2060 // Effect chains will be actually deleted here if they were removed from 2061 // mEffectChains list during mixing or effects processing 2062 effectChains.clear(); 2063 } 2064 2065 if (!mStandby) { 2066 mOutput->stream->common.standby(&mOutput->stream->common); 2067 } 2068 2069 releaseWakeLock(); 2070 2071 ALOGV("MixerThread %p exiting", this); 2072 return false; 2073} 2074 2075// prepareTracks_l() must be called with ThreadBase::mLock held 2076uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2077{ 2078 2079 uint32_t mixerStatus = MIXER_IDLE; 2080 // find out which tracks need to be processed 2081 size_t count = activeTracks.size(); 2082 size_t mixedTracks = 0; 2083 size_t tracksWithEffect = 0; 2084 2085 float masterVolume = mMasterVolume; 2086 bool masterMute = mMasterMute; 2087 2088 if (masterMute) { 2089 masterVolume = 0; 2090 } 2091 // Delegate master volume control to effect in output mix effect chain if needed 2092 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2093 if (chain != 0) { 2094 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2095 chain->setVolume_l(&v, &v); 2096 masterVolume = (float)((v + (1 << 23)) >> 24); 2097 chain.clear(); 2098 } 2099 2100 for (size_t i=0 ; i<count ; i++) { 2101 sp<Track> t = activeTracks[i].promote(); 2102 if (t == 0) continue; 2103 2104 // this const just means the local variable doesn't change 2105 Track* const track = t.get(); 2106 audio_track_cblk_t* cblk = track->cblk(); 2107 2108 // The first time a track is added we wait 2109 // for all its buffers to be filled before processing it 2110 int name = track->name(); 2111 // make sure that we have enough frames to mix one full buffer. 2112 // enforce this condition only once to enable draining the buffer in case the client 2113 // app does not call stop() and relies on underrun to stop: 2114 // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed 2115 // during last round 2116 uint32_t minFrames = 1; 2117 if (!track->isStopped() && !track->isPausing() && 2118 (track->mRetryCount >= kMaxTrackRetries)) { 2119 if (t->sampleRate() == (int)mSampleRate) { 2120 minFrames = mFrameCount; 2121 } else { 2122 // +1 for rounding and +1 for additional sample needed for interpolation 2123 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2124 // add frames already consumed but not yet released by the resampler 2125 // because cblk->framesReady() will include these frames 2126 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2127 // the minimum track buffer size is normally twice the number of frames necessary 2128 // to fill one buffer and the resampler should not leave more than one buffer worth 2129 // of unreleased frames after each pass, but just in case... 2130 ALOG_ASSERT(minFrames <= cblk->frameCount); 2131 } 2132 } 2133 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2134 !track->isPaused() && !track->isTerminated()) 2135 { 2136 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2137 2138 mixedTracks++; 2139 2140 // track->mainBuffer() != mMixBuffer means there is an effect chain 2141 // connected to the track 2142 chain.clear(); 2143 if (track->mainBuffer() != mMixBuffer) { 2144 chain = getEffectChain_l(track->sessionId()); 2145 // Delegate volume control to effect in track effect chain if needed 2146 if (chain != 0) { 2147 tracksWithEffect++; 2148 } else { 2149 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2150 name, track->sessionId()); 2151 } 2152 } 2153 2154 2155 int param = AudioMixer::VOLUME; 2156 if (track->mFillingUpStatus == Track::FS_FILLED) { 2157 // no ramp for the first volume setting 2158 track->mFillingUpStatus = Track::FS_ACTIVE; 2159 if (track->mState == TrackBase::RESUMING) { 2160 track->mState = TrackBase::ACTIVE; 2161 param = AudioMixer::RAMP_VOLUME; 2162 } 2163 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2164 } else if (cblk->server != 0) { 2165 // If the track is stopped before the first frame was mixed, 2166 // do not apply ramp 2167 param = AudioMixer::RAMP_VOLUME; 2168 } 2169 2170 // compute volume for this track 2171 uint32_t vl, vr, va; 2172 if (track->isMuted() || track->isPausing() || 2173 mStreamTypes[track->type()].mute) { 2174 vl = vr = va = 0; 2175 if (track->isPausing()) { 2176 track->setPaused(); 2177 } 2178 } else { 2179 2180 // read original volumes with volume control 2181 float typeVolume = mStreamTypes[track->type()].volume; 2182 float v = masterVolume * typeVolume; 2183 vl = (uint32_t)(v * cblk->volume[0]) << 12; 2184 vr = (uint32_t)(v * cblk->volume[1]) << 12; 2185 2186 va = (uint32_t)(v * cblk->sendLevel); 2187 } 2188 // Delegate volume control to effect in track effect chain if needed 2189 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2190 // Do not ramp volume if volume is controlled by effect 2191 param = AudioMixer::VOLUME; 2192 track->mHasVolumeController = true; 2193 } else { 2194 // force no volume ramp when volume controller was just disabled or removed 2195 // from effect chain to avoid volume spike 2196 if (track->mHasVolumeController) { 2197 param = AudioMixer::VOLUME; 2198 } 2199 track->mHasVolumeController = false; 2200 } 2201 2202 // Convert volumes from 8.24 to 4.12 format 2203 int16_t left, right, aux; 2204 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2205 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2206 left = int16_t(v_clamped); 2207 v_clamped = (vr + (1 << 11)) >> 12; 2208 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2209 right = int16_t(v_clamped); 2210 2211 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2212 aux = int16_t(va); 2213 2214 // XXX: these things DON'T need to be done each time 2215 mAudioMixer->setBufferProvider(name, track); 2216 mAudioMixer->enable(name); 2217 2218 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2219 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2220 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2221 mAudioMixer->setParameter( 2222 name, 2223 AudioMixer::TRACK, 2224 AudioMixer::FORMAT, (void *)track->format()); 2225 mAudioMixer->setParameter( 2226 name, 2227 AudioMixer::TRACK, 2228 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2229 mAudioMixer->setParameter( 2230 name, 2231 AudioMixer::RESAMPLE, 2232 AudioMixer::SAMPLE_RATE, 2233 (void *)(cblk->sampleRate)); 2234 mAudioMixer->setParameter( 2235 name, 2236 AudioMixer::TRACK, 2237 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2238 mAudioMixer->setParameter( 2239 name, 2240 AudioMixer::TRACK, 2241 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2242 2243 // reset retry count 2244 track->mRetryCount = kMaxTrackRetries; 2245 mixerStatus = MIXER_TRACKS_READY; 2246 } else { 2247 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2248 if (track->isStopped()) { 2249 track->reset(); 2250 } 2251 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2252 // We have consumed all the buffers of this track. 2253 // Remove it from the list of active tracks. 2254 tracksToRemove->add(track); 2255 } else { 2256 // No buffers for this track. Give it a few chances to 2257 // fill a buffer, then remove it from active list. 2258 if (--(track->mRetryCount) <= 0) { 2259 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2260 tracksToRemove->add(track); 2261 // indicate to client process that the track was disabled because of underrun 2262 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2263 } else if (mixerStatus != MIXER_TRACKS_READY) { 2264 mixerStatus = MIXER_TRACKS_ENABLED; 2265 } 2266 } 2267 mAudioMixer->disable(name); 2268 } 2269 } 2270 2271 // remove all the tracks that need to be... 2272 count = tracksToRemove->size(); 2273 if (CC_UNLIKELY(count)) { 2274 for (size_t i=0 ; i<count ; i++) { 2275 const sp<Track>& track = tracksToRemove->itemAt(i); 2276 mActiveTracks.remove(track); 2277 if (track->mainBuffer() != mMixBuffer) { 2278 chain = getEffectChain_l(track->sessionId()); 2279 if (chain != 0) { 2280 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2281 chain->decActiveTrackCnt(); 2282 } 2283 } 2284 if (track->isTerminated()) { 2285 removeTrack_l(track); 2286 } 2287 } 2288 } 2289 2290 // mix buffer must be cleared if all tracks are connected to an 2291 // effect chain as in this case the mixer will not write to 2292 // mix buffer and track effects will accumulate into it 2293 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2294 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2295 } 2296 2297 return mixerStatus; 2298} 2299 2300void AudioFlinger::MixerThread::invalidateTracks(int streamType) 2301{ 2302 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2303 this, streamType, mTracks.size()); 2304 Mutex::Autolock _l(mLock); 2305 2306 size_t size = mTracks.size(); 2307 for (size_t i = 0; i < size; i++) { 2308 sp<Track> t = mTracks[i]; 2309 if (t->type() == streamType) { 2310 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2311 t->mCblk->cv.signal(); 2312 } 2313 } 2314} 2315 2316void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid) 2317{ 2318 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2319 this, streamType, valid); 2320 Mutex::Autolock _l(mLock); 2321 2322 mStreamTypes[streamType].valid = valid; 2323} 2324 2325// getTrackName_l() must be called with ThreadBase::mLock held 2326int AudioFlinger::MixerThread::getTrackName_l() 2327{ 2328 return mAudioMixer->getTrackName(); 2329} 2330 2331// deleteTrackName_l() must be called with ThreadBase::mLock held 2332void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2333{ 2334 ALOGV("remove track (%d) and delete from mixer", name); 2335 mAudioMixer->deleteTrackName(name); 2336} 2337 2338// checkForNewParameters_l() must be called with ThreadBase::mLock held 2339bool AudioFlinger::MixerThread::checkForNewParameters_l() 2340{ 2341 bool reconfig = false; 2342 2343 while (!mNewParameters.isEmpty()) { 2344 status_t status = NO_ERROR; 2345 String8 keyValuePair = mNewParameters[0]; 2346 AudioParameter param = AudioParameter(keyValuePair); 2347 int value; 2348 2349 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2350 reconfig = true; 2351 } 2352 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2353 if (value != AUDIO_FORMAT_PCM_16_BIT) { 2354 status = BAD_VALUE; 2355 } else { 2356 reconfig = true; 2357 } 2358 } 2359 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2360 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2361 status = BAD_VALUE; 2362 } else { 2363 reconfig = true; 2364 } 2365 } 2366 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2367 // do not accept frame count changes if tracks are open as the track buffer 2368 // size depends on frame count and correct behavior would not be guaranteed 2369 // if frame count is changed after track creation 2370 if (!mTracks.isEmpty()) { 2371 status = INVALID_OPERATION; 2372 } else { 2373 reconfig = true; 2374 } 2375 } 2376 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2377 // when changing the audio output device, call addBatteryData to notify 2378 // the change 2379 if ((int)mDevice != value) { 2380 uint32_t params = 0; 2381 // check whether speaker is on 2382 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2383 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2384 } 2385 2386 int deviceWithoutSpeaker 2387 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2388 // check if any other device (except speaker) is on 2389 if (value & deviceWithoutSpeaker ) { 2390 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2391 } 2392 2393 if (params != 0) { 2394 addBatteryData(params); 2395 } 2396 } 2397 2398 // forward device change to effects that have requested to be 2399 // aware of attached audio device. 2400 mDevice = (uint32_t)value; 2401 for (size_t i = 0; i < mEffectChains.size(); i++) { 2402 mEffectChains[i]->setDevice_l(mDevice); 2403 } 2404 } 2405 2406 if (status == NO_ERROR) { 2407 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2408 keyValuePair.string()); 2409 if (!mStandby && status == INVALID_OPERATION) { 2410 mOutput->stream->common.standby(&mOutput->stream->common); 2411 mStandby = true; 2412 mBytesWritten = 0; 2413 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2414 keyValuePair.string()); 2415 } 2416 if (status == NO_ERROR && reconfig) { 2417 delete mAudioMixer; 2418 readOutputParameters(); 2419 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2420 for (size_t i = 0; i < mTracks.size() ; i++) { 2421 int name = getTrackName_l(); 2422 if (name < 0) break; 2423 mTracks[i]->mName = name; 2424 // limit track sample rate to 2 x new output sample rate 2425 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2426 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2427 } 2428 } 2429 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2430 } 2431 } 2432 2433 mNewParameters.removeAt(0); 2434 2435 mParamStatus = status; 2436 mParamCond.signal(); 2437 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2438 // already timed out waiting for the status and will never signal the condition. 2439 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2440 } 2441 return reconfig; 2442} 2443 2444status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2445{ 2446 const size_t SIZE = 256; 2447 char buffer[SIZE]; 2448 String8 result; 2449 2450 PlaybackThread::dumpInternals(fd, args); 2451 2452 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2453 result.append(buffer); 2454 write(fd, result.string(), result.size()); 2455 return NO_ERROR; 2456} 2457 2458uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2459{ 2460 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2461} 2462 2463uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2464{ 2465 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2466} 2467 2468// ---------------------------------------------------------------------------- 2469AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2470 : PlaybackThread(audioFlinger, output, id, device) 2471{ 2472 mType = ThreadBase::DIRECT; 2473} 2474 2475AudioFlinger::DirectOutputThread::~DirectOutputThread() 2476{ 2477} 2478 2479static inline 2480int32_t mul(int16_t in, int16_t v) 2481{ 2482#if defined(__arm__) && !defined(__thumb__) 2483 int32_t out; 2484 asm( "smulbb %[out], %[in], %[v] \n" 2485 : [out]"=r"(out) 2486 : [in]"%r"(in), [v]"r"(v) 2487 : ); 2488 return out; 2489#else 2490 return in * int32_t(v); 2491#endif 2492} 2493 2494void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2495{ 2496 // Do not apply volume on compressed audio 2497 if (!audio_is_linear_pcm(mFormat)) { 2498 return; 2499 } 2500 2501 // convert to signed 16 bit before volume calculation 2502 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2503 size_t count = mFrameCount * mChannelCount; 2504 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2505 int16_t *dst = mMixBuffer + count-1; 2506 while(count--) { 2507 *dst-- = (int16_t)(*src--^0x80) << 8; 2508 } 2509 } 2510 2511 size_t frameCount = mFrameCount; 2512 int16_t *out = mMixBuffer; 2513 if (ramp) { 2514 if (mChannelCount == 1) { 2515 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2516 int32_t vlInc = d / (int32_t)frameCount; 2517 int32_t vl = ((int32_t)mLeftVolShort << 16); 2518 do { 2519 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2520 out++; 2521 vl += vlInc; 2522 } while (--frameCount); 2523 2524 } else { 2525 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2526 int32_t vlInc = d / (int32_t)frameCount; 2527 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2528 int32_t vrInc = d / (int32_t)frameCount; 2529 int32_t vl = ((int32_t)mLeftVolShort << 16); 2530 int32_t vr = ((int32_t)mRightVolShort << 16); 2531 do { 2532 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2533 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2534 out += 2; 2535 vl += vlInc; 2536 vr += vrInc; 2537 } while (--frameCount); 2538 } 2539 } else { 2540 if (mChannelCount == 1) { 2541 do { 2542 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2543 out++; 2544 } while (--frameCount); 2545 } else { 2546 do { 2547 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2548 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2549 out += 2; 2550 } while (--frameCount); 2551 } 2552 } 2553 2554 // convert back to unsigned 8 bit after volume calculation 2555 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2556 size_t count = mFrameCount * mChannelCount; 2557 int16_t *src = mMixBuffer; 2558 uint8_t *dst = (uint8_t *)mMixBuffer; 2559 while(count--) { 2560 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2561 } 2562 } 2563 2564 mLeftVolShort = leftVol; 2565 mRightVolShort = rightVol; 2566} 2567 2568bool AudioFlinger::DirectOutputThread::threadLoop() 2569{ 2570 uint32_t mixerStatus = MIXER_IDLE; 2571 sp<Track> trackToRemove; 2572 sp<Track> activeTrack; 2573 nsecs_t standbyTime = systemTime(); 2574 int8_t *curBuf; 2575 size_t mixBufferSize = mFrameCount*mFrameSize; 2576 uint32_t activeSleepTime = activeSleepTimeUs(); 2577 uint32_t idleSleepTime = idleSleepTimeUs(); 2578 uint32_t sleepTime = idleSleepTime; 2579 // use shorter standby delay as on normal output to release 2580 // hardware resources as soon as possible 2581 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2582 2583 acquireWakeLock(); 2584 2585 while (!exitPending()) 2586 { 2587 bool rampVolume; 2588 uint16_t leftVol; 2589 uint16_t rightVol; 2590 Vector< sp<EffectChain> > effectChains; 2591 2592 processConfigEvents(); 2593 2594 mixerStatus = MIXER_IDLE; 2595 2596 { // scope for the mLock 2597 2598 Mutex::Autolock _l(mLock); 2599 2600 if (checkForNewParameters_l()) { 2601 mixBufferSize = mFrameCount*mFrameSize; 2602 activeSleepTime = activeSleepTimeUs(); 2603 idleSleepTime = idleSleepTimeUs(); 2604 standbyDelay = microseconds(activeSleepTime*2); 2605 } 2606 2607 // put audio hardware into standby after short delay 2608 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2609 mSuspended)) { 2610 // wait until we have something to do... 2611 if (!mStandby) { 2612 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2613 mOutput->stream->common.standby(&mOutput->stream->common); 2614 mStandby = true; 2615 mBytesWritten = 0; 2616 } 2617 2618 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2619 // we're about to wait, flush the binder command buffer 2620 IPCThreadState::self()->flushCommands(); 2621 2622 if (exitPending()) break; 2623 2624 releaseWakeLock_l(); 2625 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2626 mWaitWorkCV.wait(mLock); 2627 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2628 acquireWakeLock_l(); 2629 2630 if (mMasterMute == false) { 2631 char value[PROPERTY_VALUE_MAX]; 2632 property_get("ro.audio.silent", value, "0"); 2633 if (atoi(value)) { 2634 ALOGD("Silence is golden"); 2635 setMasterMute(true); 2636 } 2637 } 2638 2639 standbyTime = systemTime() + standbyDelay; 2640 sleepTime = idleSleepTime; 2641 continue; 2642 } 2643 } 2644 2645 effectChains = mEffectChains; 2646 2647 // find out which tracks need to be processed 2648 if (mActiveTracks.size() != 0) { 2649 sp<Track> t = mActiveTracks[0].promote(); 2650 if (t == 0) continue; 2651 2652 Track* const track = t.get(); 2653 audio_track_cblk_t* cblk = track->cblk(); 2654 2655 // The first time a track is added we wait 2656 // for all its buffers to be filled before processing it 2657 if (cblk->framesReady() && track->isReady() && 2658 !track->isPaused() && !track->isTerminated()) 2659 { 2660 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2661 2662 if (track->mFillingUpStatus == Track::FS_FILLED) { 2663 track->mFillingUpStatus = Track::FS_ACTIVE; 2664 mLeftVolFloat = mRightVolFloat = 0; 2665 mLeftVolShort = mRightVolShort = 0; 2666 if (track->mState == TrackBase::RESUMING) { 2667 track->mState = TrackBase::ACTIVE; 2668 rampVolume = true; 2669 } 2670 } else if (cblk->server != 0) { 2671 // If the track is stopped before the first frame was mixed, 2672 // do not apply ramp 2673 rampVolume = true; 2674 } 2675 // compute volume for this track 2676 float left, right; 2677 if (track->isMuted() || mMasterMute || track->isPausing() || 2678 mStreamTypes[track->type()].mute) { 2679 left = right = 0; 2680 if (track->isPausing()) { 2681 track->setPaused(); 2682 } 2683 } else { 2684 float typeVolume = mStreamTypes[track->type()].volume; 2685 float v = mMasterVolume * typeVolume; 2686 float v_clamped = v * cblk->volume[0]; 2687 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2688 left = v_clamped/MAX_GAIN; 2689 v_clamped = v * cblk->volume[1]; 2690 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2691 right = v_clamped/MAX_GAIN; 2692 } 2693 2694 if (left != mLeftVolFloat || right != mRightVolFloat) { 2695 mLeftVolFloat = left; 2696 mRightVolFloat = right; 2697 2698 // If audio HAL implements volume control, 2699 // force software volume to nominal value 2700 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2701 left = 1.0f; 2702 right = 1.0f; 2703 } 2704 2705 // Convert volumes from float to 8.24 2706 uint32_t vl = (uint32_t)(left * (1 << 24)); 2707 uint32_t vr = (uint32_t)(right * (1 << 24)); 2708 2709 // Delegate volume control to effect in track effect chain if needed 2710 // only one effect chain can be present on DirectOutputThread, so if 2711 // there is one, the track is connected to it 2712 if (!effectChains.isEmpty()) { 2713 // Do not ramp volume if volume is controlled by effect 2714 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2715 rampVolume = false; 2716 } 2717 } 2718 2719 // Convert volumes from 8.24 to 4.12 format 2720 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2721 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2722 leftVol = (uint16_t)v_clamped; 2723 v_clamped = (vr + (1 << 11)) >> 12; 2724 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2725 rightVol = (uint16_t)v_clamped; 2726 } else { 2727 leftVol = mLeftVolShort; 2728 rightVol = mRightVolShort; 2729 rampVolume = false; 2730 } 2731 2732 // reset retry count 2733 track->mRetryCount = kMaxTrackRetriesDirect; 2734 activeTrack = t; 2735 mixerStatus = MIXER_TRACKS_READY; 2736 } else { 2737 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2738 if (track->isStopped()) { 2739 track->reset(); 2740 } 2741 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2742 // We have consumed all the buffers of this track. 2743 // Remove it from the list of active tracks. 2744 trackToRemove = track; 2745 } else { 2746 // No buffers for this track. Give it a few chances to 2747 // fill a buffer, then remove it from active list. 2748 if (--(track->mRetryCount) <= 0) { 2749 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2750 trackToRemove = track; 2751 } else { 2752 mixerStatus = MIXER_TRACKS_ENABLED; 2753 } 2754 } 2755 } 2756 } 2757 2758 // remove all the tracks that need to be... 2759 if (CC_UNLIKELY(trackToRemove != 0)) { 2760 mActiveTracks.remove(trackToRemove); 2761 if (!effectChains.isEmpty()) { 2762 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2763 trackToRemove->sessionId()); 2764 effectChains[0]->decActiveTrackCnt(); 2765 } 2766 if (trackToRemove->isTerminated()) { 2767 removeTrack_l(trackToRemove); 2768 } 2769 } 2770 2771 lockEffectChains_l(effectChains); 2772 } 2773 2774 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2775 AudioBufferProvider::Buffer buffer; 2776 size_t frameCount = mFrameCount; 2777 curBuf = (int8_t *)mMixBuffer; 2778 // output audio to hardware 2779 while (frameCount) { 2780 buffer.frameCount = frameCount; 2781 activeTrack->getNextBuffer(&buffer); 2782 if (CC_UNLIKELY(buffer.raw == NULL)) { 2783 memset(curBuf, 0, frameCount * mFrameSize); 2784 break; 2785 } 2786 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2787 frameCount -= buffer.frameCount; 2788 curBuf += buffer.frameCount * mFrameSize; 2789 activeTrack->releaseBuffer(&buffer); 2790 } 2791 sleepTime = 0; 2792 standbyTime = systemTime() + standbyDelay; 2793 } else { 2794 if (sleepTime == 0) { 2795 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2796 sleepTime = activeSleepTime; 2797 } else { 2798 sleepTime = idleSleepTime; 2799 } 2800 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2801 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2802 sleepTime = 0; 2803 } 2804 } 2805 2806 if (mSuspended) { 2807 sleepTime = suspendSleepTimeUs(); 2808 } 2809 // sleepTime == 0 means we must write to audio hardware 2810 if (sleepTime == 0) { 2811 if (mixerStatus == MIXER_TRACKS_READY) { 2812 applyVolume(leftVol, rightVol, rampVolume); 2813 } 2814 for (size_t i = 0; i < effectChains.size(); i ++) { 2815 effectChains[i]->process_l(); 2816 } 2817 unlockEffectChains(effectChains); 2818 2819 mLastWriteTime = systemTime(); 2820 mInWrite = true; 2821 mBytesWritten += mixBufferSize; 2822 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2823 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2824 mNumWrites++; 2825 mInWrite = false; 2826 mStandby = false; 2827 } else { 2828 unlockEffectChains(effectChains); 2829 usleep(sleepTime); 2830 } 2831 2832 // finally let go of removed track, without the lock held 2833 // since we can't guarantee the destructors won't acquire that 2834 // same lock. 2835 trackToRemove.clear(); 2836 activeTrack.clear(); 2837 2838 // Effect chains will be actually deleted here if they were removed from 2839 // mEffectChains list during mixing or effects processing 2840 effectChains.clear(); 2841 } 2842 2843 if (!mStandby) { 2844 mOutput->stream->common.standby(&mOutput->stream->common); 2845 } 2846 2847 releaseWakeLock(); 2848 2849 ALOGV("DirectOutputThread %p exiting", this); 2850 return false; 2851} 2852 2853// getTrackName_l() must be called with ThreadBase::mLock held 2854int AudioFlinger::DirectOutputThread::getTrackName_l() 2855{ 2856 return 0; 2857} 2858 2859// deleteTrackName_l() must be called with ThreadBase::mLock held 2860void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2861{ 2862} 2863 2864// checkForNewParameters_l() must be called with ThreadBase::mLock held 2865bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2866{ 2867 bool reconfig = false; 2868 2869 while (!mNewParameters.isEmpty()) { 2870 status_t status = NO_ERROR; 2871 String8 keyValuePair = mNewParameters[0]; 2872 AudioParameter param = AudioParameter(keyValuePair); 2873 int value; 2874 2875 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2876 // do not accept frame count changes if tracks are open as the track buffer 2877 // size depends on frame count and correct behavior would not be garantied 2878 // if frame count is changed after track creation 2879 if (!mTracks.isEmpty()) { 2880 status = INVALID_OPERATION; 2881 } else { 2882 reconfig = true; 2883 } 2884 } 2885 if (status == NO_ERROR) { 2886 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2887 keyValuePair.string()); 2888 if (!mStandby && status == INVALID_OPERATION) { 2889 mOutput->stream->common.standby(&mOutput->stream->common); 2890 mStandby = true; 2891 mBytesWritten = 0; 2892 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2893 keyValuePair.string()); 2894 } 2895 if (status == NO_ERROR && reconfig) { 2896 readOutputParameters(); 2897 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2898 } 2899 } 2900 2901 mNewParameters.removeAt(0); 2902 2903 mParamStatus = status; 2904 mParamCond.signal(); 2905 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2906 // already timed out waiting for the status and will never signal the condition. 2907 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2908 } 2909 return reconfig; 2910} 2911 2912uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2913{ 2914 uint32_t time; 2915 if (audio_is_linear_pcm(mFormat)) { 2916 time = PlaybackThread::activeSleepTimeUs(); 2917 } else { 2918 time = 10000; 2919 } 2920 return time; 2921} 2922 2923uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2924{ 2925 uint32_t time; 2926 if (audio_is_linear_pcm(mFormat)) { 2927 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2928 } else { 2929 time = 10000; 2930 } 2931 return time; 2932} 2933 2934uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2935{ 2936 uint32_t time; 2937 if (audio_is_linear_pcm(mFormat)) { 2938 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2939 } else { 2940 time = 10000; 2941 } 2942 return time; 2943} 2944 2945 2946// ---------------------------------------------------------------------------- 2947 2948AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2949 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2950{ 2951 mType = ThreadBase::DUPLICATING; 2952 addOutputTrack(mainThread); 2953} 2954 2955AudioFlinger::DuplicatingThread::~DuplicatingThread() 2956{ 2957 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2958 mOutputTracks[i]->destroy(); 2959 } 2960 mOutputTracks.clear(); 2961} 2962 2963bool AudioFlinger::DuplicatingThread::threadLoop() 2964{ 2965 Vector< sp<Track> > tracksToRemove; 2966 uint32_t mixerStatus = MIXER_IDLE; 2967 nsecs_t standbyTime = systemTime(); 2968 size_t mixBufferSize = mFrameCount*mFrameSize; 2969 SortedVector< sp<OutputTrack> > outputTracks; 2970 uint32_t writeFrames = 0; 2971 uint32_t activeSleepTime = activeSleepTimeUs(); 2972 uint32_t idleSleepTime = idleSleepTimeUs(); 2973 uint32_t sleepTime = idleSleepTime; 2974 Vector< sp<EffectChain> > effectChains; 2975 2976 acquireWakeLock(); 2977 2978 while (!exitPending()) 2979 { 2980 processConfigEvents(); 2981 2982 mixerStatus = MIXER_IDLE; 2983 { // scope for the mLock 2984 2985 Mutex::Autolock _l(mLock); 2986 2987 if (checkForNewParameters_l()) { 2988 mixBufferSize = mFrameCount*mFrameSize; 2989 updateWaitTime(); 2990 activeSleepTime = activeSleepTimeUs(); 2991 idleSleepTime = idleSleepTimeUs(); 2992 } 2993 2994 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2995 2996 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2997 outputTracks.add(mOutputTracks[i]); 2998 } 2999 3000 // put audio hardware into standby after short delay 3001 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3002 mSuspended)) { 3003 if (!mStandby) { 3004 for (size_t i = 0; i < outputTracks.size(); i++) { 3005 outputTracks[i]->stop(); 3006 } 3007 mStandby = true; 3008 mBytesWritten = 0; 3009 } 3010 3011 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3012 // we're about to wait, flush the binder command buffer 3013 IPCThreadState::self()->flushCommands(); 3014 outputTracks.clear(); 3015 3016 if (exitPending()) break; 3017 3018 releaseWakeLock_l(); 3019 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3020 mWaitWorkCV.wait(mLock); 3021 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3022 acquireWakeLock_l(); 3023 3024 if (mMasterMute == false) { 3025 char value[PROPERTY_VALUE_MAX]; 3026 property_get("ro.audio.silent", value, "0"); 3027 if (atoi(value)) { 3028 ALOGD("Silence is golden"); 3029 setMasterMute(true); 3030 } 3031 } 3032 3033 standbyTime = systemTime() + kStandbyTimeInNsecs; 3034 sleepTime = idleSleepTime; 3035 continue; 3036 } 3037 } 3038 3039 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3040 3041 // prevent any changes in effect chain list and in each effect chain 3042 // during mixing and effect process as the audio buffers could be deleted 3043 // or modified if an effect is created or deleted 3044 lockEffectChains_l(effectChains); 3045 } 3046 3047 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3048 // mix buffers... 3049 if (outputsReady(outputTracks)) { 3050 mAudioMixer->process(); 3051 } else { 3052 memset(mMixBuffer, 0, mixBufferSize); 3053 } 3054 sleepTime = 0; 3055 writeFrames = mFrameCount; 3056 } else { 3057 if (sleepTime == 0) { 3058 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3059 sleepTime = activeSleepTime; 3060 } else { 3061 sleepTime = idleSleepTime; 3062 } 3063 } else if (mBytesWritten != 0) { 3064 // flush remaining overflow buffers in output tracks 3065 for (size_t i = 0; i < outputTracks.size(); i++) { 3066 if (outputTracks[i]->isActive()) { 3067 sleepTime = 0; 3068 writeFrames = 0; 3069 memset(mMixBuffer, 0, mixBufferSize); 3070 break; 3071 } 3072 } 3073 } 3074 } 3075 3076 if (mSuspended) { 3077 sleepTime = suspendSleepTimeUs(); 3078 } 3079 // sleepTime == 0 means we must write to audio hardware 3080 if (sleepTime == 0) { 3081 for (size_t i = 0; i < effectChains.size(); i ++) { 3082 effectChains[i]->process_l(); 3083 } 3084 // enable changes in effect chain 3085 unlockEffectChains(effectChains); 3086 3087 standbyTime = systemTime() + kStandbyTimeInNsecs; 3088 for (size_t i = 0; i < outputTracks.size(); i++) { 3089 outputTracks[i]->write(mMixBuffer, writeFrames); 3090 } 3091 mStandby = false; 3092 mBytesWritten += mixBufferSize; 3093 } else { 3094 // enable changes in effect chain 3095 unlockEffectChains(effectChains); 3096 usleep(sleepTime); 3097 } 3098 3099 // finally let go of all our tracks, without the lock held 3100 // since we can't guarantee the destructors won't acquire that 3101 // same lock. 3102 tracksToRemove.clear(); 3103 outputTracks.clear(); 3104 3105 // Effect chains will be actually deleted here if they were removed from 3106 // mEffectChains list during mixing or effects processing 3107 effectChains.clear(); 3108 } 3109 3110 releaseWakeLock(); 3111 3112 return false; 3113} 3114 3115void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3116{ 3117 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3118 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3119 this, 3120 mSampleRate, 3121 mFormat, 3122 mChannelMask, 3123 frameCount); 3124 if (outputTrack->cblk() != NULL) { 3125 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3126 mOutputTracks.add(outputTrack); 3127 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3128 updateWaitTime(); 3129 } 3130} 3131 3132void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3133{ 3134 Mutex::Autolock _l(mLock); 3135 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3136 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3137 mOutputTracks[i]->destroy(); 3138 mOutputTracks.removeAt(i); 3139 updateWaitTime(); 3140 return; 3141 } 3142 } 3143 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3144} 3145 3146void AudioFlinger::DuplicatingThread::updateWaitTime() 3147{ 3148 mWaitTimeMs = UINT_MAX; 3149 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3150 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3151 if (strong != NULL) { 3152 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3153 if (waitTimeMs < mWaitTimeMs) { 3154 mWaitTimeMs = waitTimeMs; 3155 } 3156 } 3157 } 3158} 3159 3160 3161bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3162{ 3163 for (size_t i = 0; i < outputTracks.size(); i++) { 3164 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3165 if (thread == 0) { 3166 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3167 return false; 3168 } 3169 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3170 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3171 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3172 return false; 3173 } 3174 } 3175 return true; 3176} 3177 3178uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3179{ 3180 return (mWaitTimeMs * 1000) / 2; 3181} 3182 3183// ---------------------------------------------------------------------------- 3184 3185// TrackBase constructor must be called with AudioFlinger::mLock held 3186AudioFlinger::ThreadBase::TrackBase::TrackBase( 3187 const wp<ThreadBase>& thread, 3188 const sp<Client>& client, 3189 uint32_t sampleRate, 3190 uint32_t format, 3191 uint32_t channelMask, 3192 int frameCount, 3193 uint32_t flags, 3194 const sp<IMemory>& sharedBuffer, 3195 int sessionId) 3196 : RefBase(), 3197 mThread(thread), 3198 mClient(client), 3199 mCblk(0), 3200 mFrameCount(0), 3201 mState(IDLE), 3202 mClientTid(-1), 3203 mFormat(format), 3204 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3205 mSessionId(sessionId) 3206{ 3207 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3208 3209 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3210 size_t size = sizeof(audio_track_cblk_t); 3211 uint8_t channelCount = popcount(channelMask); 3212 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3213 if (sharedBuffer == 0) { 3214 size += bufferSize; 3215 } 3216 3217 if (client != NULL) { 3218 mCblkMemory = client->heap()->allocate(size); 3219 if (mCblkMemory != 0) { 3220 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3221 if (mCblk) { // construct the shared structure in-place. 3222 new(mCblk) audio_track_cblk_t(); 3223 // clear all buffers 3224 mCblk->frameCount = frameCount; 3225 mCblk->sampleRate = sampleRate; 3226 mChannelCount = channelCount; 3227 mChannelMask = channelMask; 3228 if (sharedBuffer == 0) { 3229 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3230 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3231 // Force underrun condition to avoid false underrun callback until first data is 3232 // written to buffer (other flags are cleared) 3233 mCblk->flags = CBLK_UNDERRUN_ON; 3234 } else { 3235 mBuffer = sharedBuffer->pointer(); 3236 } 3237 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3238 } 3239 } else { 3240 ALOGE("not enough memory for AudioTrack size=%u", size); 3241 client->heap()->dump("AudioTrack"); 3242 return; 3243 } 3244 } else { 3245 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3246 // construct the shared structure in-place. 3247 new(mCblk) audio_track_cblk_t(); 3248 // clear all buffers 3249 mCblk->frameCount = frameCount; 3250 mCblk->sampleRate = sampleRate; 3251 mChannelCount = channelCount; 3252 mChannelMask = channelMask; 3253 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3254 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3255 // Force underrun condition to avoid false underrun callback until first data is 3256 // written to buffer (other flags are cleared) 3257 mCblk->flags = CBLK_UNDERRUN_ON; 3258 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3259 } 3260} 3261 3262AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3263{ 3264 if (mCblk) { 3265 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3266 if (mClient == NULL) { 3267 delete mCblk; 3268 } 3269 } 3270 mCblkMemory.clear(); // and free the shared memory 3271 if (mClient != NULL) { 3272 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3273 mClient.clear(); 3274 } 3275} 3276 3277void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3278{ 3279 buffer->raw = NULL; 3280 mFrameCount = buffer->frameCount; 3281 step(); 3282 buffer->frameCount = 0; 3283} 3284 3285bool AudioFlinger::ThreadBase::TrackBase::step() { 3286 bool result; 3287 audio_track_cblk_t* cblk = this->cblk(); 3288 3289 result = cblk->stepServer(mFrameCount); 3290 if (!result) { 3291 ALOGV("stepServer failed acquiring cblk mutex"); 3292 mFlags |= STEPSERVER_FAILED; 3293 } 3294 return result; 3295} 3296 3297void AudioFlinger::ThreadBase::TrackBase::reset() { 3298 audio_track_cblk_t* cblk = this->cblk(); 3299 3300 cblk->user = 0; 3301 cblk->server = 0; 3302 cblk->userBase = 0; 3303 cblk->serverBase = 0; 3304 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3305 ALOGV("TrackBase::reset"); 3306} 3307 3308sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3309{ 3310 return mCblkMemory; 3311} 3312 3313int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3314 return (int)mCblk->sampleRate; 3315} 3316 3317int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3318 return (const int)mChannelCount; 3319} 3320 3321uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3322 return mChannelMask; 3323} 3324 3325void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3326 audio_track_cblk_t* cblk = this->cblk(); 3327 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 3328 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 3329 3330 // Check validity of returned pointer in case the track control block would have been corrupted. 3331 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3332 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 3333 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3334 server %d, serverBase %d, user %d, userBase %d", 3335 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3336 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3337 return 0; 3338 } 3339 3340 return bufferStart; 3341} 3342 3343// ---------------------------------------------------------------------------- 3344 3345// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3346AudioFlinger::PlaybackThread::Track::Track( 3347 const wp<ThreadBase>& thread, 3348 const sp<Client>& client, 3349 int streamType, 3350 uint32_t sampleRate, 3351 uint32_t format, 3352 uint32_t channelMask, 3353 int frameCount, 3354 const sp<IMemory>& sharedBuffer, 3355 int sessionId) 3356 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3357 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3358 mAuxEffectId(0), mHasVolumeController(false) 3359{ 3360 if (mCblk != NULL) { 3361 sp<ThreadBase> baseThread = thread.promote(); 3362 if (baseThread != 0) { 3363 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3364 mName = playbackThread->getTrackName_l(); 3365 mMainBuffer = playbackThread->mixBuffer(); 3366 } 3367 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3368 if (mName < 0) { 3369 ALOGE("no more track names available"); 3370 } 3371 mVolume[0] = 1.0f; 3372 mVolume[1] = 1.0f; 3373 mStreamType = streamType; 3374 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3375 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3376 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3377 } 3378} 3379 3380AudioFlinger::PlaybackThread::Track::~Track() 3381{ 3382 ALOGV("PlaybackThread::Track destructor"); 3383 sp<ThreadBase> thread = mThread.promote(); 3384 if (thread != 0) { 3385 Mutex::Autolock _l(thread->mLock); 3386 mState = TERMINATED; 3387 } 3388} 3389 3390void AudioFlinger::PlaybackThread::Track::destroy() 3391{ 3392 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3393 // by removing it from mTracks vector, so there is a risk that this Tracks's 3394 // desctructor is called. As the destructor needs to lock mLock, 3395 // we must acquire a strong reference on this Track before locking mLock 3396 // here so that the destructor is called only when exiting this function. 3397 // On the other hand, as long as Track::destroy() is only called by 3398 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3399 // this Track with its member mTrack. 3400 sp<Track> keep(this); 3401 { // scope for mLock 3402 sp<ThreadBase> thread = mThread.promote(); 3403 if (thread != 0) { 3404 if (!isOutputTrack()) { 3405 if (mState == ACTIVE || mState == RESUMING) { 3406 AudioSystem::stopOutput(thread->id(), 3407 (audio_stream_type_t)mStreamType, 3408 mSessionId); 3409 3410 // to track the speaker usage 3411 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3412 } 3413 AudioSystem::releaseOutput(thread->id()); 3414 } 3415 Mutex::Autolock _l(thread->mLock); 3416 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3417 playbackThread->destroyTrack_l(this); 3418 } 3419 } 3420} 3421 3422void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3423{ 3424 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", 3425 mName - AudioMixer::TRACK0, 3426 (mClient == NULL) ? getpid() : mClient->pid(), 3427 mStreamType, 3428 mFormat, 3429 mChannelMask, 3430 mSessionId, 3431 mFrameCount, 3432 mState, 3433 mMute, 3434 mFillingUpStatus, 3435 mCblk->sampleRate, 3436 mCblk->volume[0], 3437 mCblk->volume[1], 3438 mCblk->server, 3439 mCblk->user, 3440 (int)mMainBuffer, 3441 (int)mAuxBuffer); 3442} 3443 3444status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3445{ 3446 audio_track_cblk_t* cblk = this->cblk(); 3447 uint32_t framesReady; 3448 uint32_t framesReq = buffer->frameCount; 3449 3450 // Check if last stepServer failed, try to step now 3451 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3452 if (!step()) goto getNextBuffer_exit; 3453 ALOGV("stepServer recovered"); 3454 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3455 } 3456 3457 framesReady = cblk->framesReady(); 3458 3459 if (CC_LIKELY(framesReady)) { 3460 uint32_t s = cblk->server; 3461 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3462 3463 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3464 if (framesReq > framesReady) { 3465 framesReq = framesReady; 3466 } 3467 if (s + framesReq > bufferEnd) { 3468 framesReq = bufferEnd - s; 3469 } 3470 3471 buffer->raw = getBuffer(s, framesReq); 3472 if (buffer->raw == NULL) goto getNextBuffer_exit; 3473 3474 buffer->frameCount = framesReq; 3475 return NO_ERROR; 3476 } 3477 3478getNextBuffer_exit: 3479 buffer->raw = NULL; 3480 buffer->frameCount = 0; 3481 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3482 return NOT_ENOUGH_DATA; 3483} 3484 3485bool AudioFlinger::PlaybackThread::Track::isReady() const { 3486 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3487 3488 if (mCblk->framesReady() >= mCblk->frameCount || 3489 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3490 mFillingUpStatus = FS_FILLED; 3491 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3492 return true; 3493 } 3494 return false; 3495} 3496 3497status_t AudioFlinger::PlaybackThread::Track::start() 3498{ 3499 status_t status = NO_ERROR; 3500 ALOGV("start(%d), calling thread %d session %d", 3501 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3502 sp<ThreadBase> thread = mThread.promote(); 3503 if (thread != 0) { 3504 Mutex::Autolock _l(thread->mLock); 3505 int state = mState; 3506 // here the track could be either new, or restarted 3507 // in both cases "unstop" the track 3508 if (mState == PAUSED) { 3509 mState = TrackBase::RESUMING; 3510 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3511 } else { 3512 mState = TrackBase::ACTIVE; 3513 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3514 } 3515 3516 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3517 thread->mLock.unlock(); 3518 status = AudioSystem::startOutput(thread->id(), 3519 (audio_stream_type_t)mStreamType, 3520 mSessionId); 3521 thread->mLock.lock(); 3522 3523 // to track the speaker usage 3524 if (status == NO_ERROR) { 3525 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3526 } 3527 } 3528 if (status == NO_ERROR) { 3529 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3530 playbackThread->addTrack_l(this); 3531 } else { 3532 mState = state; 3533 } 3534 } else { 3535 status = BAD_VALUE; 3536 } 3537 return status; 3538} 3539 3540void AudioFlinger::PlaybackThread::Track::stop() 3541{ 3542 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3543 sp<ThreadBase> thread = mThread.promote(); 3544 if (thread != 0) { 3545 Mutex::Autolock _l(thread->mLock); 3546 int state = mState; 3547 if (mState > STOPPED) { 3548 mState = STOPPED; 3549 // If the track is not active (PAUSED and buffers full), flush buffers 3550 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3551 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3552 reset(); 3553 } 3554 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3555 } 3556 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3557 thread->mLock.unlock(); 3558 AudioSystem::stopOutput(thread->id(), 3559 (audio_stream_type_t)mStreamType, 3560 mSessionId); 3561 thread->mLock.lock(); 3562 3563 // to track the speaker usage 3564 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3565 } 3566 } 3567} 3568 3569void AudioFlinger::PlaybackThread::Track::pause() 3570{ 3571 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3572 sp<ThreadBase> thread = mThread.promote(); 3573 if (thread != 0) { 3574 Mutex::Autolock _l(thread->mLock); 3575 if (mState == ACTIVE || mState == RESUMING) { 3576 mState = PAUSING; 3577 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3578 if (!isOutputTrack()) { 3579 thread->mLock.unlock(); 3580 AudioSystem::stopOutput(thread->id(), 3581 (audio_stream_type_t)mStreamType, 3582 mSessionId); 3583 thread->mLock.lock(); 3584 3585 // to track the speaker usage 3586 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3587 } 3588 } 3589 } 3590} 3591 3592void AudioFlinger::PlaybackThread::Track::flush() 3593{ 3594 ALOGV("flush(%d)", mName); 3595 sp<ThreadBase> thread = mThread.promote(); 3596 if (thread != 0) { 3597 Mutex::Autolock _l(thread->mLock); 3598 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3599 return; 3600 } 3601 // No point remaining in PAUSED state after a flush => go to 3602 // STOPPED state 3603 mState = STOPPED; 3604 3605 // do not reset the track if it is still in the process of being stopped or paused. 3606 // this will be done by prepareTracks_l() when the track is stopped. 3607 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3608 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3609 reset(); 3610 } 3611 } 3612} 3613 3614void AudioFlinger::PlaybackThread::Track::reset() 3615{ 3616 // Do not reset twice to avoid discarding data written just after a flush and before 3617 // the audioflinger thread detects the track is stopped. 3618 if (!mResetDone) { 3619 TrackBase::reset(); 3620 // Force underrun condition to avoid false underrun callback until first data is 3621 // written to buffer 3622 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3623 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3624 mFillingUpStatus = FS_FILLING; 3625 mResetDone = true; 3626 } 3627} 3628 3629void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3630{ 3631 mMute = muted; 3632} 3633 3634void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3635{ 3636 mVolume[0] = left; 3637 mVolume[1] = right; 3638} 3639 3640status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3641{ 3642 status_t status = DEAD_OBJECT; 3643 sp<ThreadBase> thread = mThread.promote(); 3644 if (thread != 0) { 3645 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3646 status = playbackThread->attachAuxEffect(this, EffectId); 3647 } 3648 return status; 3649} 3650 3651void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3652{ 3653 mAuxEffectId = EffectId; 3654 mAuxBuffer = buffer; 3655} 3656 3657// ---------------------------------------------------------------------------- 3658 3659// RecordTrack constructor must be called with AudioFlinger::mLock held 3660AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3661 const wp<ThreadBase>& thread, 3662 const sp<Client>& client, 3663 uint32_t sampleRate, 3664 uint32_t format, 3665 uint32_t channelMask, 3666 int frameCount, 3667 uint32_t flags, 3668 int sessionId) 3669 : TrackBase(thread, client, sampleRate, format, 3670 channelMask, frameCount, flags, 0, sessionId), 3671 mOverflow(false) 3672{ 3673 if (mCblk != NULL) { 3674 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3675 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3676 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3677 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3678 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3679 } else { 3680 mCblk->frameSize = sizeof(int8_t); 3681 } 3682 } 3683} 3684 3685AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3686{ 3687 sp<ThreadBase> thread = mThread.promote(); 3688 if (thread != 0) { 3689 AudioSystem::releaseInput(thread->id()); 3690 } 3691} 3692 3693status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3694{ 3695 audio_track_cblk_t* cblk = this->cblk(); 3696 uint32_t framesAvail; 3697 uint32_t framesReq = buffer->frameCount; 3698 3699 // Check if last stepServer failed, try to step now 3700 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3701 if (!step()) goto getNextBuffer_exit; 3702 ALOGV("stepServer recovered"); 3703 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3704 } 3705 3706 framesAvail = cblk->framesAvailable_l(); 3707 3708 if (CC_LIKELY(framesAvail)) { 3709 uint32_t s = cblk->server; 3710 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3711 3712 if (framesReq > framesAvail) { 3713 framesReq = framesAvail; 3714 } 3715 if (s + framesReq > bufferEnd) { 3716 framesReq = bufferEnd - s; 3717 } 3718 3719 buffer->raw = getBuffer(s, framesReq); 3720 if (buffer->raw == NULL) goto getNextBuffer_exit; 3721 3722 buffer->frameCount = framesReq; 3723 return NO_ERROR; 3724 } 3725 3726getNextBuffer_exit: 3727 buffer->raw = NULL; 3728 buffer->frameCount = 0; 3729 return NOT_ENOUGH_DATA; 3730} 3731 3732status_t AudioFlinger::RecordThread::RecordTrack::start() 3733{ 3734 sp<ThreadBase> thread = mThread.promote(); 3735 if (thread != 0) { 3736 RecordThread *recordThread = (RecordThread *)thread.get(); 3737 return recordThread->start(this); 3738 } else { 3739 return BAD_VALUE; 3740 } 3741} 3742 3743void AudioFlinger::RecordThread::RecordTrack::stop() 3744{ 3745 sp<ThreadBase> thread = mThread.promote(); 3746 if (thread != 0) { 3747 RecordThread *recordThread = (RecordThread *)thread.get(); 3748 recordThread->stop(this); 3749 TrackBase::reset(); 3750 // Force overerrun condition to avoid false overrun callback until first data is 3751 // read from buffer 3752 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3753 } 3754} 3755 3756void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3757{ 3758 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3759 (mClient == NULL) ? getpid() : mClient->pid(), 3760 mFormat, 3761 mChannelMask, 3762 mSessionId, 3763 mFrameCount, 3764 mState, 3765 mCblk->sampleRate, 3766 mCblk->server, 3767 mCblk->user); 3768} 3769 3770 3771// ---------------------------------------------------------------------------- 3772 3773AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3774 const wp<ThreadBase>& thread, 3775 DuplicatingThread *sourceThread, 3776 uint32_t sampleRate, 3777 uint32_t format, 3778 uint32_t channelMask, 3779 int frameCount) 3780 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3781 mActive(false), mSourceThread(sourceThread) 3782{ 3783 3784 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3785 if (mCblk != NULL) { 3786 mCblk->flags |= CBLK_DIRECTION_OUT; 3787 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3788 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3789 mOutBuffer.frameCount = 0; 3790 playbackThread->mTracks.add(this); 3791 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3792 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3793 mCblk, mBuffer, mCblk->buffers, 3794 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3795 } else { 3796 ALOGW("Error creating output track on thread %p", playbackThread); 3797 } 3798} 3799 3800AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3801{ 3802 clearBufferQueue(); 3803} 3804 3805status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3806{ 3807 status_t status = Track::start(); 3808 if (status != NO_ERROR) { 3809 return status; 3810 } 3811 3812 mActive = true; 3813 mRetryCount = 127; 3814 return status; 3815} 3816 3817void AudioFlinger::PlaybackThread::OutputTrack::stop() 3818{ 3819 Track::stop(); 3820 clearBufferQueue(); 3821 mOutBuffer.frameCount = 0; 3822 mActive = false; 3823} 3824 3825bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3826{ 3827 Buffer *pInBuffer; 3828 Buffer inBuffer; 3829 uint32_t channelCount = mChannelCount; 3830 bool outputBufferFull = false; 3831 inBuffer.frameCount = frames; 3832 inBuffer.i16 = data; 3833 3834 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3835 3836 if (!mActive && frames != 0) { 3837 start(); 3838 sp<ThreadBase> thread = mThread.promote(); 3839 if (thread != 0) { 3840 MixerThread *mixerThread = (MixerThread *)thread.get(); 3841 if (mCblk->frameCount > frames){ 3842 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3843 uint32_t startFrames = (mCblk->frameCount - frames); 3844 pInBuffer = new Buffer; 3845 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3846 pInBuffer->frameCount = startFrames; 3847 pInBuffer->i16 = pInBuffer->mBuffer; 3848 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3849 mBufferQueue.add(pInBuffer); 3850 } else { 3851 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3852 } 3853 } 3854 } 3855 } 3856 3857 while (waitTimeLeftMs) { 3858 // First write pending buffers, then new data 3859 if (mBufferQueue.size()) { 3860 pInBuffer = mBufferQueue.itemAt(0); 3861 } else { 3862 pInBuffer = &inBuffer; 3863 } 3864 3865 if (pInBuffer->frameCount == 0) { 3866 break; 3867 } 3868 3869 if (mOutBuffer.frameCount == 0) { 3870 mOutBuffer.frameCount = pInBuffer->frameCount; 3871 nsecs_t startTime = systemTime(); 3872 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3873 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3874 outputBufferFull = true; 3875 break; 3876 } 3877 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3878 if (waitTimeLeftMs >= waitTimeMs) { 3879 waitTimeLeftMs -= waitTimeMs; 3880 } else { 3881 waitTimeLeftMs = 0; 3882 } 3883 } 3884 3885 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3886 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3887 mCblk->stepUser(outFrames); 3888 pInBuffer->frameCount -= outFrames; 3889 pInBuffer->i16 += outFrames * channelCount; 3890 mOutBuffer.frameCount -= outFrames; 3891 mOutBuffer.i16 += outFrames * channelCount; 3892 3893 if (pInBuffer->frameCount == 0) { 3894 if (mBufferQueue.size()) { 3895 mBufferQueue.removeAt(0); 3896 delete [] pInBuffer->mBuffer; 3897 delete pInBuffer; 3898 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3899 } else { 3900 break; 3901 } 3902 } 3903 } 3904 3905 // If we could not write all frames, allocate a buffer and queue it for next time. 3906 if (inBuffer.frameCount) { 3907 sp<ThreadBase> thread = mThread.promote(); 3908 if (thread != 0 && !thread->standby()) { 3909 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3910 pInBuffer = new Buffer; 3911 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3912 pInBuffer->frameCount = inBuffer.frameCount; 3913 pInBuffer->i16 = pInBuffer->mBuffer; 3914 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3915 mBufferQueue.add(pInBuffer); 3916 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3917 } else { 3918 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3919 } 3920 } 3921 } 3922 3923 // Calling write() with a 0 length buffer, means that no more data will be written: 3924 // If no more buffers are pending, fill output track buffer to make sure it is started 3925 // by output mixer. 3926 if (frames == 0 && mBufferQueue.size() == 0) { 3927 if (mCblk->user < mCblk->frameCount) { 3928 frames = mCblk->frameCount - mCblk->user; 3929 pInBuffer = new Buffer; 3930 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3931 pInBuffer->frameCount = frames; 3932 pInBuffer->i16 = pInBuffer->mBuffer; 3933 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3934 mBufferQueue.add(pInBuffer); 3935 } else if (mActive) { 3936 stop(); 3937 } 3938 } 3939 3940 return outputBufferFull; 3941} 3942 3943status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3944{ 3945 int active; 3946 status_t result; 3947 audio_track_cblk_t* cblk = mCblk; 3948 uint32_t framesReq = buffer->frameCount; 3949 3950// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3951 buffer->frameCount = 0; 3952 3953 uint32_t framesAvail = cblk->framesAvailable(); 3954 3955 3956 if (framesAvail == 0) { 3957 Mutex::Autolock _l(cblk->lock); 3958 goto start_loop_here; 3959 while (framesAvail == 0) { 3960 active = mActive; 3961 if (CC_UNLIKELY(!active)) { 3962 ALOGV("Not active and NO_MORE_BUFFERS"); 3963 return AudioTrack::NO_MORE_BUFFERS; 3964 } 3965 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3966 if (result != NO_ERROR) { 3967 return AudioTrack::NO_MORE_BUFFERS; 3968 } 3969 // read the server count again 3970 start_loop_here: 3971 framesAvail = cblk->framesAvailable_l(); 3972 } 3973 } 3974 3975// if (framesAvail < framesReq) { 3976// return AudioTrack::NO_MORE_BUFFERS; 3977// } 3978 3979 if (framesReq > framesAvail) { 3980 framesReq = framesAvail; 3981 } 3982 3983 uint32_t u = cblk->user; 3984 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3985 3986 if (u + framesReq > bufferEnd) { 3987 framesReq = bufferEnd - u; 3988 } 3989 3990 buffer->frameCount = framesReq; 3991 buffer->raw = (void *)cblk->buffer(u); 3992 return NO_ERROR; 3993} 3994 3995 3996void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3997{ 3998 size_t size = mBufferQueue.size(); 3999 Buffer *pBuffer; 4000 4001 for (size_t i = 0; i < size; i++) { 4002 pBuffer = mBufferQueue.itemAt(i); 4003 delete [] pBuffer->mBuffer; 4004 delete pBuffer; 4005 } 4006 mBufferQueue.clear(); 4007} 4008 4009// ---------------------------------------------------------------------------- 4010 4011AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4012 : RefBase(), 4013 mAudioFlinger(audioFlinger), 4014 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4015 mPid(pid) 4016{ 4017 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4018} 4019 4020// Client destructor must be called with AudioFlinger::mLock held 4021AudioFlinger::Client::~Client() 4022{ 4023 mAudioFlinger->removeClient_l(mPid); 4024} 4025 4026const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4027{ 4028 return mMemoryDealer; 4029} 4030 4031// ---------------------------------------------------------------------------- 4032 4033AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4034 const sp<IAudioFlingerClient>& client, 4035 pid_t pid) 4036 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4037{ 4038} 4039 4040AudioFlinger::NotificationClient::~NotificationClient() 4041{ 4042 mClient.clear(); 4043} 4044 4045void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4046{ 4047 sp<NotificationClient> keep(this); 4048 { 4049 mAudioFlinger->removeNotificationClient(mPid); 4050 } 4051} 4052 4053// ---------------------------------------------------------------------------- 4054 4055AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4056 : BnAudioTrack(), 4057 mTrack(track) 4058{ 4059} 4060 4061AudioFlinger::TrackHandle::~TrackHandle() { 4062 // just stop the track on deletion, associated resources 4063 // will be freed from the main thread once all pending buffers have 4064 // been played. Unless it's not in the active track list, in which 4065 // case we free everything now... 4066 mTrack->destroy(); 4067} 4068 4069status_t AudioFlinger::TrackHandle::start() { 4070 return mTrack->start(); 4071} 4072 4073void AudioFlinger::TrackHandle::stop() { 4074 mTrack->stop(); 4075} 4076 4077void AudioFlinger::TrackHandle::flush() { 4078 mTrack->flush(); 4079} 4080 4081void AudioFlinger::TrackHandle::mute(bool e) { 4082 mTrack->mute(e); 4083} 4084 4085void AudioFlinger::TrackHandle::pause() { 4086 mTrack->pause(); 4087} 4088 4089void AudioFlinger::TrackHandle::setVolume(float left, float right) { 4090 mTrack->setVolume(left, right); 4091} 4092 4093sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4094 return mTrack->getCblk(); 4095} 4096 4097status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4098{ 4099 return mTrack->attachAuxEffect(EffectId); 4100} 4101 4102status_t AudioFlinger::TrackHandle::onTransact( 4103 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4104{ 4105 return BnAudioTrack::onTransact(code, data, reply, flags); 4106} 4107 4108// ---------------------------------------------------------------------------- 4109 4110sp<IAudioRecord> AudioFlinger::openRecord( 4111 pid_t pid, 4112 int input, 4113 uint32_t sampleRate, 4114 uint32_t format, 4115 uint32_t channelMask, 4116 int frameCount, 4117 uint32_t flags, 4118 int *sessionId, 4119 status_t *status) 4120{ 4121 sp<RecordThread::RecordTrack> recordTrack; 4122 sp<RecordHandle> recordHandle; 4123 sp<Client> client; 4124 wp<Client> wclient; 4125 status_t lStatus; 4126 RecordThread *thread; 4127 size_t inFrameCount; 4128 int lSessionId; 4129 4130 // check calling permissions 4131 if (!recordingAllowed()) { 4132 lStatus = PERMISSION_DENIED; 4133 goto Exit; 4134 } 4135 4136 // add client to list 4137 { // scope for mLock 4138 Mutex::Autolock _l(mLock); 4139 thread = checkRecordThread_l(input); 4140 if (thread == NULL) { 4141 lStatus = BAD_VALUE; 4142 goto Exit; 4143 } 4144 4145 wclient = mClients.valueFor(pid); 4146 if (wclient != NULL) { 4147 client = wclient.promote(); 4148 } else { 4149 client = new Client(this, pid); 4150 mClients.add(pid, client); 4151 } 4152 4153 // If no audio session id is provided, create one here 4154 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4155 lSessionId = *sessionId; 4156 } else { 4157 lSessionId = nextUniqueId(); 4158 if (sessionId != NULL) { 4159 *sessionId = lSessionId; 4160 } 4161 } 4162 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4163 recordTrack = thread->createRecordTrack_l(client, 4164 sampleRate, 4165 format, 4166 channelMask, 4167 frameCount, 4168 flags, 4169 lSessionId, 4170 &lStatus); 4171 } 4172 if (lStatus != NO_ERROR) { 4173 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4174 // destructor is called by the TrackBase destructor with mLock held 4175 client.clear(); 4176 recordTrack.clear(); 4177 goto Exit; 4178 } 4179 4180 // return to handle to client 4181 recordHandle = new RecordHandle(recordTrack); 4182 lStatus = NO_ERROR; 4183 4184Exit: 4185 if (status) { 4186 *status = lStatus; 4187 } 4188 return recordHandle; 4189} 4190 4191// ---------------------------------------------------------------------------- 4192 4193AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4194 : BnAudioRecord(), 4195 mRecordTrack(recordTrack) 4196{ 4197} 4198 4199AudioFlinger::RecordHandle::~RecordHandle() { 4200 stop(); 4201} 4202 4203status_t AudioFlinger::RecordHandle::start() { 4204 ALOGV("RecordHandle::start()"); 4205 return mRecordTrack->start(); 4206} 4207 4208void AudioFlinger::RecordHandle::stop() { 4209 ALOGV("RecordHandle::stop()"); 4210 mRecordTrack->stop(); 4211} 4212 4213sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4214 return mRecordTrack->getCblk(); 4215} 4216 4217status_t AudioFlinger::RecordHandle::onTransact( 4218 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4219{ 4220 return BnAudioRecord::onTransact(code, data, reply, flags); 4221} 4222 4223// ---------------------------------------------------------------------------- 4224 4225AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4226 AudioStreamIn *input, 4227 uint32_t sampleRate, 4228 uint32_t channels, 4229 int id, 4230 uint32_t device) : 4231 ThreadBase(audioFlinger, id, device), 4232 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4233{ 4234 mType = ThreadBase::RECORD; 4235 4236 snprintf(mName, kNameLength, "AudioIn_%d", id); 4237 4238 mReqChannelCount = popcount(channels); 4239 mReqSampleRate = sampleRate; 4240 readInputParameters(); 4241} 4242 4243 4244AudioFlinger::RecordThread::~RecordThread() 4245{ 4246 delete[] mRsmpInBuffer; 4247 if (mResampler != NULL) { 4248 delete mResampler; 4249 delete[] mRsmpOutBuffer; 4250 } 4251} 4252 4253void AudioFlinger::RecordThread::onFirstRef() 4254{ 4255 run(mName, PRIORITY_URGENT_AUDIO); 4256} 4257 4258status_t AudioFlinger::RecordThread::readyToRun() 4259{ 4260 status_t status = initCheck(); 4261 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4262 return status; 4263} 4264 4265bool AudioFlinger::RecordThread::threadLoop() 4266{ 4267 AudioBufferProvider::Buffer buffer; 4268 sp<RecordTrack> activeTrack; 4269 Vector< sp<EffectChain> > effectChains; 4270 4271 nsecs_t lastWarning = 0; 4272 4273 acquireWakeLock(); 4274 4275 // start recording 4276 while (!exitPending()) { 4277 4278 processConfigEvents(); 4279 4280 { // scope for mLock 4281 Mutex::Autolock _l(mLock); 4282 checkForNewParameters_l(); 4283 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4284 if (!mStandby) { 4285 mInput->stream->common.standby(&mInput->stream->common); 4286 mStandby = true; 4287 } 4288 4289 if (exitPending()) break; 4290 4291 releaseWakeLock_l(); 4292 ALOGV("RecordThread: loop stopping"); 4293 // go to sleep 4294 mWaitWorkCV.wait(mLock); 4295 ALOGV("RecordThread: loop starting"); 4296 acquireWakeLock_l(); 4297 continue; 4298 } 4299 if (mActiveTrack != 0) { 4300 if (mActiveTrack->mState == TrackBase::PAUSING) { 4301 if (!mStandby) { 4302 mInput->stream->common.standby(&mInput->stream->common); 4303 mStandby = true; 4304 } 4305 mActiveTrack.clear(); 4306 mStartStopCond.broadcast(); 4307 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4308 if (mReqChannelCount != mActiveTrack->channelCount()) { 4309 mActiveTrack.clear(); 4310 mStartStopCond.broadcast(); 4311 } else if (mBytesRead != 0) { 4312 // record start succeeds only if first read from audio input 4313 // succeeds 4314 if (mBytesRead > 0) { 4315 mActiveTrack->mState = TrackBase::ACTIVE; 4316 } else { 4317 mActiveTrack.clear(); 4318 } 4319 mStartStopCond.broadcast(); 4320 } 4321 mStandby = false; 4322 } 4323 } 4324 lockEffectChains_l(effectChains); 4325 } 4326 4327 if (mActiveTrack != 0) { 4328 if (mActiveTrack->mState != TrackBase::ACTIVE && 4329 mActiveTrack->mState != TrackBase::RESUMING) { 4330 unlockEffectChains(effectChains); 4331 usleep(kRecordThreadSleepUs); 4332 continue; 4333 } 4334 for (size_t i = 0; i < effectChains.size(); i ++) { 4335 effectChains[i]->process_l(); 4336 } 4337 4338 buffer.frameCount = mFrameCount; 4339 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4340 size_t framesOut = buffer.frameCount; 4341 if (mResampler == NULL) { 4342 // no resampling 4343 while (framesOut) { 4344 size_t framesIn = mFrameCount - mRsmpInIndex; 4345 if (framesIn) { 4346 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4347 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4348 if (framesIn > framesOut) 4349 framesIn = framesOut; 4350 mRsmpInIndex += framesIn; 4351 framesOut -= framesIn; 4352 if ((int)mChannelCount == mReqChannelCount || 4353 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4354 memcpy(dst, src, framesIn * mFrameSize); 4355 } else { 4356 int16_t *src16 = (int16_t *)src; 4357 int16_t *dst16 = (int16_t *)dst; 4358 if (mChannelCount == 1) { 4359 while (framesIn--) { 4360 *dst16++ = *src16; 4361 *dst16++ = *src16++; 4362 } 4363 } else { 4364 while (framesIn--) { 4365 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4366 src16 += 2; 4367 } 4368 } 4369 } 4370 } 4371 if (framesOut && mFrameCount == mRsmpInIndex) { 4372 if (framesOut == mFrameCount && 4373 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4374 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4375 framesOut = 0; 4376 } else { 4377 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4378 mRsmpInIndex = 0; 4379 } 4380 if (mBytesRead < 0) { 4381 ALOGE("Error reading audio input"); 4382 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4383 // Force input into standby so that it tries to 4384 // recover at next read attempt 4385 mInput->stream->common.standby(&mInput->stream->common); 4386 usleep(kRecordThreadSleepUs); 4387 } 4388 mRsmpInIndex = mFrameCount; 4389 framesOut = 0; 4390 buffer.frameCount = 0; 4391 } 4392 } 4393 } 4394 } else { 4395 // resampling 4396 4397 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4398 // alter output frame count as if we were expecting stereo samples 4399 if (mChannelCount == 1 && mReqChannelCount == 1) { 4400 framesOut >>= 1; 4401 } 4402 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4403 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4404 // are 32 bit aligned which should be always true. 4405 if (mChannelCount == 2 && mReqChannelCount == 1) { 4406 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4407 // the resampler always outputs stereo samples: do post stereo to mono conversion 4408 int16_t *src = (int16_t *)mRsmpOutBuffer; 4409 int16_t *dst = buffer.i16; 4410 while (framesOut--) { 4411 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4412 src += 2; 4413 } 4414 } else { 4415 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4416 } 4417 4418 } 4419 mActiveTrack->releaseBuffer(&buffer); 4420 mActiveTrack->overflow(); 4421 } 4422 // client isn't retrieving buffers fast enough 4423 else { 4424 if (!mActiveTrack->setOverflow()) { 4425 nsecs_t now = systemTime(); 4426 if ((now - lastWarning) > kWarningThrottleNs) { 4427 ALOGW("RecordThread: buffer overflow"); 4428 lastWarning = now; 4429 } 4430 } 4431 // Release the processor for a while before asking for a new buffer. 4432 // This will give the application more chance to read from the buffer and 4433 // clear the overflow. 4434 usleep(kRecordThreadSleepUs); 4435 } 4436 } 4437 // enable changes in effect chain 4438 unlockEffectChains(effectChains); 4439 effectChains.clear(); 4440 } 4441 4442 if (!mStandby) { 4443 mInput->stream->common.standby(&mInput->stream->common); 4444 } 4445 mActiveTrack.clear(); 4446 4447 mStartStopCond.broadcast(); 4448 4449 releaseWakeLock(); 4450 4451 ALOGV("RecordThread %p exiting", this); 4452 return false; 4453} 4454 4455 4456sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4457 const sp<AudioFlinger::Client>& client, 4458 uint32_t sampleRate, 4459 int format, 4460 int channelMask, 4461 int frameCount, 4462 uint32_t flags, 4463 int sessionId, 4464 status_t *status) 4465{ 4466 sp<RecordTrack> track; 4467 status_t lStatus; 4468 4469 lStatus = initCheck(); 4470 if (lStatus != NO_ERROR) { 4471 ALOGE("Audio driver not initialized."); 4472 goto Exit; 4473 } 4474 4475 { // scope for mLock 4476 Mutex::Autolock _l(mLock); 4477 4478 track = new RecordTrack(this, client, sampleRate, 4479 format, channelMask, frameCount, flags, sessionId); 4480 4481 if (track->getCblk() == NULL) { 4482 lStatus = NO_MEMORY; 4483 goto Exit; 4484 } 4485 4486 mTrack = track.get(); 4487 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4488 bool suspend = audio_is_bluetooth_sco_device( 4489 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4490 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4491 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4492 } 4493 lStatus = NO_ERROR; 4494 4495Exit: 4496 if (status) { 4497 *status = lStatus; 4498 } 4499 return track; 4500} 4501 4502status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4503{ 4504 ALOGV("RecordThread::start"); 4505 sp <ThreadBase> strongMe = this; 4506 status_t status = NO_ERROR; 4507 { 4508 AutoMutex lock(mLock); 4509 if (mActiveTrack != 0) { 4510 if (recordTrack != mActiveTrack.get()) { 4511 status = -EBUSY; 4512 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4513 mActiveTrack->mState = TrackBase::ACTIVE; 4514 } 4515 return status; 4516 } 4517 4518 recordTrack->mState = TrackBase::IDLE; 4519 mActiveTrack = recordTrack; 4520 mLock.unlock(); 4521 status_t status = AudioSystem::startInput(mId); 4522 mLock.lock(); 4523 if (status != NO_ERROR) { 4524 mActiveTrack.clear(); 4525 return status; 4526 } 4527 mRsmpInIndex = mFrameCount; 4528 mBytesRead = 0; 4529 if (mResampler != NULL) { 4530 mResampler->reset(); 4531 } 4532 mActiveTrack->mState = TrackBase::RESUMING; 4533 // signal thread to start 4534 ALOGV("Signal record thread"); 4535 mWaitWorkCV.signal(); 4536 // do not wait for mStartStopCond if exiting 4537 if (mExiting) { 4538 mActiveTrack.clear(); 4539 status = INVALID_OPERATION; 4540 goto startError; 4541 } 4542 mStartStopCond.wait(mLock); 4543 if (mActiveTrack == 0) { 4544 ALOGV("Record failed to start"); 4545 status = BAD_VALUE; 4546 goto startError; 4547 } 4548 ALOGV("Record started OK"); 4549 return status; 4550 } 4551startError: 4552 AudioSystem::stopInput(mId); 4553 return status; 4554} 4555 4556void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4557 ALOGV("RecordThread::stop"); 4558 sp <ThreadBase> strongMe = this; 4559 { 4560 AutoMutex lock(mLock); 4561 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4562 mActiveTrack->mState = TrackBase::PAUSING; 4563 // do not wait for mStartStopCond if exiting 4564 if (mExiting) { 4565 return; 4566 } 4567 mStartStopCond.wait(mLock); 4568 // if we have been restarted, recordTrack == mActiveTrack.get() here 4569 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4570 mLock.unlock(); 4571 AudioSystem::stopInput(mId); 4572 mLock.lock(); 4573 ALOGV("Record stopped OK"); 4574 } 4575 } 4576 } 4577} 4578 4579status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4580{ 4581 const size_t SIZE = 256; 4582 char buffer[SIZE]; 4583 String8 result; 4584 pid_t pid = 0; 4585 4586 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4587 result.append(buffer); 4588 4589 if (mActiveTrack != 0) { 4590 result.append("Active Track:\n"); 4591 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4592 mActiveTrack->dump(buffer, SIZE); 4593 result.append(buffer); 4594 4595 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4596 result.append(buffer); 4597 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4598 result.append(buffer); 4599 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4600 result.append(buffer); 4601 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4602 result.append(buffer); 4603 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4604 result.append(buffer); 4605 4606 4607 } else { 4608 result.append("No record client\n"); 4609 } 4610 write(fd, result.string(), result.size()); 4611 4612 dumpBase(fd, args); 4613 dumpEffectChains(fd, args); 4614 4615 return NO_ERROR; 4616} 4617 4618status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4619{ 4620 size_t framesReq = buffer->frameCount; 4621 size_t framesReady = mFrameCount - mRsmpInIndex; 4622 int channelCount; 4623 4624 if (framesReady == 0) { 4625 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4626 if (mBytesRead < 0) { 4627 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4628 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4629 // Force input into standby so that it tries to 4630 // recover at next read attempt 4631 mInput->stream->common.standby(&mInput->stream->common); 4632 usleep(kRecordThreadSleepUs); 4633 } 4634 buffer->raw = NULL; 4635 buffer->frameCount = 0; 4636 return NOT_ENOUGH_DATA; 4637 } 4638 mRsmpInIndex = 0; 4639 framesReady = mFrameCount; 4640 } 4641 4642 if (framesReq > framesReady) { 4643 framesReq = framesReady; 4644 } 4645 4646 if (mChannelCount == 1 && mReqChannelCount == 2) { 4647 channelCount = 1; 4648 } else { 4649 channelCount = 2; 4650 } 4651 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4652 buffer->frameCount = framesReq; 4653 return NO_ERROR; 4654} 4655 4656void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4657{ 4658 mRsmpInIndex += buffer->frameCount; 4659 buffer->frameCount = 0; 4660} 4661 4662bool AudioFlinger::RecordThread::checkForNewParameters_l() 4663{ 4664 bool reconfig = false; 4665 4666 while (!mNewParameters.isEmpty()) { 4667 status_t status = NO_ERROR; 4668 String8 keyValuePair = mNewParameters[0]; 4669 AudioParameter param = AudioParameter(keyValuePair); 4670 int value; 4671 int reqFormat = mFormat; 4672 int reqSamplingRate = mReqSampleRate; 4673 int reqChannelCount = mReqChannelCount; 4674 4675 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4676 reqSamplingRate = value; 4677 reconfig = true; 4678 } 4679 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4680 reqFormat = value; 4681 reconfig = true; 4682 } 4683 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4684 reqChannelCount = popcount(value); 4685 reconfig = true; 4686 } 4687 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4688 // do not accept frame count changes if tracks are open as the track buffer 4689 // size depends on frame count and correct behavior would not be garantied 4690 // if frame count is changed after track creation 4691 if (mActiveTrack != 0) { 4692 status = INVALID_OPERATION; 4693 } else { 4694 reconfig = true; 4695 } 4696 } 4697 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4698 // forward device change to effects that have requested to be 4699 // aware of attached audio device. 4700 for (size_t i = 0; i < mEffectChains.size(); i++) { 4701 mEffectChains[i]->setDevice_l(value); 4702 } 4703 // store input device and output device but do not forward output device to audio HAL. 4704 // Note that status is ignored by the caller for output device 4705 // (see AudioFlinger::setParameters() 4706 if (value & AUDIO_DEVICE_OUT_ALL) { 4707 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4708 status = BAD_VALUE; 4709 } else { 4710 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4711 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4712 if (mTrack != NULL) { 4713 bool suspend = audio_is_bluetooth_sco_device( 4714 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4715 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4716 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4717 } 4718 } 4719 mDevice |= (uint32_t)value; 4720 } 4721 if (status == NO_ERROR) { 4722 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4723 if (status == INVALID_OPERATION) { 4724 mInput->stream->common.standby(&mInput->stream->common); 4725 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4726 } 4727 if (reconfig) { 4728 if (status == BAD_VALUE && 4729 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4730 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4731 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4732 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4733 (reqChannelCount < 3)) { 4734 status = NO_ERROR; 4735 } 4736 if (status == NO_ERROR) { 4737 readInputParameters(); 4738 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4739 } 4740 } 4741 } 4742 4743 mNewParameters.removeAt(0); 4744 4745 mParamStatus = status; 4746 mParamCond.signal(); 4747 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4748 // already timed out waiting for the status and will never signal the condition. 4749 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4750 } 4751 return reconfig; 4752} 4753 4754String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4755{ 4756 char *s; 4757 String8 out_s8 = String8(); 4758 4759 Mutex::Autolock _l(mLock); 4760 if (initCheck() != NO_ERROR) { 4761 return out_s8; 4762 } 4763 4764 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4765 out_s8 = String8(s); 4766 free(s); 4767 return out_s8; 4768} 4769 4770void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4771 AudioSystem::OutputDescriptor desc; 4772 void *param2 = 0; 4773 4774 switch (event) { 4775 case AudioSystem::INPUT_OPENED: 4776 case AudioSystem::INPUT_CONFIG_CHANGED: 4777 desc.channels = mChannelMask; 4778 desc.samplingRate = mSampleRate; 4779 desc.format = mFormat; 4780 desc.frameCount = mFrameCount; 4781 desc.latency = 0; 4782 param2 = &desc; 4783 break; 4784 4785 case AudioSystem::INPUT_CLOSED: 4786 default: 4787 break; 4788 } 4789 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4790} 4791 4792void AudioFlinger::RecordThread::readInputParameters() 4793{ 4794 if (mRsmpInBuffer) delete mRsmpInBuffer; 4795 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4796 if (mResampler) delete mResampler; 4797 mResampler = NULL; 4798 4799 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4800 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4801 mChannelCount = (uint16_t)popcount(mChannelMask); 4802 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4803 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); 4804 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4805 mFrameCount = mInputBytes / mFrameSize; 4806 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4807 4808 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4809 { 4810 int channelCount; 4811 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4812 // stereo to mono post process as the resampler always outputs stereo. 4813 if (mChannelCount == 1 && mReqChannelCount == 2) { 4814 channelCount = 1; 4815 } else { 4816 channelCount = 2; 4817 } 4818 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4819 mResampler->setSampleRate(mSampleRate); 4820 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4821 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4822 4823 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4824 if (mChannelCount == 1 && mReqChannelCount == 1) { 4825 mFrameCount >>= 1; 4826 } 4827 4828 } 4829 mRsmpInIndex = mFrameCount; 4830} 4831 4832unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4833{ 4834 Mutex::Autolock _l(mLock); 4835 if (initCheck() != NO_ERROR) { 4836 return 0; 4837 } 4838 4839 return mInput->stream->get_input_frames_lost(mInput->stream); 4840} 4841 4842uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4843{ 4844 Mutex::Autolock _l(mLock); 4845 uint32_t result = 0; 4846 if (getEffectChain_l(sessionId) != 0) { 4847 result = EFFECT_SESSION; 4848 } 4849 4850 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4851 result |= TRACK_SESSION; 4852 } 4853 4854 return result; 4855} 4856 4857AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4858{ 4859 Mutex::Autolock _l(mLock); 4860 return mTrack; 4861} 4862 4863AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() 4864{ 4865 Mutex::Autolock _l(mLock); 4866 return mInput; 4867} 4868 4869AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4870{ 4871 Mutex::Autolock _l(mLock); 4872 AudioStreamIn *input = mInput; 4873 mInput = NULL; 4874 return input; 4875} 4876 4877// this method must always be called either with ThreadBase mLock held or inside the thread loop 4878audio_stream_t* AudioFlinger::RecordThread::stream() 4879{ 4880 if (mInput == NULL) { 4881 return NULL; 4882 } 4883 return &mInput->stream->common; 4884} 4885 4886 4887// ---------------------------------------------------------------------------- 4888 4889int AudioFlinger::openOutput(uint32_t *pDevices, 4890 uint32_t *pSamplingRate, 4891 uint32_t *pFormat, 4892 uint32_t *pChannels, 4893 uint32_t *pLatencyMs, 4894 uint32_t flags) 4895{ 4896 status_t status; 4897 PlaybackThread *thread = NULL; 4898 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4899 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4900 uint32_t format = pFormat ? *pFormat : 0; 4901 uint32_t channels = pChannels ? *pChannels : 0; 4902 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4903 audio_stream_out_t *outStream; 4904 audio_hw_device_t *outHwDev; 4905 4906 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4907 pDevices ? *pDevices : 0, 4908 samplingRate, 4909 format, 4910 channels, 4911 flags); 4912 4913 if (pDevices == NULL || *pDevices == 0) { 4914 return 0; 4915 } 4916 4917 Mutex::Autolock _l(mLock); 4918 4919 outHwDev = findSuitableHwDev_l(*pDevices); 4920 if (outHwDev == NULL) 4921 return 0; 4922 4923 status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, 4924 &channels, &samplingRate, &outStream); 4925 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4926 outStream, 4927 samplingRate, 4928 format, 4929 channels, 4930 status); 4931 4932 mHardwareStatus = AUDIO_HW_IDLE; 4933 if (outStream != NULL) { 4934 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4935 int id = nextUniqueId(); 4936 4937 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4938 (format != AUDIO_FORMAT_PCM_16_BIT) || 4939 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4940 thread = new DirectOutputThread(this, output, id, *pDevices); 4941 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4942 } else { 4943 thread = new MixerThread(this, output, id, *pDevices); 4944 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4945 } 4946 mPlaybackThreads.add(id, thread); 4947 4948 if (pSamplingRate) *pSamplingRate = samplingRate; 4949 if (pFormat) *pFormat = format; 4950 if (pChannels) *pChannels = channels; 4951 if (pLatencyMs) *pLatencyMs = thread->latency(); 4952 4953 // notify client processes of the new output creation 4954 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4955 return id; 4956 } 4957 4958 return 0; 4959} 4960 4961int AudioFlinger::openDuplicateOutput(int output1, int output2) 4962{ 4963 Mutex::Autolock _l(mLock); 4964 MixerThread *thread1 = checkMixerThread_l(output1); 4965 MixerThread *thread2 = checkMixerThread_l(output2); 4966 4967 if (thread1 == NULL || thread2 == NULL) { 4968 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4969 return 0; 4970 } 4971 4972 int id = nextUniqueId(); 4973 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4974 thread->addOutputTrack(thread2); 4975 mPlaybackThreads.add(id, thread); 4976 // notify client processes of the new output creation 4977 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4978 return id; 4979} 4980 4981status_t AudioFlinger::closeOutput(int output) 4982{ 4983 // keep strong reference on the playback thread so that 4984 // it is not destroyed while exit() is executed 4985 sp <PlaybackThread> thread; 4986 { 4987 Mutex::Autolock _l(mLock); 4988 thread = checkPlaybackThread_l(output); 4989 if (thread == NULL) { 4990 return BAD_VALUE; 4991 } 4992 4993 ALOGV("closeOutput() %d", output); 4994 4995 if (thread->type() == ThreadBase::MIXER) { 4996 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4997 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4998 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4999 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5000 } 5001 } 5002 } 5003 void *param2 = 0; 5004 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5005 mPlaybackThreads.removeItem(output); 5006 } 5007 thread->exit(); 5008 5009 if (thread->type() != ThreadBase::DUPLICATING) { 5010 AudioStreamOut *out = thread->clearOutput(); 5011 // from now on thread->mOutput is NULL 5012 out->hwDev->close_output_stream(out->hwDev, out->stream); 5013 delete out; 5014 } 5015 return NO_ERROR; 5016} 5017 5018status_t AudioFlinger::suspendOutput(int output) 5019{ 5020 Mutex::Autolock _l(mLock); 5021 PlaybackThread *thread = checkPlaybackThread_l(output); 5022 5023 if (thread == NULL) { 5024 return BAD_VALUE; 5025 } 5026 5027 ALOGV("suspendOutput() %d", output); 5028 thread->suspend(); 5029 5030 return NO_ERROR; 5031} 5032 5033status_t AudioFlinger::restoreOutput(int output) 5034{ 5035 Mutex::Autolock _l(mLock); 5036 PlaybackThread *thread = checkPlaybackThread_l(output); 5037 5038 if (thread == NULL) { 5039 return BAD_VALUE; 5040 } 5041 5042 ALOGV("restoreOutput() %d", output); 5043 5044 thread->restore(); 5045 5046 return NO_ERROR; 5047} 5048 5049int AudioFlinger::openInput(uint32_t *pDevices, 5050 uint32_t *pSamplingRate, 5051 uint32_t *pFormat, 5052 uint32_t *pChannels, 5053 uint32_t acoustics) 5054{ 5055 status_t status; 5056 RecordThread *thread = NULL; 5057 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5058 uint32_t format = pFormat ? *pFormat : 0; 5059 uint32_t channels = pChannels ? *pChannels : 0; 5060 uint32_t reqSamplingRate = samplingRate; 5061 uint32_t reqFormat = format; 5062 uint32_t reqChannels = channels; 5063 audio_stream_in_t *inStream; 5064 audio_hw_device_t *inHwDev; 5065 5066 if (pDevices == NULL || *pDevices == 0) { 5067 return 0; 5068 } 5069 5070 Mutex::Autolock _l(mLock); 5071 5072 inHwDev = findSuitableHwDev_l(*pDevices); 5073 if (inHwDev == NULL) 5074 return 0; 5075 5076 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5077 &channels, &samplingRate, 5078 (audio_in_acoustics_t)acoustics, 5079 &inStream); 5080 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5081 inStream, 5082 samplingRate, 5083 format, 5084 channels, 5085 acoustics, 5086 status); 5087 5088 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5089 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5090 // or stereo to mono conversions on 16 bit PCM inputs. 5091 if (inStream == NULL && status == BAD_VALUE && 5092 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5093 (samplingRate <= 2 * reqSamplingRate) && 5094 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5095 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5096 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5097 &channels, &samplingRate, 5098 (audio_in_acoustics_t)acoustics, 5099 &inStream); 5100 } 5101 5102 if (inStream != NULL) { 5103 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5104 5105 int id = nextUniqueId(); 5106 // Start record thread 5107 // RecorThread require both input and output device indication to forward to audio 5108 // pre processing modules 5109 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5110 thread = new RecordThread(this, 5111 input, 5112 reqSamplingRate, 5113 reqChannels, 5114 id, 5115 device); 5116 mRecordThreads.add(id, thread); 5117 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5118 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5119 if (pFormat) *pFormat = format; 5120 if (pChannels) *pChannels = reqChannels; 5121 5122 input->stream->common.standby(&input->stream->common); 5123 5124 // notify client processes of the new input creation 5125 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5126 return id; 5127 } 5128 5129 return 0; 5130} 5131 5132status_t AudioFlinger::closeInput(int input) 5133{ 5134 // keep strong reference on the record thread so that 5135 // it is not destroyed while exit() is executed 5136 sp <RecordThread> thread; 5137 { 5138 Mutex::Autolock _l(mLock); 5139 thread = checkRecordThread_l(input); 5140 if (thread == NULL) { 5141 return BAD_VALUE; 5142 } 5143 5144 ALOGV("closeInput() %d", input); 5145 void *param2 = 0; 5146 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5147 mRecordThreads.removeItem(input); 5148 } 5149 thread->exit(); 5150 5151 AudioStreamIn *in = thread->clearInput(); 5152 // from now on thread->mInput is NULL 5153 in->hwDev->close_input_stream(in->hwDev, in->stream); 5154 delete in; 5155 5156 return NO_ERROR; 5157} 5158 5159status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 5160{ 5161 Mutex::Autolock _l(mLock); 5162 MixerThread *dstThread = checkMixerThread_l(output); 5163 if (dstThread == NULL) { 5164 ALOGW("setStreamOutput() bad output id %d", output); 5165 return BAD_VALUE; 5166 } 5167 5168 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5169 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5170 5171 dstThread->setStreamValid(stream, true); 5172 5173 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5174 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5175 if (thread != dstThread && 5176 thread->type() != ThreadBase::DIRECT) { 5177 MixerThread *srcThread = (MixerThread *)thread; 5178 srcThread->setStreamValid(stream, false); 5179 srcThread->invalidateTracks(stream); 5180 } 5181 } 5182 5183 return NO_ERROR; 5184} 5185 5186 5187int AudioFlinger::newAudioSessionId() 5188{ 5189 return nextUniqueId(); 5190} 5191 5192void AudioFlinger::acquireAudioSessionId(int audioSession) 5193{ 5194 Mutex::Autolock _l(mLock); 5195 int caller = IPCThreadState::self()->getCallingPid(); 5196 ALOGV("acquiring %d from %d", audioSession, caller); 5197 int num = mAudioSessionRefs.size(); 5198 for (int i = 0; i< num; i++) { 5199 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5200 if (ref->sessionid == audioSession && ref->pid == caller) { 5201 ref->cnt++; 5202 ALOGV(" incremented refcount to %d", ref->cnt); 5203 return; 5204 } 5205 } 5206 AudioSessionRef *ref = new AudioSessionRef(); 5207 ref->sessionid = audioSession; 5208 ref->pid = caller; 5209 ref->cnt = 1; 5210 mAudioSessionRefs.push(ref); 5211 ALOGV(" added new entry for %d", ref->sessionid); 5212} 5213 5214void AudioFlinger::releaseAudioSessionId(int audioSession) 5215{ 5216 Mutex::Autolock _l(mLock); 5217 int caller = IPCThreadState::self()->getCallingPid(); 5218 ALOGV("releasing %d from %d", audioSession, caller); 5219 int num = mAudioSessionRefs.size(); 5220 for (int i = 0; i< num; i++) { 5221 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5222 if (ref->sessionid == audioSession && ref->pid == caller) { 5223 ref->cnt--; 5224 ALOGV(" decremented refcount to %d", ref->cnt); 5225 if (ref->cnt == 0) { 5226 mAudioSessionRefs.removeAt(i); 5227 delete ref; 5228 purgeStaleEffects_l(); 5229 } 5230 return; 5231 } 5232 } 5233 ALOGW("session id %d not found for pid %d", audioSession, caller); 5234} 5235 5236void AudioFlinger::purgeStaleEffects_l() { 5237 5238 ALOGV("purging stale effects"); 5239 5240 Vector< sp<EffectChain> > chains; 5241 5242 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5243 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5244 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5245 sp<EffectChain> ec = t->mEffectChains[j]; 5246 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5247 chains.push(ec); 5248 } 5249 } 5250 } 5251 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5252 sp<RecordThread> t = mRecordThreads.valueAt(i); 5253 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5254 sp<EffectChain> ec = t->mEffectChains[j]; 5255 chains.push(ec); 5256 } 5257 } 5258 5259 for (size_t i = 0; i < chains.size(); i++) { 5260 sp<EffectChain> ec = chains[i]; 5261 int sessionid = ec->sessionId(); 5262 sp<ThreadBase> t = ec->mThread.promote(); 5263 if (t == 0) { 5264 continue; 5265 } 5266 size_t numsessionrefs = mAudioSessionRefs.size(); 5267 bool found = false; 5268 for (size_t k = 0; k < numsessionrefs; k++) { 5269 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5270 if (ref->sessionid == sessionid) { 5271 ALOGV(" session %d still exists for %d with %d refs", 5272 sessionid, ref->pid, ref->cnt); 5273 found = true; 5274 break; 5275 } 5276 } 5277 if (!found) { 5278 // remove all effects from the chain 5279 while (ec->mEffects.size()) { 5280 sp<EffectModule> effect = ec->mEffects[0]; 5281 effect->unPin(); 5282 Mutex::Autolock _l (t->mLock); 5283 t->removeEffect_l(effect); 5284 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5285 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5286 if (handle != 0) { 5287 handle->mEffect.clear(); 5288 if (handle->mHasControl && handle->mEnabled) { 5289 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5290 } 5291 } 5292 } 5293 AudioSystem::unregisterEffect(effect->id()); 5294 } 5295 } 5296 } 5297 return; 5298} 5299 5300// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5301AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5302{ 5303 PlaybackThread *thread = NULL; 5304 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5305 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5306 } 5307 return thread; 5308} 5309 5310// checkMixerThread_l() must be called with AudioFlinger::mLock held 5311AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5312{ 5313 PlaybackThread *thread = checkPlaybackThread_l(output); 5314 if (thread != NULL) { 5315 if (thread->type() == ThreadBase::DIRECT) { 5316 thread = NULL; 5317 } 5318 } 5319 return (MixerThread *)thread; 5320} 5321 5322// checkRecordThread_l() must be called with AudioFlinger::mLock held 5323AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5324{ 5325 RecordThread *thread = NULL; 5326 if (mRecordThreads.indexOfKey(input) >= 0) { 5327 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5328 } 5329 return thread; 5330} 5331 5332uint32_t AudioFlinger::nextUniqueId() 5333{ 5334 return android_atomic_inc(&mNextUniqueId); 5335} 5336 5337AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5338{ 5339 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5340 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5341 AudioStreamOut *output = thread->getOutput(); 5342 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5343 return thread; 5344 } 5345 } 5346 return NULL; 5347} 5348 5349uint32_t AudioFlinger::primaryOutputDevice_l() 5350{ 5351 PlaybackThread *thread = primaryPlaybackThread_l(); 5352 5353 if (thread == NULL) { 5354 return 0; 5355 } 5356 5357 return thread->device(); 5358} 5359 5360 5361// ---------------------------------------------------------------------------- 5362// Effect management 5363// ---------------------------------------------------------------------------- 5364 5365 5366status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5367{ 5368 Mutex::Autolock _l(mLock); 5369 return EffectQueryNumberEffects(numEffects); 5370} 5371 5372status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5373{ 5374 Mutex::Autolock _l(mLock); 5375 return EffectQueryEffect(index, descriptor); 5376} 5377 5378status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5379{ 5380 Mutex::Autolock _l(mLock); 5381 return EffectGetDescriptor(pUuid, descriptor); 5382} 5383 5384 5385sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5386 effect_descriptor_t *pDesc, 5387 const sp<IEffectClient>& effectClient, 5388 int32_t priority, 5389 int io, 5390 int sessionId, 5391 status_t *status, 5392 int *id, 5393 int *enabled) 5394{ 5395 status_t lStatus = NO_ERROR; 5396 sp<EffectHandle> handle; 5397 effect_descriptor_t desc; 5398 sp<Client> client; 5399 wp<Client> wclient; 5400 5401 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5402 pid, effectClient.get(), priority, sessionId, io); 5403 5404 if (pDesc == NULL) { 5405 lStatus = BAD_VALUE; 5406 goto Exit; 5407 } 5408 5409 // check audio settings permission for global effects 5410 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5411 lStatus = PERMISSION_DENIED; 5412 goto Exit; 5413 } 5414 5415 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5416 // that can only be created by audio policy manager (running in same process) 5417 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5418 lStatus = PERMISSION_DENIED; 5419 goto Exit; 5420 } 5421 5422 if (io == 0) { 5423 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5424 // output must be specified by AudioPolicyManager when using session 5425 // AUDIO_SESSION_OUTPUT_STAGE 5426 lStatus = BAD_VALUE; 5427 goto Exit; 5428 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5429 // if the output returned by getOutputForEffect() is removed before we lock the 5430 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5431 // and we will exit safely 5432 io = AudioSystem::getOutputForEffect(&desc); 5433 } 5434 } 5435 5436 { 5437 Mutex::Autolock _l(mLock); 5438 5439 5440 if (!EffectIsNullUuid(&pDesc->uuid)) { 5441 // if uuid is specified, request effect descriptor 5442 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5443 if (lStatus < 0) { 5444 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5445 goto Exit; 5446 } 5447 } else { 5448 // if uuid is not specified, look for an available implementation 5449 // of the required type in effect factory 5450 if (EffectIsNullUuid(&pDesc->type)) { 5451 ALOGW("createEffect() no effect type"); 5452 lStatus = BAD_VALUE; 5453 goto Exit; 5454 } 5455 uint32_t numEffects = 0; 5456 effect_descriptor_t d; 5457 d.flags = 0; // prevent compiler warning 5458 bool found = false; 5459 5460 lStatus = EffectQueryNumberEffects(&numEffects); 5461 if (lStatus < 0) { 5462 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5463 goto Exit; 5464 } 5465 for (uint32_t i = 0; i < numEffects; i++) { 5466 lStatus = EffectQueryEffect(i, &desc); 5467 if (lStatus < 0) { 5468 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5469 continue; 5470 } 5471 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5472 // If matching type found save effect descriptor. If the session is 5473 // 0 and the effect is not auxiliary, continue enumeration in case 5474 // an auxiliary version of this effect type is available 5475 found = true; 5476 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5477 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5478 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5479 break; 5480 } 5481 } 5482 } 5483 if (!found) { 5484 lStatus = BAD_VALUE; 5485 ALOGW("createEffect() effect not found"); 5486 goto Exit; 5487 } 5488 // For same effect type, chose auxiliary version over insert version if 5489 // connect to output mix (Compliance to OpenSL ES) 5490 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5491 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5492 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5493 } 5494 } 5495 5496 // Do not allow auxiliary effects on a session different from 0 (output mix) 5497 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5498 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5499 lStatus = INVALID_OPERATION; 5500 goto Exit; 5501 } 5502 5503 // check recording permission for visualizer 5504 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5505 !recordingAllowed()) { 5506 lStatus = PERMISSION_DENIED; 5507 goto Exit; 5508 } 5509 5510 // return effect descriptor 5511 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5512 5513 // If output is not specified try to find a matching audio session ID in one of the 5514 // output threads. 5515 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5516 // because of code checking output when entering the function. 5517 // Note: io is never 0 when creating an effect on an input 5518 if (io == 0) { 5519 // look for the thread where the specified audio session is present 5520 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5521 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5522 io = mPlaybackThreads.keyAt(i); 5523 break; 5524 } 5525 } 5526 if (io == 0) { 5527 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5528 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5529 io = mRecordThreads.keyAt(i); 5530 break; 5531 } 5532 } 5533 } 5534 // If no output thread contains the requested session ID, default to 5535 // first output. The effect chain will be moved to the correct output 5536 // thread when a track with the same session ID is created 5537 if (io == 0 && mPlaybackThreads.size()) { 5538 io = mPlaybackThreads.keyAt(0); 5539 } 5540 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5541 } 5542 ThreadBase *thread = checkRecordThread_l(io); 5543 if (thread == NULL) { 5544 thread = checkPlaybackThread_l(io); 5545 if (thread == NULL) { 5546 ALOGE("createEffect() unknown output thread"); 5547 lStatus = BAD_VALUE; 5548 goto Exit; 5549 } 5550 } 5551 5552 wclient = mClients.valueFor(pid); 5553 5554 if (wclient != NULL) { 5555 client = wclient.promote(); 5556 } else { 5557 client = new Client(this, pid); 5558 mClients.add(pid, client); 5559 } 5560 5561 // create effect on selected output thread 5562 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5563 &desc, enabled, &lStatus); 5564 if (handle != 0 && id != NULL) { 5565 *id = handle->id(); 5566 } 5567 } 5568 5569Exit: 5570 if(status) { 5571 *status = lStatus; 5572 } 5573 return handle; 5574} 5575 5576status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5577{ 5578 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5579 sessionId, srcOutput, dstOutput); 5580 Mutex::Autolock _l(mLock); 5581 if (srcOutput == dstOutput) { 5582 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5583 return NO_ERROR; 5584 } 5585 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5586 if (srcThread == NULL) { 5587 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5588 return BAD_VALUE; 5589 } 5590 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5591 if (dstThread == NULL) { 5592 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5593 return BAD_VALUE; 5594 } 5595 5596 Mutex::Autolock _dl(dstThread->mLock); 5597 Mutex::Autolock _sl(srcThread->mLock); 5598 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5599 5600 return NO_ERROR; 5601} 5602 5603// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5604status_t AudioFlinger::moveEffectChain_l(int sessionId, 5605 AudioFlinger::PlaybackThread *srcThread, 5606 AudioFlinger::PlaybackThread *dstThread, 5607 bool reRegister) 5608{ 5609 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5610 sessionId, srcThread, dstThread); 5611 5612 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5613 if (chain == 0) { 5614 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5615 sessionId, srcThread); 5616 return INVALID_OPERATION; 5617 } 5618 5619 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5620 // so that a new chain is created with correct parameters when first effect is added. This is 5621 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5622 // removed. 5623 srcThread->removeEffectChain_l(chain); 5624 5625 // transfer all effects one by one so that new effect chain is created on new thread with 5626 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5627 int dstOutput = dstThread->id(); 5628 sp<EffectChain> dstChain; 5629 uint32_t strategy = 0; // prevent compiler warning 5630 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5631 while (effect != 0) { 5632 srcThread->removeEffect_l(effect); 5633 dstThread->addEffect_l(effect); 5634 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5635 if (effect->state() == EffectModule::ACTIVE || 5636 effect->state() == EffectModule::STOPPING) { 5637 effect->start(); 5638 } 5639 // if the move request is not received from audio policy manager, the effect must be 5640 // re-registered with the new strategy and output 5641 if (dstChain == 0) { 5642 dstChain = effect->chain().promote(); 5643 if (dstChain == 0) { 5644 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5645 srcThread->addEffect_l(effect); 5646 return NO_INIT; 5647 } 5648 strategy = dstChain->strategy(); 5649 } 5650 if (reRegister) { 5651 AudioSystem::unregisterEffect(effect->id()); 5652 AudioSystem::registerEffect(&effect->desc(), 5653 dstOutput, 5654 strategy, 5655 sessionId, 5656 effect->id()); 5657 } 5658 effect = chain->getEffectFromId_l(0); 5659 } 5660 5661 return NO_ERROR; 5662} 5663 5664 5665// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5666sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5667 const sp<AudioFlinger::Client>& client, 5668 const sp<IEffectClient>& effectClient, 5669 int32_t priority, 5670 int sessionId, 5671 effect_descriptor_t *desc, 5672 int *enabled, 5673 status_t *status 5674 ) 5675{ 5676 sp<EffectModule> effect; 5677 sp<EffectHandle> handle; 5678 status_t lStatus; 5679 sp<EffectChain> chain; 5680 bool chainCreated = false; 5681 bool effectCreated = false; 5682 bool effectRegistered = false; 5683 5684 lStatus = initCheck(); 5685 if (lStatus != NO_ERROR) { 5686 ALOGW("createEffect_l() Audio driver not initialized."); 5687 goto Exit; 5688 } 5689 5690 // Do not allow effects with session ID 0 on direct output or duplicating threads 5691 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5692 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5693 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5694 desc->name, sessionId); 5695 lStatus = BAD_VALUE; 5696 goto Exit; 5697 } 5698 // Only Pre processor effects are allowed on input threads and only on input threads 5699 if ((mType == RECORD && 5700 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5701 (mType != RECORD && 5702 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5703 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5704 desc->name, desc->flags, mType); 5705 lStatus = BAD_VALUE; 5706 goto Exit; 5707 } 5708 5709 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5710 5711 { // scope for mLock 5712 Mutex::Autolock _l(mLock); 5713 5714 // check for existing effect chain with the requested audio session 5715 chain = getEffectChain_l(sessionId); 5716 if (chain == 0) { 5717 // create a new chain for this session 5718 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5719 chain = new EffectChain(this, sessionId); 5720 addEffectChain_l(chain); 5721 chain->setStrategy(getStrategyForSession_l(sessionId)); 5722 chainCreated = true; 5723 } else { 5724 effect = chain->getEffectFromDesc_l(desc); 5725 } 5726 5727 ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 5728 5729 if (effect == 0) { 5730 int id = mAudioFlinger->nextUniqueId(); 5731 // Check CPU and memory usage 5732 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5733 if (lStatus != NO_ERROR) { 5734 goto Exit; 5735 } 5736 effectRegistered = true; 5737 // create a new effect module if none present in the chain 5738 effect = new EffectModule(this, chain, desc, id, sessionId); 5739 lStatus = effect->status(); 5740 if (lStatus != NO_ERROR) { 5741 goto Exit; 5742 } 5743 lStatus = chain->addEffect_l(effect); 5744 if (lStatus != NO_ERROR) { 5745 goto Exit; 5746 } 5747 effectCreated = true; 5748 5749 effect->setDevice(mDevice); 5750 effect->setMode(mAudioFlinger->getMode()); 5751 } 5752 // create effect handle and connect it to effect module 5753 handle = new EffectHandle(effect, client, effectClient, priority); 5754 lStatus = effect->addHandle(handle); 5755 if (enabled) { 5756 *enabled = (int)effect->isEnabled(); 5757 } 5758 } 5759 5760Exit: 5761 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5762 Mutex::Autolock _l(mLock); 5763 if (effectCreated) { 5764 chain->removeEffect_l(effect); 5765 } 5766 if (effectRegistered) { 5767 AudioSystem::unregisterEffect(effect->id()); 5768 } 5769 if (chainCreated) { 5770 removeEffectChain_l(chain); 5771 } 5772 handle.clear(); 5773 } 5774 5775 if(status) { 5776 *status = lStatus; 5777 } 5778 return handle; 5779} 5780 5781sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5782{ 5783 sp<EffectModule> effect; 5784 5785 sp<EffectChain> chain = getEffectChain_l(sessionId); 5786 if (chain != 0) { 5787 effect = chain->getEffectFromId_l(effectId); 5788 } 5789 return effect; 5790} 5791 5792// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5793// PlaybackThread::mLock held 5794status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5795{ 5796 // check for existing effect chain with the requested audio session 5797 int sessionId = effect->sessionId(); 5798 sp<EffectChain> chain = getEffectChain_l(sessionId); 5799 bool chainCreated = false; 5800 5801 if (chain == 0) { 5802 // create a new chain for this session 5803 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5804 chain = new EffectChain(this, sessionId); 5805 addEffectChain_l(chain); 5806 chain->setStrategy(getStrategyForSession_l(sessionId)); 5807 chainCreated = true; 5808 } 5809 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5810 5811 if (chain->getEffectFromId_l(effect->id()) != 0) { 5812 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5813 this, effect->desc().name, chain.get()); 5814 return BAD_VALUE; 5815 } 5816 5817 status_t status = chain->addEffect_l(effect); 5818 if (status != NO_ERROR) { 5819 if (chainCreated) { 5820 removeEffectChain_l(chain); 5821 } 5822 return status; 5823 } 5824 5825 effect->setDevice(mDevice); 5826 effect->setMode(mAudioFlinger->getMode()); 5827 return NO_ERROR; 5828} 5829 5830void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5831 5832 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5833 effect_descriptor_t desc = effect->desc(); 5834 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5835 detachAuxEffect_l(effect->id()); 5836 } 5837 5838 sp<EffectChain> chain = effect->chain().promote(); 5839 if (chain != 0) { 5840 // remove effect chain if removing last effect 5841 if (chain->removeEffect_l(effect) == 0) { 5842 removeEffectChain_l(chain); 5843 } 5844 } else { 5845 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5846 } 5847} 5848 5849void AudioFlinger::ThreadBase::lockEffectChains_l( 5850 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5851{ 5852 effectChains = mEffectChains; 5853 for (size_t i = 0; i < mEffectChains.size(); i++) { 5854 mEffectChains[i]->lock(); 5855 } 5856} 5857 5858void AudioFlinger::ThreadBase::unlockEffectChains( 5859 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5860{ 5861 for (size_t i = 0; i < effectChains.size(); i++) { 5862 effectChains[i]->unlock(); 5863 } 5864} 5865 5866sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5867{ 5868 Mutex::Autolock _l(mLock); 5869 return getEffectChain_l(sessionId); 5870} 5871 5872sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5873{ 5874 sp<EffectChain> chain; 5875 5876 size_t size = mEffectChains.size(); 5877 for (size_t i = 0; i < size; i++) { 5878 if (mEffectChains[i]->sessionId() == sessionId) { 5879 chain = mEffectChains[i]; 5880 break; 5881 } 5882 } 5883 return chain; 5884} 5885 5886void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5887{ 5888 Mutex::Autolock _l(mLock); 5889 size_t size = mEffectChains.size(); 5890 for (size_t i = 0; i < size; i++) { 5891 mEffectChains[i]->setMode_l(mode); 5892 } 5893} 5894 5895void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5896 const wp<EffectHandle>& handle, 5897 bool unpiniflast) { 5898 5899 Mutex::Autolock _l(mLock); 5900 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5901 // delete the effect module if removing last handle on it 5902 if (effect->removeHandle(handle) == 0) { 5903 if (!effect->isPinned() || unpiniflast) { 5904 removeEffect_l(effect); 5905 AudioSystem::unregisterEffect(effect->id()); 5906 } 5907 } 5908} 5909 5910status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5911{ 5912 int session = chain->sessionId(); 5913 int16_t *buffer = mMixBuffer; 5914 bool ownsBuffer = false; 5915 5916 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5917 if (session > 0) { 5918 // Only one effect chain can be present in direct output thread and it uses 5919 // the mix buffer as input 5920 if (mType != DIRECT) { 5921 size_t numSamples = mFrameCount * mChannelCount; 5922 buffer = new int16_t[numSamples]; 5923 memset(buffer, 0, numSamples * sizeof(int16_t)); 5924 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5925 ownsBuffer = true; 5926 } 5927 5928 // Attach all tracks with same session ID to this chain. 5929 for (size_t i = 0; i < mTracks.size(); ++i) { 5930 sp<Track> track = mTracks[i]; 5931 if (session == track->sessionId()) { 5932 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5933 track->setMainBuffer(buffer); 5934 chain->incTrackCnt(); 5935 } 5936 } 5937 5938 // indicate all active tracks in the chain 5939 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5940 sp<Track> track = mActiveTracks[i].promote(); 5941 if (track == 0) continue; 5942 if (session == track->sessionId()) { 5943 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5944 chain->incActiveTrackCnt(); 5945 } 5946 } 5947 } 5948 5949 chain->setInBuffer(buffer, ownsBuffer); 5950 chain->setOutBuffer(mMixBuffer); 5951 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5952 // chains list in order to be processed last as it contains output stage effects 5953 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5954 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5955 // after track specific effects and before output stage 5956 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5957 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5958 // Effect chain for other sessions are inserted at beginning of effect 5959 // chains list to be processed before output mix effects. Relative order between other 5960 // sessions is not important 5961 size_t size = mEffectChains.size(); 5962 size_t i = 0; 5963 for (i = 0; i < size; i++) { 5964 if (mEffectChains[i]->sessionId() < session) break; 5965 } 5966 mEffectChains.insertAt(chain, i); 5967 checkSuspendOnAddEffectChain_l(chain); 5968 5969 return NO_ERROR; 5970} 5971 5972size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5973{ 5974 int session = chain->sessionId(); 5975 5976 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5977 5978 for (size_t i = 0; i < mEffectChains.size(); i++) { 5979 if (chain == mEffectChains[i]) { 5980 mEffectChains.removeAt(i); 5981 // detach all active tracks from the chain 5982 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5983 sp<Track> track = mActiveTracks[i].promote(); 5984 if (track == 0) continue; 5985 if (session == track->sessionId()) { 5986 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5987 chain.get(), session); 5988 chain->decActiveTrackCnt(); 5989 } 5990 } 5991 5992 // detach all tracks with same session ID from this chain 5993 for (size_t i = 0; i < mTracks.size(); ++i) { 5994 sp<Track> track = mTracks[i]; 5995 if (session == track->sessionId()) { 5996 track->setMainBuffer(mMixBuffer); 5997 chain->decTrackCnt(); 5998 } 5999 } 6000 break; 6001 } 6002 } 6003 return mEffectChains.size(); 6004} 6005 6006status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6007 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6008{ 6009 Mutex::Autolock _l(mLock); 6010 return attachAuxEffect_l(track, EffectId); 6011} 6012 6013status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6014 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6015{ 6016 status_t status = NO_ERROR; 6017 6018 if (EffectId == 0) { 6019 track->setAuxBuffer(0, NULL); 6020 } else { 6021 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6022 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6023 if (effect != 0) { 6024 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6025 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6026 } else { 6027 status = INVALID_OPERATION; 6028 } 6029 } else { 6030 status = BAD_VALUE; 6031 } 6032 } 6033 return status; 6034} 6035 6036void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6037{ 6038 for (size_t i = 0; i < mTracks.size(); ++i) { 6039 sp<Track> track = mTracks[i]; 6040 if (track->auxEffectId() == effectId) { 6041 attachAuxEffect_l(track, 0); 6042 } 6043 } 6044} 6045 6046status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6047{ 6048 // only one chain per input thread 6049 if (mEffectChains.size() != 0) { 6050 return INVALID_OPERATION; 6051 } 6052 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6053 6054 chain->setInBuffer(NULL); 6055 chain->setOutBuffer(NULL); 6056 6057 checkSuspendOnAddEffectChain_l(chain); 6058 6059 mEffectChains.add(chain); 6060 6061 return NO_ERROR; 6062} 6063 6064size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6065{ 6066 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6067 ALOGW_IF(mEffectChains.size() != 1, 6068 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6069 chain.get(), mEffectChains.size(), this); 6070 if (mEffectChains.size() == 1) { 6071 mEffectChains.removeAt(0); 6072 } 6073 return 0; 6074} 6075 6076// ---------------------------------------------------------------------------- 6077// EffectModule implementation 6078// ---------------------------------------------------------------------------- 6079 6080#undef LOG_TAG 6081#define LOG_TAG "AudioFlinger::EffectModule" 6082 6083AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6084 const wp<AudioFlinger::EffectChain>& chain, 6085 effect_descriptor_t *desc, 6086 int id, 6087 int sessionId) 6088 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6089 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6090{ 6091 ALOGV("Constructor %p", this); 6092 int lStatus; 6093 sp<ThreadBase> thread = mThread.promote(); 6094 if (thread == 0) { 6095 return; 6096 } 6097 6098 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6099 6100 // create effect engine from effect factory 6101 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6102 6103 if (mStatus != NO_ERROR) { 6104 return; 6105 } 6106 lStatus = init(); 6107 if (lStatus < 0) { 6108 mStatus = lStatus; 6109 goto Error; 6110 } 6111 6112 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6113 mPinned = true; 6114 } 6115 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6116 return; 6117Error: 6118 EffectRelease(mEffectInterface); 6119 mEffectInterface = NULL; 6120 ALOGV("Constructor Error %d", mStatus); 6121} 6122 6123AudioFlinger::EffectModule::~EffectModule() 6124{ 6125 ALOGV("Destructor %p", this); 6126 if (mEffectInterface != NULL) { 6127 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6128 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6129 sp<ThreadBase> thread = mThread.promote(); 6130 if (thread != 0) { 6131 audio_stream_t *stream = thread->stream(); 6132 if (stream != NULL) { 6133 stream->remove_audio_effect(stream, mEffectInterface); 6134 } 6135 } 6136 } 6137 // release effect engine 6138 EffectRelease(mEffectInterface); 6139 } 6140} 6141 6142status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6143{ 6144 status_t status; 6145 6146 Mutex::Autolock _l(mLock); 6147 // First handle in mHandles has highest priority and controls the effect module 6148 int priority = handle->priority(); 6149 size_t size = mHandles.size(); 6150 sp<EffectHandle> h; 6151 size_t i; 6152 for (i = 0; i < size; i++) { 6153 h = mHandles[i].promote(); 6154 if (h == 0) continue; 6155 if (h->priority() <= priority) break; 6156 } 6157 // if inserted in first place, move effect control from previous owner to this handle 6158 if (i == 0) { 6159 bool enabled = false; 6160 if (h != 0) { 6161 enabled = h->enabled(); 6162 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6163 } 6164 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6165 status = NO_ERROR; 6166 } else { 6167 status = ALREADY_EXISTS; 6168 } 6169 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6170 mHandles.insertAt(handle, i); 6171 return status; 6172} 6173 6174size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6175{ 6176 Mutex::Autolock _l(mLock); 6177 size_t size = mHandles.size(); 6178 size_t i; 6179 for (i = 0; i < size; i++) { 6180 if (mHandles[i] == handle) break; 6181 } 6182 if (i == size) { 6183 return size; 6184 } 6185 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6186 6187 bool enabled = false; 6188 EffectHandle *hdl = handle.unsafe_get(); 6189 if (hdl) { 6190 ALOGV("removeHandle() unsafe_get OK"); 6191 enabled = hdl->enabled(); 6192 } 6193 mHandles.removeAt(i); 6194 size = mHandles.size(); 6195 // if removed from first place, move effect control from this handle to next in line 6196 if (i == 0 && size != 0) { 6197 sp<EffectHandle> h = mHandles[0].promote(); 6198 if (h != 0) { 6199 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6200 } 6201 } 6202 6203 // Prevent calls to process() and other functions on effect interface from now on. 6204 // The effect engine will be released by the destructor when the last strong reference on 6205 // this object is released which can happen after next process is called. 6206 if (size == 0 && !mPinned) { 6207 mState = DESTROYED; 6208 } 6209 6210 return size; 6211} 6212 6213sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6214{ 6215 Mutex::Autolock _l(mLock); 6216 sp<EffectHandle> handle; 6217 if (mHandles.size() != 0) { 6218 handle = mHandles[0].promote(); 6219 } 6220 return handle; 6221} 6222 6223void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6224{ 6225 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6226 // keep a strong reference on this EffectModule to avoid calling the 6227 // destructor before we exit 6228 sp<EffectModule> keep(this); 6229 { 6230 sp<ThreadBase> thread = mThread.promote(); 6231 if (thread != 0) { 6232 thread->disconnectEffect(keep, handle, unpiniflast); 6233 } 6234 } 6235} 6236 6237void AudioFlinger::EffectModule::updateState() { 6238 Mutex::Autolock _l(mLock); 6239 6240 switch (mState) { 6241 case RESTART: 6242 reset_l(); 6243 // FALL THROUGH 6244 6245 case STARTING: 6246 // clear auxiliary effect input buffer for next accumulation 6247 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6248 memset(mConfig.inputCfg.buffer.raw, 6249 0, 6250 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6251 } 6252 start_l(); 6253 mState = ACTIVE; 6254 break; 6255 case STOPPING: 6256 stop_l(); 6257 mDisableWaitCnt = mMaxDisableWaitCnt; 6258 mState = STOPPED; 6259 break; 6260 case STOPPED: 6261 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6262 // turn off sequence. 6263 if (--mDisableWaitCnt == 0) { 6264 reset_l(); 6265 mState = IDLE; 6266 } 6267 break; 6268 default: //IDLE , ACTIVE, DESTROYED 6269 break; 6270 } 6271} 6272 6273void AudioFlinger::EffectModule::process() 6274{ 6275 Mutex::Autolock _l(mLock); 6276 6277 if (mState == DESTROYED || mEffectInterface == NULL || 6278 mConfig.inputCfg.buffer.raw == NULL || 6279 mConfig.outputCfg.buffer.raw == NULL) { 6280 return; 6281 } 6282 6283 if (isProcessEnabled()) { 6284 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6285 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6286 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6287 mConfig.inputCfg.buffer.s32, 6288 mConfig.inputCfg.buffer.frameCount/2); 6289 } 6290 6291 // do the actual processing in the effect engine 6292 int ret = (*mEffectInterface)->process(mEffectInterface, 6293 &mConfig.inputCfg.buffer, 6294 &mConfig.outputCfg.buffer); 6295 6296 // force transition to IDLE state when engine is ready 6297 if (mState == STOPPED && ret == -ENODATA) { 6298 mDisableWaitCnt = 1; 6299 } 6300 6301 // clear auxiliary effect input buffer for next accumulation 6302 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6303 memset(mConfig.inputCfg.buffer.raw, 0, 6304 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6305 } 6306 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6307 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6308 // If an insert effect is idle and input buffer is different from output buffer, 6309 // accumulate input onto output 6310 sp<EffectChain> chain = mChain.promote(); 6311 if (chain != 0 && chain->activeTrackCnt() != 0) { 6312 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6313 int16_t *in = mConfig.inputCfg.buffer.s16; 6314 int16_t *out = mConfig.outputCfg.buffer.s16; 6315 for (size_t i = 0; i < frameCnt; i++) { 6316 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6317 } 6318 } 6319 } 6320} 6321 6322void AudioFlinger::EffectModule::reset_l() 6323{ 6324 if (mEffectInterface == NULL) { 6325 return; 6326 } 6327 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6328} 6329 6330status_t AudioFlinger::EffectModule::configure() 6331{ 6332 uint32_t channels; 6333 if (mEffectInterface == NULL) { 6334 return NO_INIT; 6335 } 6336 6337 sp<ThreadBase> thread = mThread.promote(); 6338 if (thread == 0) { 6339 return DEAD_OBJECT; 6340 } 6341 6342 // TODO: handle configuration of effects replacing track process 6343 if (thread->channelCount() == 1) { 6344 channels = AUDIO_CHANNEL_OUT_MONO; 6345 } else { 6346 channels = AUDIO_CHANNEL_OUT_STEREO; 6347 } 6348 6349 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6350 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6351 } else { 6352 mConfig.inputCfg.channels = channels; 6353 } 6354 mConfig.outputCfg.channels = channels; 6355 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6356 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6357 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6358 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6359 mConfig.inputCfg.bufferProvider.cookie = NULL; 6360 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6361 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6362 mConfig.outputCfg.bufferProvider.cookie = NULL; 6363 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6364 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6365 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6366 // Insert effect: 6367 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6368 // always overwrites output buffer: input buffer == output buffer 6369 // - in other sessions: 6370 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6371 // other effect: overwrites output buffer: input buffer == output buffer 6372 // Auxiliary effect: 6373 // accumulates in output buffer: input buffer != output buffer 6374 // Therefore: accumulate <=> input buffer != output buffer 6375 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6376 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6377 } else { 6378 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6379 } 6380 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6381 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6382 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6383 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6384 6385 ALOGV("configure() %p thread %p buffer %p framecount %d", 6386 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6387 6388 status_t cmdStatus; 6389 uint32_t size = sizeof(int); 6390 status_t status = (*mEffectInterface)->command(mEffectInterface, 6391 EFFECT_CMD_SET_CONFIG, 6392 sizeof(effect_config_t), 6393 &mConfig, 6394 &size, 6395 &cmdStatus); 6396 if (status == 0) { 6397 status = cmdStatus; 6398 } 6399 6400 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6401 (1000 * mConfig.outputCfg.buffer.frameCount); 6402 6403 return status; 6404} 6405 6406status_t AudioFlinger::EffectModule::init() 6407{ 6408 Mutex::Autolock _l(mLock); 6409 if (mEffectInterface == NULL) { 6410 return NO_INIT; 6411 } 6412 status_t cmdStatus; 6413 uint32_t size = sizeof(status_t); 6414 status_t status = (*mEffectInterface)->command(mEffectInterface, 6415 EFFECT_CMD_INIT, 6416 0, 6417 NULL, 6418 &size, 6419 &cmdStatus); 6420 if (status == 0) { 6421 status = cmdStatus; 6422 } 6423 return status; 6424} 6425 6426status_t AudioFlinger::EffectModule::start() 6427{ 6428 Mutex::Autolock _l(mLock); 6429 return start_l(); 6430} 6431 6432status_t AudioFlinger::EffectModule::start_l() 6433{ 6434 if (mEffectInterface == NULL) { 6435 return NO_INIT; 6436 } 6437 status_t cmdStatus; 6438 uint32_t size = sizeof(status_t); 6439 status_t status = (*mEffectInterface)->command(mEffectInterface, 6440 EFFECT_CMD_ENABLE, 6441 0, 6442 NULL, 6443 &size, 6444 &cmdStatus); 6445 if (status == 0) { 6446 status = cmdStatus; 6447 } 6448 if (status == 0 && 6449 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6450 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6451 sp<ThreadBase> thread = mThread.promote(); 6452 if (thread != 0) { 6453 audio_stream_t *stream = thread->stream(); 6454 if (stream != NULL) { 6455 stream->add_audio_effect(stream, mEffectInterface); 6456 } 6457 } 6458 } 6459 return status; 6460} 6461 6462status_t AudioFlinger::EffectModule::stop() 6463{ 6464 Mutex::Autolock _l(mLock); 6465 return stop_l(); 6466} 6467 6468status_t AudioFlinger::EffectModule::stop_l() 6469{ 6470 if (mEffectInterface == NULL) { 6471 return NO_INIT; 6472 } 6473 status_t cmdStatus; 6474 uint32_t size = sizeof(status_t); 6475 status_t status = (*mEffectInterface)->command(mEffectInterface, 6476 EFFECT_CMD_DISABLE, 6477 0, 6478 NULL, 6479 &size, 6480 &cmdStatus); 6481 if (status == 0) { 6482 status = cmdStatus; 6483 } 6484 if (status == 0 && 6485 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6486 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6487 sp<ThreadBase> thread = mThread.promote(); 6488 if (thread != 0) { 6489 audio_stream_t *stream = thread->stream(); 6490 if (stream != NULL) { 6491 stream->remove_audio_effect(stream, mEffectInterface); 6492 } 6493 } 6494 } 6495 return status; 6496} 6497 6498status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6499 uint32_t cmdSize, 6500 void *pCmdData, 6501 uint32_t *replySize, 6502 void *pReplyData) 6503{ 6504 Mutex::Autolock _l(mLock); 6505// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6506 6507 if (mState == DESTROYED || mEffectInterface == NULL) { 6508 return NO_INIT; 6509 } 6510 status_t status = (*mEffectInterface)->command(mEffectInterface, 6511 cmdCode, 6512 cmdSize, 6513 pCmdData, 6514 replySize, 6515 pReplyData); 6516 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6517 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6518 for (size_t i = 1; i < mHandles.size(); i++) { 6519 sp<EffectHandle> h = mHandles[i].promote(); 6520 if (h != 0) { 6521 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6522 } 6523 } 6524 } 6525 return status; 6526} 6527 6528status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6529{ 6530 6531 Mutex::Autolock _l(mLock); 6532 ALOGV("setEnabled %p enabled %d", this, enabled); 6533 6534 if (enabled != isEnabled()) { 6535 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6536 if (enabled && status != NO_ERROR) { 6537 return status; 6538 } 6539 6540 switch (mState) { 6541 // going from disabled to enabled 6542 case IDLE: 6543 mState = STARTING; 6544 break; 6545 case STOPPED: 6546 mState = RESTART; 6547 break; 6548 case STOPPING: 6549 mState = ACTIVE; 6550 break; 6551 6552 // going from enabled to disabled 6553 case RESTART: 6554 mState = STOPPED; 6555 break; 6556 case STARTING: 6557 mState = IDLE; 6558 break; 6559 case ACTIVE: 6560 mState = STOPPING; 6561 break; 6562 case DESTROYED: 6563 return NO_ERROR; // simply ignore as we are being destroyed 6564 } 6565 for (size_t i = 1; i < mHandles.size(); i++) { 6566 sp<EffectHandle> h = mHandles[i].promote(); 6567 if (h != 0) { 6568 h->setEnabled(enabled); 6569 } 6570 } 6571 } 6572 return NO_ERROR; 6573} 6574 6575bool AudioFlinger::EffectModule::isEnabled() 6576{ 6577 switch (mState) { 6578 case RESTART: 6579 case STARTING: 6580 case ACTIVE: 6581 return true; 6582 case IDLE: 6583 case STOPPING: 6584 case STOPPED: 6585 case DESTROYED: 6586 default: 6587 return false; 6588 } 6589} 6590 6591bool AudioFlinger::EffectModule::isProcessEnabled() 6592{ 6593 switch (mState) { 6594 case RESTART: 6595 case ACTIVE: 6596 case STOPPING: 6597 case STOPPED: 6598 return true; 6599 case IDLE: 6600 case STARTING: 6601 case DESTROYED: 6602 default: 6603 return false; 6604 } 6605} 6606 6607status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6608{ 6609 Mutex::Autolock _l(mLock); 6610 status_t status = NO_ERROR; 6611 6612 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6613 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6614 if (isProcessEnabled() && 6615 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6616 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6617 status_t cmdStatus; 6618 uint32_t volume[2]; 6619 uint32_t *pVolume = NULL; 6620 uint32_t size = sizeof(volume); 6621 volume[0] = *left; 6622 volume[1] = *right; 6623 if (controller) { 6624 pVolume = volume; 6625 } 6626 status = (*mEffectInterface)->command(mEffectInterface, 6627 EFFECT_CMD_SET_VOLUME, 6628 size, 6629 volume, 6630 &size, 6631 pVolume); 6632 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6633 *left = volume[0]; 6634 *right = volume[1]; 6635 } 6636 } 6637 return status; 6638} 6639 6640status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6641{ 6642 Mutex::Autolock _l(mLock); 6643 status_t status = NO_ERROR; 6644 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6645 // audio pre processing modules on RecordThread can receive both output and 6646 // input device indication in the same call 6647 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6648 if (dev) { 6649 status_t cmdStatus; 6650 uint32_t size = sizeof(status_t); 6651 6652 status = (*mEffectInterface)->command(mEffectInterface, 6653 EFFECT_CMD_SET_DEVICE, 6654 sizeof(uint32_t), 6655 &dev, 6656 &size, 6657 &cmdStatus); 6658 if (status == NO_ERROR) { 6659 status = cmdStatus; 6660 } 6661 } 6662 dev = device & AUDIO_DEVICE_IN_ALL; 6663 if (dev) { 6664 status_t cmdStatus; 6665 uint32_t size = sizeof(status_t); 6666 6667 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6668 EFFECT_CMD_SET_INPUT_DEVICE, 6669 sizeof(uint32_t), 6670 &dev, 6671 &size, 6672 &cmdStatus); 6673 if (status2 == NO_ERROR) { 6674 status2 = cmdStatus; 6675 } 6676 if (status == NO_ERROR) { 6677 status = status2; 6678 } 6679 } 6680 } 6681 return status; 6682} 6683 6684status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6685{ 6686 Mutex::Autolock _l(mLock); 6687 status_t status = NO_ERROR; 6688 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6689 status_t cmdStatus; 6690 uint32_t size = sizeof(status_t); 6691 status = (*mEffectInterface)->command(mEffectInterface, 6692 EFFECT_CMD_SET_AUDIO_MODE, 6693 sizeof(audio_mode_t), 6694 &mode, 6695 &size, 6696 &cmdStatus); 6697 if (status == NO_ERROR) { 6698 status = cmdStatus; 6699 } 6700 } 6701 return status; 6702} 6703 6704void AudioFlinger::EffectModule::setSuspended(bool suspended) 6705{ 6706 Mutex::Autolock _l(mLock); 6707 mSuspended = suspended; 6708} 6709 6710bool AudioFlinger::EffectModule::suspended() const 6711{ 6712 Mutex::Autolock _l(mLock); 6713 return mSuspended; 6714} 6715 6716status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6717{ 6718 const size_t SIZE = 256; 6719 char buffer[SIZE]; 6720 String8 result; 6721 6722 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6723 result.append(buffer); 6724 6725 bool locked = tryLock(mLock); 6726 // failed to lock - AudioFlinger is probably deadlocked 6727 if (!locked) { 6728 result.append("\t\tCould not lock Fx mutex:\n"); 6729 } 6730 6731 result.append("\t\tSession Status State Engine:\n"); 6732 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6733 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6734 result.append(buffer); 6735 6736 result.append("\t\tDescriptor:\n"); 6737 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6738 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6739 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6740 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6741 result.append(buffer); 6742 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6743 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6744 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6745 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6746 result.append(buffer); 6747 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6748 mDescriptor.apiVersion, 6749 mDescriptor.flags); 6750 result.append(buffer); 6751 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6752 mDescriptor.name); 6753 result.append(buffer); 6754 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6755 mDescriptor.implementor); 6756 result.append(buffer); 6757 6758 result.append("\t\t- Input configuration:\n"); 6759 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6760 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6761 (uint32_t)mConfig.inputCfg.buffer.raw, 6762 mConfig.inputCfg.buffer.frameCount, 6763 mConfig.inputCfg.samplingRate, 6764 mConfig.inputCfg.channels, 6765 mConfig.inputCfg.format); 6766 result.append(buffer); 6767 6768 result.append("\t\t- Output configuration:\n"); 6769 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6770 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6771 (uint32_t)mConfig.outputCfg.buffer.raw, 6772 mConfig.outputCfg.buffer.frameCount, 6773 mConfig.outputCfg.samplingRate, 6774 mConfig.outputCfg.channels, 6775 mConfig.outputCfg.format); 6776 result.append(buffer); 6777 6778 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6779 result.append(buffer); 6780 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6781 for (size_t i = 0; i < mHandles.size(); ++i) { 6782 sp<EffectHandle> handle = mHandles[i].promote(); 6783 if (handle != 0) { 6784 handle->dump(buffer, SIZE); 6785 result.append(buffer); 6786 } 6787 } 6788 6789 result.append("\n"); 6790 6791 write(fd, result.string(), result.length()); 6792 6793 if (locked) { 6794 mLock.unlock(); 6795 } 6796 6797 return NO_ERROR; 6798} 6799 6800// ---------------------------------------------------------------------------- 6801// EffectHandle implementation 6802// ---------------------------------------------------------------------------- 6803 6804#undef LOG_TAG 6805#define LOG_TAG "AudioFlinger::EffectHandle" 6806 6807AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6808 const sp<AudioFlinger::Client>& client, 6809 const sp<IEffectClient>& effectClient, 6810 int32_t priority) 6811 : BnEffect(), 6812 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6813 mPriority(priority), mHasControl(false), mEnabled(false) 6814{ 6815 ALOGV("constructor %p", this); 6816 6817 if (client == 0) { 6818 return; 6819 } 6820 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6821 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6822 if (mCblkMemory != 0) { 6823 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6824 6825 if (mCblk) { 6826 new(mCblk) effect_param_cblk_t(); 6827 mBuffer = (uint8_t *)mCblk + bufOffset; 6828 } 6829 } else { 6830 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6831 return; 6832 } 6833} 6834 6835AudioFlinger::EffectHandle::~EffectHandle() 6836{ 6837 ALOGV("Destructor %p", this); 6838 disconnect(false); 6839 ALOGV("Destructor DONE %p", this); 6840} 6841 6842status_t AudioFlinger::EffectHandle::enable() 6843{ 6844 ALOGV("enable %p", this); 6845 if (!mHasControl) return INVALID_OPERATION; 6846 if (mEffect == 0) return DEAD_OBJECT; 6847 6848 if (mEnabled) { 6849 return NO_ERROR; 6850 } 6851 6852 mEnabled = true; 6853 6854 sp<ThreadBase> thread = mEffect->thread().promote(); 6855 if (thread != 0) { 6856 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6857 } 6858 6859 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6860 if (mEffect->suspended()) { 6861 return NO_ERROR; 6862 } 6863 6864 status_t status = mEffect->setEnabled(true); 6865 if (status != NO_ERROR) { 6866 if (thread != 0) { 6867 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6868 } 6869 mEnabled = false; 6870 } 6871 return status; 6872} 6873 6874status_t AudioFlinger::EffectHandle::disable() 6875{ 6876 ALOGV("disable %p", this); 6877 if (!mHasControl) return INVALID_OPERATION; 6878 if (mEffect == 0) return DEAD_OBJECT; 6879 6880 if (!mEnabled) { 6881 return NO_ERROR; 6882 } 6883 mEnabled = false; 6884 6885 if (mEffect->suspended()) { 6886 return NO_ERROR; 6887 } 6888 6889 status_t status = mEffect->setEnabled(false); 6890 6891 sp<ThreadBase> thread = mEffect->thread().promote(); 6892 if (thread != 0) { 6893 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6894 } 6895 6896 return status; 6897} 6898 6899void AudioFlinger::EffectHandle::disconnect() 6900{ 6901 disconnect(true); 6902} 6903 6904void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6905{ 6906 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6907 if (mEffect == 0) { 6908 return; 6909 } 6910 mEffect->disconnect(this, unpiniflast); 6911 6912 if (mHasControl && mEnabled) { 6913 sp<ThreadBase> thread = mEffect->thread().promote(); 6914 if (thread != 0) { 6915 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6916 } 6917 } 6918 6919 // release sp on module => module destructor can be called now 6920 mEffect.clear(); 6921 if (mClient != 0) { 6922 if (mCblk) { 6923 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6924 } 6925 mCblkMemory.clear(); // and free the shared memory 6926 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6927 mClient.clear(); 6928 } 6929} 6930 6931status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6932 uint32_t cmdSize, 6933 void *pCmdData, 6934 uint32_t *replySize, 6935 void *pReplyData) 6936{ 6937// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6938// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6939 6940 // only get parameter command is permitted for applications not controlling the effect 6941 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6942 return INVALID_OPERATION; 6943 } 6944 if (mEffect == 0) return DEAD_OBJECT; 6945 if (mClient == 0) return INVALID_OPERATION; 6946 6947 // handle commands that are not forwarded transparently to effect engine 6948 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6949 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6950 // no risk to block the whole media server process or mixer threads is we are stuck here 6951 Mutex::Autolock _l(mCblk->lock); 6952 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6953 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6954 mCblk->serverIndex = 0; 6955 mCblk->clientIndex = 0; 6956 return BAD_VALUE; 6957 } 6958 status_t status = NO_ERROR; 6959 while (mCblk->serverIndex < mCblk->clientIndex) { 6960 int reply; 6961 uint32_t rsize = sizeof(int); 6962 int *p = (int *)(mBuffer + mCblk->serverIndex); 6963 int size = *p++; 6964 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6965 ALOGW("command(): invalid parameter block size"); 6966 break; 6967 } 6968 effect_param_t *param = (effect_param_t *)p; 6969 if (param->psize == 0 || param->vsize == 0) { 6970 ALOGW("command(): null parameter or value size"); 6971 mCblk->serverIndex += size; 6972 continue; 6973 } 6974 uint32_t psize = sizeof(effect_param_t) + 6975 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6976 param->vsize; 6977 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6978 psize, 6979 p, 6980 &rsize, 6981 &reply); 6982 // stop at first error encountered 6983 if (ret != NO_ERROR) { 6984 status = ret; 6985 *(int *)pReplyData = reply; 6986 break; 6987 } else if (reply != NO_ERROR) { 6988 *(int *)pReplyData = reply; 6989 break; 6990 } 6991 mCblk->serverIndex += size; 6992 } 6993 mCblk->serverIndex = 0; 6994 mCblk->clientIndex = 0; 6995 return status; 6996 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6997 *(int *)pReplyData = NO_ERROR; 6998 return enable(); 6999 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7000 *(int *)pReplyData = NO_ERROR; 7001 return disable(); 7002 } 7003 7004 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7005} 7006 7007sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7008 return mCblkMemory; 7009} 7010 7011void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7012{ 7013 ALOGV("setControl %p control %d", this, hasControl); 7014 7015 mHasControl = hasControl; 7016 mEnabled = enabled; 7017 7018 if (signal && mEffectClient != 0) { 7019 mEffectClient->controlStatusChanged(hasControl); 7020 } 7021} 7022 7023void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7024 uint32_t cmdSize, 7025 void *pCmdData, 7026 uint32_t replySize, 7027 void *pReplyData) 7028{ 7029 if (mEffectClient != 0) { 7030 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7031 } 7032} 7033 7034 7035 7036void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7037{ 7038 if (mEffectClient != 0) { 7039 mEffectClient->enableStatusChanged(enabled); 7040 } 7041} 7042 7043status_t AudioFlinger::EffectHandle::onTransact( 7044 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7045{ 7046 return BnEffect::onTransact(code, data, reply, flags); 7047} 7048 7049 7050void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7051{ 7052 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7053 7054 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7055 (mClient == NULL) ? getpid() : mClient->pid(), 7056 mPriority, 7057 mHasControl, 7058 !locked, 7059 mCblk ? mCblk->clientIndex : 0, 7060 mCblk ? mCblk->serverIndex : 0 7061 ); 7062 7063 if (locked) { 7064 mCblk->lock.unlock(); 7065 } 7066} 7067 7068#undef LOG_TAG 7069#define LOG_TAG "AudioFlinger::EffectChain" 7070 7071AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7072 int sessionId) 7073 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7074 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7075 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7076{ 7077 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7078 sp<ThreadBase> thread = mThread.promote(); 7079 if (thread == 0) { 7080 return; 7081 } 7082 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7083 thread->frameCount(); 7084} 7085 7086AudioFlinger::EffectChain::~EffectChain() 7087{ 7088 if (mOwnInBuffer) { 7089 delete mInBuffer; 7090 } 7091 7092} 7093 7094// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7095sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7096{ 7097 sp<EffectModule> effect; 7098 size_t size = mEffects.size(); 7099 7100 for (size_t i = 0; i < size; i++) { 7101 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7102 effect = mEffects[i]; 7103 break; 7104 } 7105 } 7106 return effect; 7107} 7108 7109// getEffectFromId_l() must be called with ThreadBase::mLock held 7110sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7111{ 7112 sp<EffectModule> effect; 7113 size_t size = mEffects.size(); 7114 7115 for (size_t i = 0; i < size; i++) { 7116 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7117 if (id == 0 || mEffects[i]->id() == id) { 7118 effect = mEffects[i]; 7119 break; 7120 } 7121 } 7122 return effect; 7123} 7124 7125// getEffectFromType_l() must be called with ThreadBase::mLock held 7126sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7127 const effect_uuid_t *type) 7128{ 7129 sp<EffectModule> effect; 7130 size_t size = mEffects.size(); 7131 7132 for (size_t i = 0; i < size; i++) { 7133 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7134 effect = mEffects[i]; 7135 break; 7136 } 7137 } 7138 return effect; 7139} 7140 7141// Must be called with EffectChain::mLock locked 7142void AudioFlinger::EffectChain::process_l() 7143{ 7144 sp<ThreadBase> thread = mThread.promote(); 7145 if (thread == 0) { 7146 ALOGW("process_l(): cannot promote mixer thread"); 7147 return; 7148 } 7149 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7150 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7151 // always process effects unless no more tracks are on the session and the effect tail 7152 // has been rendered 7153 bool doProcess = true; 7154 if (!isGlobalSession) { 7155 bool tracksOnSession = (trackCnt() != 0); 7156 7157 if (!tracksOnSession && mTailBufferCount == 0) { 7158 doProcess = false; 7159 } 7160 7161 if (activeTrackCnt() == 0) { 7162 // if no track is active and the effect tail has not been rendered, 7163 // the input buffer must be cleared here as the mixer process will not do it 7164 if (tracksOnSession || mTailBufferCount > 0) { 7165 size_t numSamples = thread->frameCount() * thread->channelCount(); 7166 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7167 if (mTailBufferCount > 0) { 7168 mTailBufferCount--; 7169 } 7170 } 7171 } 7172 } 7173 7174 size_t size = mEffects.size(); 7175 if (doProcess) { 7176 for (size_t i = 0; i < size; i++) { 7177 mEffects[i]->process(); 7178 } 7179 } 7180 for (size_t i = 0; i < size; i++) { 7181 mEffects[i]->updateState(); 7182 } 7183} 7184 7185// addEffect_l() must be called with PlaybackThread::mLock held 7186status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7187{ 7188 effect_descriptor_t desc = effect->desc(); 7189 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7190 7191 Mutex::Autolock _l(mLock); 7192 effect->setChain(this); 7193 sp<ThreadBase> thread = mThread.promote(); 7194 if (thread == 0) { 7195 return NO_INIT; 7196 } 7197 effect->setThread(thread); 7198 7199 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7200 // Auxiliary effects are inserted at the beginning of mEffects vector as 7201 // they are processed first and accumulated in chain input buffer 7202 mEffects.insertAt(effect, 0); 7203 7204 // the input buffer for auxiliary effect contains mono samples in 7205 // 32 bit format. This is to avoid saturation in AudoMixer 7206 // accumulation stage. Saturation is done in EffectModule::process() before 7207 // calling the process in effect engine 7208 size_t numSamples = thread->frameCount(); 7209 int32_t *buffer = new int32_t[numSamples]; 7210 memset(buffer, 0, numSamples * sizeof(int32_t)); 7211 effect->setInBuffer((int16_t *)buffer); 7212 // auxiliary effects output samples to chain input buffer for further processing 7213 // by insert effects 7214 effect->setOutBuffer(mInBuffer); 7215 } else { 7216 // Insert effects are inserted at the end of mEffects vector as they are processed 7217 // after track and auxiliary effects. 7218 // Insert effect order as a function of indicated preference: 7219 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7220 // another effect is present 7221 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7222 // last effect claiming first position 7223 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7224 // first effect claiming last position 7225 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7226 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7227 // already present 7228 7229 int size = (int)mEffects.size(); 7230 int idx_insert = size; 7231 int idx_insert_first = -1; 7232 int idx_insert_last = -1; 7233 7234 for (int i = 0; i < size; i++) { 7235 effect_descriptor_t d = mEffects[i]->desc(); 7236 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7237 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7238 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7239 // check invalid effect chaining combinations 7240 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7241 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7242 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7243 return INVALID_OPERATION; 7244 } 7245 // remember position of first insert effect and by default 7246 // select this as insert position for new effect 7247 if (idx_insert == size) { 7248 idx_insert = i; 7249 } 7250 // remember position of last insert effect claiming 7251 // first position 7252 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7253 idx_insert_first = i; 7254 } 7255 // remember position of first insert effect claiming 7256 // last position 7257 if (iPref == EFFECT_FLAG_INSERT_LAST && 7258 idx_insert_last == -1) { 7259 idx_insert_last = i; 7260 } 7261 } 7262 } 7263 7264 // modify idx_insert from first position if needed 7265 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7266 if (idx_insert_last != -1) { 7267 idx_insert = idx_insert_last; 7268 } else { 7269 idx_insert = size; 7270 } 7271 } else { 7272 if (idx_insert_first != -1) { 7273 idx_insert = idx_insert_first + 1; 7274 } 7275 } 7276 7277 // always read samples from chain input buffer 7278 effect->setInBuffer(mInBuffer); 7279 7280 // if last effect in the chain, output samples to chain 7281 // output buffer, otherwise to chain input buffer 7282 if (idx_insert == size) { 7283 if (idx_insert != 0) { 7284 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7285 mEffects[idx_insert-1]->configure(); 7286 } 7287 effect->setOutBuffer(mOutBuffer); 7288 } else { 7289 effect->setOutBuffer(mInBuffer); 7290 } 7291 mEffects.insertAt(effect, idx_insert); 7292 7293 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7294 } 7295 effect->configure(); 7296 return NO_ERROR; 7297} 7298 7299// removeEffect_l() must be called with PlaybackThread::mLock held 7300size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7301{ 7302 Mutex::Autolock _l(mLock); 7303 int size = (int)mEffects.size(); 7304 int i; 7305 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7306 7307 for (i = 0; i < size; i++) { 7308 if (effect == mEffects[i]) { 7309 // calling stop here will remove pre-processing effect from the audio HAL. 7310 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7311 // the middle of a read from audio HAL 7312 if (mEffects[i]->state() == EffectModule::ACTIVE || 7313 mEffects[i]->state() == EffectModule::STOPPING) { 7314 mEffects[i]->stop(); 7315 } 7316 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7317 delete[] effect->inBuffer(); 7318 } else { 7319 if (i == size - 1 && i != 0) { 7320 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7321 mEffects[i - 1]->configure(); 7322 } 7323 } 7324 mEffects.removeAt(i); 7325 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7326 break; 7327 } 7328 } 7329 7330 return mEffects.size(); 7331} 7332 7333// setDevice_l() must be called with PlaybackThread::mLock held 7334void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7335{ 7336 size_t size = mEffects.size(); 7337 for (size_t i = 0; i < size; i++) { 7338 mEffects[i]->setDevice(device); 7339 } 7340} 7341 7342// setMode_l() must be called with PlaybackThread::mLock held 7343void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7344{ 7345 size_t size = mEffects.size(); 7346 for (size_t i = 0; i < size; i++) { 7347 mEffects[i]->setMode(mode); 7348 } 7349} 7350 7351// setVolume_l() must be called with PlaybackThread::mLock held 7352bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7353{ 7354 uint32_t newLeft = *left; 7355 uint32_t newRight = *right; 7356 bool hasControl = false; 7357 int ctrlIdx = -1; 7358 size_t size = mEffects.size(); 7359 7360 // first update volume controller 7361 for (size_t i = size; i > 0; i--) { 7362 if (mEffects[i - 1]->isProcessEnabled() && 7363 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7364 ctrlIdx = i - 1; 7365 hasControl = true; 7366 break; 7367 } 7368 } 7369 7370 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7371 if (hasControl) { 7372 *left = mNewLeftVolume; 7373 *right = mNewRightVolume; 7374 } 7375 return hasControl; 7376 } 7377 7378 mVolumeCtrlIdx = ctrlIdx; 7379 mLeftVolume = newLeft; 7380 mRightVolume = newRight; 7381 7382 // second get volume update from volume controller 7383 if (ctrlIdx >= 0) { 7384 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7385 mNewLeftVolume = newLeft; 7386 mNewRightVolume = newRight; 7387 } 7388 // then indicate volume to all other effects in chain. 7389 // Pass altered volume to effects before volume controller 7390 // and requested volume to effects after controller 7391 uint32_t lVol = newLeft; 7392 uint32_t rVol = newRight; 7393 7394 for (size_t i = 0; i < size; i++) { 7395 if ((int)i == ctrlIdx) continue; 7396 // this also works for ctrlIdx == -1 when there is no volume controller 7397 if ((int)i > ctrlIdx) { 7398 lVol = *left; 7399 rVol = *right; 7400 } 7401 mEffects[i]->setVolume(&lVol, &rVol, false); 7402 } 7403 *left = newLeft; 7404 *right = newRight; 7405 7406 return hasControl; 7407} 7408 7409status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7410{ 7411 const size_t SIZE = 256; 7412 char buffer[SIZE]; 7413 String8 result; 7414 7415 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7416 result.append(buffer); 7417 7418 bool locked = tryLock(mLock); 7419 // failed to lock - AudioFlinger is probably deadlocked 7420 if (!locked) { 7421 result.append("\tCould not lock mutex:\n"); 7422 } 7423 7424 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7425 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7426 mEffects.size(), 7427 (uint32_t)mInBuffer, 7428 (uint32_t)mOutBuffer, 7429 mActiveTrackCnt); 7430 result.append(buffer); 7431 write(fd, result.string(), result.size()); 7432 7433 for (size_t i = 0; i < mEffects.size(); ++i) { 7434 sp<EffectModule> effect = mEffects[i]; 7435 if (effect != 0) { 7436 effect->dump(fd, args); 7437 } 7438 } 7439 7440 if (locked) { 7441 mLock.unlock(); 7442 } 7443 7444 return NO_ERROR; 7445} 7446 7447// must be called with ThreadBase::mLock held 7448void AudioFlinger::EffectChain::setEffectSuspended_l( 7449 const effect_uuid_t *type, bool suspend) 7450{ 7451 sp<SuspendedEffectDesc> desc; 7452 // use effect type UUID timelow as key as there is no real risk of identical 7453 // timeLow fields among effect type UUIDs. 7454 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7455 if (suspend) { 7456 if (index >= 0) { 7457 desc = mSuspendedEffects.valueAt(index); 7458 } else { 7459 desc = new SuspendedEffectDesc(); 7460 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7461 mSuspendedEffects.add(type->timeLow, desc); 7462 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7463 } 7464 if (desc->mRefCount++ == 0) { 7465 sp<EffectModule> effect = getEffectIfEnabled(type); 7466 if (effect != 0) { 7467 desc->mEffect = effect; 7468 effect->setSuspended(true); 7469 effect->setEnabled(false); 7470 } 7471 } 7472 } else { 7473 if (index < 0) { 7474 return; 7475 } 7476 desc = mSuspendedEffects.valueAt(index); 7477 if (desc->mRefCount <= 0) { 7478 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7479 desc->mRefCount = 1; 7480 } 7481 if (--desc->mRefCount == 0) { 7482 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7483 if (desc->mEffect != 0) { 7484 sp<EffectModule> effect = desc->mEffect.promote(); 7485 if (effect != 0) { 7486 effect->setSuspended(false); 7487 sp<EffectHandle> handle = effect->controlHandle(); 7488 if (handle != 0) { 7489 effect->setEnabled(handle->enabled()); 7490 } 7491 } 7492 desc->mEffect.clear(); 7493 } 7494 mSuspendedEffects.removeItemsAt(index); 7495 } 7496 } 7497} 7498 7499// must be called with ThreadBase::mLock held 7500void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7501{ 7502 sp<SuspendedEffectDesc> desc; 7503 7504 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7505 if (suspend) { 7506 if (index >= 0) { 7507 desc = mSuspendedEffects.valueAt(index); 7508 } else { 7509 desc = new SuspendedEffectDesc(); 7510 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7511 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7512 } 7513 if (desc->mRefCount++ == 0) { 7514 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7515 for (size_t i = 0; i < effects.size(); i++) { 7516 setEffectSuspended_l(&effects[i]->desc().type, true); 7517 } 7518 } 7519 } else { 7520 if (index < 0) { 7521 return; 7522 } 7523 desc = mSuspendedEffects.valueAt(index); 7524 if (desc->mRefCount <= 0) { 7525 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7526 desc->mRefCount = 1; 7527 } 7528 if (--desc->mRefCount == 0) { 7529 Vector<const effect_uuid_t *> types; 7530 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7531 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7532 continue; 7533 } 7534 types.add(&mSuspendedEffects.valueAt(i)->mType); 7535 } 7536 for (size_t i = 0; i < types.size(); i++) { 7537 setEffectSuspended_l(types[i], false); 7538 } 7539 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7540 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7541 } 7542 } 7543} 7544 7545 7546// The volume effect is used for automated tests only 7547#ifndef OPENSL_ES_H_ 7548static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7549 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7550const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7551#endif //OPENSL_ES_H_ 7552 7553bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7554{ 7555 // auxiliary effects and visualizer are never suspended on output mix 7556 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7557 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7558 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7559 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7560 return false; 7561 } 7562 return true; 7563} 7564 7565Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7566{ 7567 Vector< sp<EffectModule> > effects; 7568 for (size_t i = 0; i < mEffects.size(); i++) { 7569 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7570 continue; 7571 } 7572 effects.add(mEffects[i]); 7573 } 7574 return effects; 7575} 7576 7577sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7578 const effect_uuid_t *type) 7579{ 7580 sp<EffectModule> effect; 7581 effect = getEffectFromType_l(type); 7582 if (effect != 0 && !effect->isEnabled()) { 7583 effect.clear(); 7584 } 7585 return effect; 7586} 7587 7588void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7589 bool enabled) 7590{ 7591 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7592 if (enabled) { 7593 if (index < 0) { 7594 // if the effect is not suspend check if all effects are suspended 7595 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7596 if (index < 0) { 7597 return; 7598 } 7599 if (!isEffectEligibleForSuspend(effect->desc())) { 7600 return; 7601 } 7602 setEffectSuspended_l(&effect->desc().type, enabled); 7603 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7604 if (index < 0) { 7605 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7606 return; 7607 } 7608 } 7609 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7610 effect->desc().type.timeLow); 7611 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7612 // if effect is requested to suspended but was not yet enabled, supend it now. 7613 if (desc->mEffect == 0) { 7614 desc->mEffect = effect; 7615 effect->setEnabled(false); 7616 effect->setSuspended(true); 7617 } 7618 } else { 7619 if (index < 0) { 7620 return; 7621 } 7622 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7623 effect->desc().type.timeLow); 7624 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7625 desc->mEffect.clear(); 7626 effect->setSuspended(false); 7627 } 7628} 7629 7630#undef LOG_TAG 7631#define LOG_TAG "AudioFlinger" 7632 7633// ---------------------------------------------------------------------------- 7634 7635status_t AudioFlinger::onTransact( 7636 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7637{ 7638 return BnAudioFlinger::onTransact(code, data, reply, flags); 7639} 7640 7641}; // namespace android 7642