AudioFlinger.cpp revision 335787fe43596f38ea2fa50b24c54d0823a3fb1d
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <audio_effects/effect_visualizer.h> 54#include <audio_effects/effect_ns.h> 55#include <audio_effects/effect_aec.h> 56 57#include <audio_utils/primitives.h> 58 59#include <cpustats/ThreadCpuUsage.h> 60#include <powermanager/PowerManager.h> 61// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 62 63// ---------------------------------------------------------------------------- 64 65 66namespace android { 67 68static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 69static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 70 71//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 72static const float MAX_GAIN = 4096.0f; 73static const uint32_t MAX_GAIN_INT = 0x1000; 74 75// retry counts for buffer fill timeout 76// 50 * ~20msecs = 1 second 77static const int8_t kMaxTrackRetries = 50; 78static const int8_t kMaxTrackStartupRetries = 50; 79// allow less retry attempts on direct output thread. 80// direct outputs can be a scarce resource in audio hardware and should 81// be released as quickly as possible. 82static const int8_t kMaxTrackRetriesDirect = 2; 83 84static const int kDumpLockRetries = 50; 85static const int kDumpLockSleepUs = 20000; 86 87// don't warn about blocked writes or record buffer overflows more often than this 88static const nsecs_t kWarningThrottleNs = seconds(5); 89 90// RecordThread loop sleep time upon application overrun or audio HAL read error 91static const int kRecordThreadSleepUs = 5000; 92 93// maximum time to wait for setParameters to complete 94static const nsecs_t kSetParametersTimeoutNs = seconds(2); 95 96// minimum sleep time for the mixer thread loop when tracks are active but in underrun 97static const uint32_t kMinThreadSleepTimeUs = 5000; 98// maximum divider applied to the active sleep time in the mixer thread loop 99static const uint32_t kMaxThreadSleepTimeShift = 2; 100 101 102// ---------------------------------------------------------------------------- 103 104static bool recordingAllowed() { 105 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 106 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 107 if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO"); 108 return ok; 109} 110 111static bool settingsAllowed() { 112 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 113 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 114 if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 115 return ok; 116} 117 118// To collect the amplifier usage 119static void addBatteryData(uint32_t params) { 120 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 121 if (service == NULL) { 122 // it already logged 123 return; 124 } 125 126 service->addBatteryData(params); 127} 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 164 mBtNrecIsOff(false) 165{ 166} 167 168void AudioFlinger::onFirstRef() 169{ 170 int rc = 0; 171 172 Mutex::Autolock _l(mLock); 173 174 /* TODO: move all this work into an Init() function */ 175 mHardwareStatus = AUDIO_HW_IDLE; 176 177 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 178 const hw_module_t *mod; 179 audio_hw_device_t *dev; 180 181 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 182 if (rc) 183 continue; 184 185 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 186 mod->name, mod->id); 187 mAudioHwDevs.push(dev); 188 189 if (!mPrimaryHardwareDev) { 190 mPrimaryHardwareDev = dev; 191 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 192 mod->name, mod->id, audio_interfaces[i]); 193 } 194 } 195 196 mHardwareStatus = AUDIO_HW_INIT; 197 198 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 199 ALOGE("Primary audio interface not found"); 200 return; 201 } 202 203 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 204 audio_hw_device_t *dev = mAudioHwDevs[i]; 205 206 mHardwareStatus = AUDIO_HW_INIT; 207 rc = dev->init_check(dev); 208 if (rc == 0) { 209 AutoMutex lock(mHardwareLock); 210 211 mMode = AUDIO_MODE_NORMAL; 212 mHardwareStatus = AUDIO_HW_SET_MODE; 213 dev->set_mode(dev, mMode); 214 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 215 dev->set_master_volume(dev, 1.0f); 216 mHardwareStatus = AUDIO_HW_IDLE; 217 } 218 } 219} 220 221status_t AudioFlinger::initCheck() const 222{ 223 Mutex::Autolock _l(mLock); 224 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 225 return NO_INIT; 226 return NO_ERROR; 227} 228 229AudioFlinger::~AudioFlinger() 230{ 231 int num_devs = mAudioHwDevs.size(); 232 233 while (!mRecordThreads.isEmpty()) { 234 // closeInput() will remove first entry from mRecordThreads 235 closeInput(mRecordThreads.keyAt(0)); 236 } 237 while (!mPlaybackThreads.isEmpty()) { 238 // closeOutput() will remove first entry from mPlaybackThreads 239 closeOutput(mPlaybackThreads.keyAt(0)); 240 } 241 242 for (int i = 0; i < num_devs; i++) { 243 audio_hw_device_t *dev = mAudioHwDevs[i]; 244 audio_hw_device_close(dev); 245 } 246 mAudioHwDevs.clear(); 247} 248 249audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 250{ 251 /* first matching HW device is returned */ 252 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 253 audio_hw_device_t *dev = mAudioHwDevs[i]; 254 if ((dev->get_supported_devices(dev) & devices) == devices) 255 return dev; 256 } 257 return NULL; 258} 259 260status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 261{ 262 const size_t SIZE = 256; 263 char buffer[SIZE]; 264 String8 result; 265 266 result.append("Clients:\n"); 267 for (size_t i = 0; i < mClients.size(); ++i) { 268 wp<Client> wClient = mClients.valueAt(i); 269 if (wClient != 0) { 270 sp<Client> client = wClient.promote(); 271 if (client != 0) { 272 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 273 result.append(buffer); 274 } 275 } 276 } 277 278 result.append("Global session refs:\n"); 279 result.append(" session pid cnt\n"); 280 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 281 AudioSessionRef *r = mAudioSessionRefs[i]; 282 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 283 result.append(buffer); 284 } 285 write(fd, result.string(), result.size()); 286 return NO_ERROR; 287} 288 289 290status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 291{ 292 const size_t SIZE = 256; 293 char buffer[SIZE]; 294 String8 result; 295 hardware_call_state hardwareStatus = mHardwareStatus; 296 297 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 298 result.append(buffer); 299 write(fd, result.string(), result.size()); 300 return NO_ERROR; 301} 302 303status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 304{ 305 const size_t SIZE = 256; 306 char buffer[SIZE]; 307 String8 result; 308 snprintf(buffer, SIZE, "Permission Denial: " 309 "can't dump AudioFlinger from pid=%d, uid=%d\n", 310 IPCThreadState::self()->getCallingPid(), 311 IPCThreadState::self()->getCallingUid()); 312 result.append(buffer); 313 write(fd, result.string(), result.size()); 314 return NO_ERROR; 315} 316 317static bool tryLock(Mutex& mutex) 318{ 319 bool locked = false; 320 for (int i = 0; i < kDumpLockRetries; ++i) { 321 if (mutex.tryLock() == NO_ERROR) { 322 locked = true; 323 break; 324 } 325 usleep(kDumpLockSleepUs); 326 } 327 return locked; 328} 329 330status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 331{ 332 if (!checkCallingPermission(String16("android.permission.DUMP"))) { 333 dumpPermissionDenial(fd, args); 334 } else { 335 // get state of hardware lock 336 bool hardwareLocked = tryLock(mHardwareLock); 337 if (!hardwareLocked) { 338 String8 result(kHardwareLockedString); 339 write(fd, result.string(), result.size()); 340 } else { 341 mHardwareLock.unlock(); 342 } 343 344 bool locked = tryLock(mLock); 345 346 // failed to lock - AudioFlinger is probably deadlocked 347 if (!locked) { 348 String8 result(kDeadlockedString); 349 write(fd, result.string(), result.size()); 350 } 351 352 dumpClients(fd, args); 353 dumpInternals(fd, args); 354 355 // dump playback threads 356 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 357 mPlaybackThreads.valueAt(i)->dump(fd, args); 358 } 359 360 // dump record threads 361 for (size_t i = 0; i < mRecordThreads.size(); i++) { 362 mRecordThreads.valueAt(i)->dump(fd, args); 363 } 364 365 // dump all hardware devs 366 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 367 audio_hw_device_t *dev = mAudioHwDevs[i]; 368 dev->dump(dev, fd); 369 } 370 if (locked) mLock.unlock(); 371 } 372 return NO_ERROR; 373} 374 375 376// IAudioFlinger interface 377 378 379sp<IAudioTrack> AudioFlinger::createTrack( 380 pid_t pid, 381 audio_stream_type_t streamType, 382 uint32_t sampleRate, 383 audio_format_t format, 384 uint32_t channelMask, 385 int frameCount, 386 uint32_t flags, 387 const sp<IMemory>& sharedBuffer, 388 int output, 389 int *sessionId, 390 status_t *status) 391{ 392 sp<PlaybackThread::Track> track; 393 sp<TrackHandle> trackHandle; 394 sp<Client> client; 395 wp<Client> wclient; 396 status_t lStatus; 397 int lSessionId; 398 399 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 400 // but if someone uses binder directly they could bypass that and cause us to crash 401 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 402 ALOGE("createTrack() invalid stream type %d", streamType); 403 lStatus = BAD_VALUE; 404 goto Exit; 405 } 406 407 { 408 Mutex::Autolock _l(mLock); 409 PlaybackThread *thread = checkPlaybackThread_l(output); 410 PlaybackThread *effectThread = NULL; 411 if (thread == NULL) { 412 ALOGE("unknown output thread"); 413 lStatus = BAD_VALUE; 414 goto Exit; 415 } 416 417 wclient = mClients.valueFor(pid); 418 419 if (wclient != NULL) { 420 client = wclient.promote(); 421 } else { 422 client = new Client(this, pid); 423 mClients.add(pid, client); 424 } 425 426 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 427 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 428 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 429 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 430 if (mPlaybackThreads.keyAt(i) != output) { 431 // prevent same audio session on different output threads 432 uint32_t sessions = t->hasAudioSession(*sessionId); 433 if (sessions & PlaybackThread::TRACK_SESSION) { 434 ALOGE("createTrack() session ID %d already in use", *sessionId); 435 lStatus = BAD_VALUE; 436 goto Exit; 437 } 438 // check if an effect with same session ID is waiting for a track to be created 439 if (sessions & PlaybackThread::EFFECT_SESSION) { 440 effectThread = t.get(); 441 } 442 } 443 } 444 lSessionId = *sessionId; 445 } else { 446 // if no audio session id is provided, create one here 447 lSessionId = nextUniqueId(); 448 if (sessionId != NULL) { 449 *sessionId = lSessionId; 450 } 451 } 452 ALOGV("createTrack() lSessionId: %d", lSessionId); 453 454 track = thread->createTrack_l(client, streamType, sampleRate, format, 455 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 456 457 // move effect chain to this output thread if an effect on same session was waiting 458 // for a track to be created 459 if (lStatus == NO_ERROR && effectThread != NULL) { 460 Mutex::Autolock _dl(thread->mLock); 461 Mutex::Autolock _sl(effectThread->mLock); 462 moveEffectChain_l(lSessionId, effectThread, thread, true); 463 } 464 } 465 if (lStatus == NO_ERROR) { 466 trackHandle = new TrackHandle(track); 467 } else { 468 // remove local strong reference to Client before deleting the Track so that the Client 469 // destructor is called by the TrackBase destructor with mLock held 470 client.clear(); 471 track.clear(); 472 } 473 474Exit: 475 if(status) { 476 *status = lStatus; 477 } 478 return trackHandle; 479} 480 481uint32_t AudioFlinger::sampleRate(int output) const 482{ 483 Mutex::Autolock _l(mLock); 484 PlaybackThread *thread = checkPlaybackThread_l(output); 485 if (thread == NULL) { 486 ALOGW("sampleRate() unknown thread %d", output); 487 return 0; 488 } 489 return thread->sampleRate(); 490} 491 492int AudioFlinger::channelCount(int output) const 493{ 494 Mutex::Autolock _l(mLock); 495 PlaybackThread *thread = checkPlaybackThread_l(output); 496 if (thread == NULL) { 497 ALOGW("channelCount() unknown thread %d", output); 498 return 0; 499 } 500 return thread->channelCount(); 501} 502 503audio_format_t AudioFlinger::format(int output) const 504{ 505 Mutex::Autolock _l(mLock); 506 PlaybackThread *thread = checkPlaybackThread_l(output); 507 if (thread == NULL) { 508 ALOGW("format() unknown thread %d", output); 509 return AUDIO_FORMAT_INVALID; 510 } 511 return thread->format(); 512} 513 514size_t AudioFlinger::frameCount(int output) const 515{ 516 Mutex::Autolock _l(mLock); 517 PlaybackThread *thread = checkPlaybackThread_l(output); 518 if (thread == NULL) { 519 ALOGW("frameCount() unknown thread %d", output); 520 return 0; 521 } 522 return thread->frameCount(); 523} 524 525uint32_t AudioFlinger::latency(int output) const 526{ 527 Mutex::Autolock _l(mLock); 528 PlaybackThread *thread = checkPlaybackThread_l(output); 529 if (thread == NULL) { 530 ALOGW("latency() unknown thread %d", output); 531 return 0; 532 } 533 return thread->latency(); 534} 535 536status_t AudioFlinger::setMasterVolume(float value) 537{ 538 status_t ret = initCheck(); 539 if (ret != NO_ERROR) { 540 return ret; 541 } 542 543 // check calling permissions 544 if (!settingsAllowed()) { 545 return PERMISSION_DENIED; 546 } 547 548 // when hw supports master volume, don't scale in sw mixer 549 { // scope for the lock 550 AutoMutex lock(mHardwareLock); 551 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 552 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 553 value = 1.0f; 554 } 555 mHardwareStatus = AUDIO_HW_IDLE; 556 } 557 558 Mutex::Autolock _l(mLock); 559 mMasterVolume = value; 560 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 561 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 562 563 return NO_ERROR; 564} 565 566status_t AudioFlinger::setMode(audio_mode_t mode) 567{ 568 status_t ret = initCheck(); 569 if (ret != NO_ERROR) { 570 return ret; 571 } 572 573 // check calling permissions 574 if (!settingsAllowed()) { 575 return PERMISSION_DENIED; 576 } 577 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 578 ALOGW("Illegal value: setMode(%d)", mode); 579 return BAD_VALUE; 580 } 581 582 { // scope for the lock 583 AutoMutex lock(mHardwareLock); 584 mHardwareStatus = AUDIO_HW_SET_MODE; 585 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 586 mHardwareStatus = AUDIO_HW_IDLE; 587 } 588 589 if (NO_ERROR == ret) { 590 Mutex::Autolock _l(mLock); 591 mMode = mode; 592 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 593 mPlaybackThreads.valueAt(i)->setMode(mode); 594 } 595 596 return ret; 597} 598 599status_t AudioFlinger::setMicMute(bool state) 600{ 601 status_t ret = initCheck(); 602 if (ret != NO_ERROR) { 603 return ret; 604 } 605 606 // check calling permissions 607 if (!settingsAllowed()) { 608 return PERMISSION_DENIED; 609 } 610 611 AutoMutex lock(mHardwareLock); 612 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 613 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 614 mHardwareStatus = AUDIO_HW_IDLE; 615 return ret; 616} 617 618bool AudioFlinger::getMicMute() const 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return false; 623 } 624 625 bool state = AUDIO_MODE_INVALID; 626 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 627 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 628 mHardwareStatus = AUDIO_HW_IDLE; 629 return state; 630} 631 632status_t AudioFlinger::setMasterMute(bool muted) 633{ 634 // check calling permissions 635 if (!settingsAllowed()) { 636 return PERMISSION_DENIED; 637 } 638 639 Mutex::Autolock _l(mLock); 640 mMasterMute = muted; 641 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 642 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 643 644 return NO_ERROR; 645} 646 647float AudioFlinger::masterVolume() const 648{ 649 Mutex::Autolock _l(mLock); 650 return masterVolume_l(); 651} 652 653bool AudioFlinger::masterMute() const 654{ 655 Mutex::Autolock _l(mLock); 656 return masterMute_l(); 657} 658 659status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output) 660{ 661 // check calling permissions 662 if (!settingsAllowed()) { 663 return PERMISSION_DENIED; 664 } 665 666 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 667 ALOGE("setStreamVolume() invalid stream %d", stream); 668 return BAD_VALUE; 669 } 670 671 AutoMutex lock(mLock); 672 PlaybackThread *thread = NULL; 673 if (output) { 674 thread = checkPlaybackThread_l(output); 675 if (thread == NULL) { 676 return BAD_VALUE; 677 } 678 } 679 680 mStreamTypes[stream].volume = value; 681 682 if (thread == NULL) { 683 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 684 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 685 } 686 } else { 687 thread->setStreamVolume(stream, value); 688 } 689 690 return NO_ERROR; 691} 692 693status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 694{ 695 // check calling permissions 696 if (!settingsAllowed()) { 697 return PERMISSION_DENIED; 698 } 699 700 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 701 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 702 ALOGE("setStreamMute() invalid stream %d", stream); 703 return BAD_VALUE; 704 } 705 706 AutoMutex lock(mLock); 707 mStreamTypes[stream].mute = muted; 708 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 709 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 710 711 return NO_ERROR; 712} 713 714float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const 715{ 716 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 717 return 0.0f; 718 } 719 720 AutoMutex lock(mLock); 721 float volume; 722 if (output) { 723 PlaybackThread *thread = checkPlaybackThread_l(output); 724 if (thread == NULL) { 725 return 0.0f; 726 } 727 volume = thread->streamVolume(stream); 728 } else { 729 volume = mStreamTypes[stream].volume; 730 } 731 732 return volume; 733} 734 735bool AudioFlinger::streamMute(audio_stream_type_t stream) const 736{ 737 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 738 return true; 739 } 740 741 return mStreamTypes[stream].mute; 742} 743 744status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 745{ 746 status_t result; 747 748 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 749 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 750 // check calling permissions 751 if (!settingsAllowed()) { 752 return PERMISSION_DENIED; 753 } 754 755 // ioHandle == 0 means the parameters are global to the audio hardware interface 756 if (ioHandle == 0) { 757 AutoMutex lock(mHardwareLock); 758 mHardwareStatus = AUDIO_SET_PARAMETER; 759 status_t final_result = NO_ERROR; 760 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 761 audio_hw_device_t *dev = mAudioHwDevs[i]; 762 result = dev->set_parameters(dev, keyValuePairs.string()); 763 final_result = result ?: final_result; 764 } 765 mHardwareStatus = AUDIO_HW_IDLE; 766 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 767 AudioParameter param = AudioParameter(keyValuePairs); 768 String8 value; 769 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 770 Mutex::Autolock _l(mLock); 771 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 772 if (mBtNrecIsOff != btNrecIsOff) { 773 for (size_t i = 0; i < mRecordThreads.size(); i++) { 774 sp<RecordThread> thread = mRecordThreads.valueAt(i); 775 RecordThread::RecordTrack *track = thread->track(); 776 if (track != NULL) { 777 audio_devices_t device = (audio_devices_t)( 778 thread->device() & AUDIO_DEVICE_IN_ALL); 779 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 780 thread->setEffectSuspended(FX_IID_AEC, 781 suspend, 782 track->sessionId()); 783 thread->setEffectSuspended(FX_IID_NS, 784 suspend, 785 track->sessionId()); 786 } 787 } 788 mBtNrecIsOff = btNrecIsOff; 789 } 790 } 791 return final_result; 792 } 793 794 // hold a strong ref on thread in case closeOutput() or closeInput() is called 795 // and the thread is exited once the lock is released 796 sp<ThreadBase> thread; 797 { 798 Mutex::Autolock _l(mLock); 799 thread = checkPlaybackThread_l(ioHandle); 800 if (thread == NULL) { 801 thread = checkRecordThread_l(ioHandle); 802 } else if (thread == primaryPlaybackThread_l()) { 803 // indicate output device change to all input threads for pre processing 804 AudioParameter param = AudioParameter(keyValuePairs); 805 int value; 806 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 807 for (size_t i = 0; i < mRecordThreads.size(); i++) { 808 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 809 } 810 } 811 } 812 } 813 if (thread != NULL) { 814 result = thread->setParameters(keyValuePairs); 815 return result; 816 } 817 return BAD_VALUE; 818} 819 820String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 821{ 822// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 823// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 824 825 if (ioHandle == 0) { 826 String8 out_s8; 827 828 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 829 audio_hw_device_t *dev = mAudioHwDevs[i]; 830 char *s = dev->get_parameters(dev, keys.string()); 831 out_s8 += String8(s); 832 free(s); 833 } 834 return out_s8; 835 } 836 837 Mutex::Autolock _l(mLock); 838 839 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 840 if (playbackThread != NULL) { 841 return playbackThread->getParameters(keys); 842 } 843 RecordThread *recordThread = checkRecordThread_l(ioHandle); 844 if (recordThread != NULL) { 845 return recordThread->getParameters(keys); 846 } 847 return String8(""); 848} 849 850size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) 851{ 852 status_t ret = initCheck(); 853 if (ret != NO_ERROR) { 854 return 0; 855 } 856 857 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 858} 859 860unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 861{ 862 if (ioHandle == 0) { 863 return 0; 864 } 865 866 Mutex::Autolock _l(mLock); 867 868 RecordThread *recordThread = checkRecordThread_l(ioHandle); 869 if (recordThread != NULL) { 870 return recordThread->getInputFramesLost(); 871 } 872 return 0; 873} 874 875status_t AudioFlinger::setVoiceVolume(float value) 876{ 877 status_t ret = initCheck(); 878 if (ret != NO_ERROR) { 879 return ret; 880 } 881 882 // check calling permissions 883 if (!settingsAllowed()) { 884 return PERMISSION_DENIED; 885 } 886 887 AutoMutex lock(mHardwareLock); 888 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 889 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 890 mHardwareStatus = AUDIO_HW_IDLE; 891 892 return ret; 893} 894 895status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 896{ 897 status_t status; 898 899 Mutex::Autolock _l(mLock); 900 901 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 902 if (playbackThread != NULL) { 903 return playbackThread->getRenderPosition(halFrames, dspFrames); 904 } 905 906 return BAD_VALUE; 907} 908 909void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 910{ 911 912 Mutex::Autolock _l(mLock); 913 914 int pid = IPCThreadState::self()->getCallingPid(); 915 if (mNotificationClients.indexOfKey(pid) < 0) { 916 sp<NotificationClient> notificationClient = new NotificationClient(this, 917 client, 918 pid); 919 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 920 921 mNotificationClients.add(pid, notificationClient); 922 923 sp<IBinder> binder = client->asBinder(); 924 binder->linkToDeath(notificationClient); 925 926 // the config change is always sent from playback or record threads to avoid deadlock 927 // with AudioSystem::gLock 928 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 929 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 930 } 931 932 for (size_t i = 0; i < mRecordThreads.size(); i++) { 933 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 934 } 935 } 936} 937 938void AudioFlinger::removeNotificationClient(pid_t pid) 939{ 940 Mutex::Autolock _l(mLock); 941 942 int index = mNotificationClients.indexOfKey(pid); 943 if (index >= 0) { 944 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 945 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 946 mNotificationClients.removeItem(pid); 947 } 948 949 ALOGV("%d died, releasing its sessions", pid); 950 int num = mAudioSessionRefs.size(); 951 bool removed = false; 952 for (int i = 0; i< num; i++) { 953 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 954 ALOGV(" pid %d @ %d", ref->pid, i); 955 if (ref->pid == pid) { 956 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 957 mAudioSessionRefs.removeAt(i); 958 delete ref; 959 removed = true; 960 i--; 961 num--; 962 } 963 } 964 if (removed) { 965 purgeStaleEffects_l(); 966 } 967} 968 969// audioConfigChanged_l() must be called with AudioFlinger::mLock held 970void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 971{ 972 size_t size = mNotificationClients.size(); 973 for (size_t i = 0; i < size; i++) { 974 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 975 } 976} 977 978// removeClient_l() must be called with AudioFlinger::mLock held 979void AudioFlinger::removeClient_l(pid_t pid) 980{ 981 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 982 mClients.removeItem(pid); 983} 984 985 986// ---------------------------------------------------------------------------- 987 988AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 989 : Thread(false), 990 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 991 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false), 992 mDevice(device) 993{ 994 mDeathRecipient = new PMDeathRecipient(this); 995} 996 997AudioFlinger::ThreadBase::~ThreadBase() 998{ 999 mParamCond.broadcast(); 1000 // do not lock the mutex in destructor 1001 releaseWakeLock_l(); 1002 if (mPowerManager != 0) { 1003 sp<IBinder> binder = mPowerManager->asBinder(); 1004 binder->unlinkToDeath(mDeathRecipient); 1005 } 1006} 1007 1008void AudioFlinger::ThreadBase::exit() 1009{ 1010 // keep a strong ref on ourself so that we won't get 1011 // destroyed in the middle of requestExitAndWait() 1012 sp <ThreadBase> strongMe = this; 1013 1014 ALOGV("ThreadBase::exit"); 1015 { 1016 AutoMutex lock(mLock); 1017 mExiting = true; 1018 requestExit(); 1019 mWaitWorkCV.signal(); 1020 } 1021 requestExitAndWait(); 1022} 1023 1024uint32_t AudioFlinger::ThreadBase::sampleRate() const 1025{ 1026 return mSampleRate; 1027} 1028 1029int AudioFlinger::ThreadBase::channelCount() const 1030{ 1031 return (int)mChannelCount; 1032} 1033 1034audio_format_t AudioFlinger::ThreadBase::format() const 1035{ 1036 return mFormat; 1037} 1038 1039size_t AudioFlinger::ThreadBase::frameCount() const 1040{ 1041 return mFrameCount; 1042} 1043 1044status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1045{ 1046 status_t status; 1047 1048 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1049 Mutex::Autolock _l(mLock); 1050 1051 mNewParameters.add(keyValuePairs); 1052 mWaitWorkCV.signal(); 1053 // wait condition with timeout in case the thread loop has exited 1054 // before the request could be processed 1055 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1056 status = mParamStatus; 1057 mWaitWorkCV.signal(); 1058 } else { 1059 status = TIMED_OUT; 1060 } 1061 return status; 1062} 1063 1064void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1065{ 1066 Mutex::Autolock _l(mLock); 1067 sendConfigEvent_l(event, param); 1068} 1069 1070// sendConfigEvent_l() must be called with ThreadBase::mLock held 1071void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1072{ 1073 ConfigEvent configEvent; 1074 configEvent.mEvent = event; 1075 configEvent.mParam = param; 1076 mConfigEvents.add(configEvent); 1077 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1078 mWaitWorkCV.signal(); 1079} 1080 1081void AudioFlinger::ThreadBase::processConfigEvents() 1082{ 1083 mLock.lock(); 1084 while(!mConfigEvents.isEmpty()) { 1085 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1086 ConfigEvent configEvent = mConfigEvents[0]; 1087 mConfigEvents.removeAt(0); 1088 // release mLock before locking AudioFlinger mLock: lock order is always 1089 // AudioFlinger then ThreadBase to avoid cross deadlock 1090 mLock.unlock(); 1091 mAudioFlinger->mLock.lock(); 1092 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1093 mAudioFlinger->mLock.unlock(); 1094 mLock.lock(); 1095 } 1096 mLock.unlock(); 1097} 1098 1099status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1100{ 1101 const size_t SIZE = 256; 1102 char buffer[SIZE]; 1103 String8 result; 1104 1105 bool locked = tryLock(mLock); 1106 if (!locked) { 1107 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1108 write(fd, buffer, strlen(buffer)); 1109 } 1110 1111 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1112 result.append(buffer); 1113 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1114 result.append(buffer); 1115 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1116 result.append(buffer); 1117 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1122 result.append(buffer); 1123 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1124 result.append(buffer); 1125 1126 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1127 result.append(buffer); 1128 result.append(" Index Command"); 1129 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1130 snprintf(buffer, SIZE, "\n %02d ", i); 1131 result.append(buffer); 1132 result.append(mNewParameters[i]); 1133 } 1134 1135 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1136 result.append(buffer); 1137 snprintf(buffer, SIZE, " Index event param\n"); 1138 result.append(buffer); 1139 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1140 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1141 result.append(buffer); 1142 } 1143 result.append("\n"); 1144 1145 write(fd, result.string(), result.size()); 1146 1147 if (locked) { 1148 mLock.unlock(); 1149 } 1150 return NO_ERROR; 1151} 1152 1153status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1154{ 1155 const size_t SIZE = 256; 1156 char buffer[SIZE]; 1157 String8 result; 1158 1159 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1160 write(fd, buffer, strlen(buffer)); 1161 1162 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1163 sp<EffectChain> chain = mEffectChains[i]; 1164 if (chain != 0) { 1165 chain->dump(fd, args); 1166 } 1167 } 1168 return NO_ERROR; 1169} 1170 1171void AudioFlinger::ThreadBase::acquireWakeLock() 1172{ 1173 Mutex::Autolock _l(mLock); 1174 acquireWakeLock_l(); 1175} 1176 1177void AudioFlinger::ThreadBase::acquireWakeLock_l() 1178{ 1179 if (mPowerManager == 0) { 1180 // use checkService() to avoid blocking if power service is not up yet 1181 sp<IBinder> binder = 1182 defaultServiceManager()->checkService(String16("power")); 1183 if (binder == 0) { 1184 ALOGW("Thread %s cannot connect to the power manager service", mName); 1185 } else { 1186 mPowerManager = interface_cast<IPowerManager>(binder); 1187 binder->linkToDeath(mDeathRecipient); 1188 } 1189 } 1190 if (mPowerManager != 0) { 1191 sp<IBinder> binder = new BBinder(); 1192 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1193 binder, 1194 String16(mName)); 1195 if (status == NO_ERROR) { 1196 mWakeLockToken = binder; 1197 } 1198 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1199 } 1200} 1201 1202void AudioFlinger::ThreadBase::releaseWakeLock() 1203{ 1204 Mutex::Autolock _l(mLock); 1205 releaseWakeLock_l(); 1206} 1207 1208void AudioFlinger::ThreadBase::releaseWakeLock_l() 1209{ 1210 if (mWakeLockToken != 0) { 1211 ALOGV("releaseWakeLock_l() %s", mName); 1212 if (mPowerManager != 0) { 1213 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1214 } 1215 mWakeLockToken.clear(); 1216 } 1217} 1218 1219void AudioFlinger::ThreadBase::clearPowerManager() 1220{ 1221 Mutex::Autolock _l(mLock); 1222 releaseWakeLock_l(); 1223 mPowerManager.clear(); 1224} 1225 1226void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1227{ 1228 sp<ThreadBase> thread = mThread.promote(); 1229 if (thread != 0) { 1230 thread->clearPowerManager(); 1231 } 1232 ALOGW("power manager service died !!!"); 1233} 1234 1235void AudioFlinger::ThreadBase::setEffectSuspended( 1236 const effect_uuid_t *type, bool suspend, int sessionId) 1237{ 1238 Mutex::Autolock _l(mLock); 1239 setEffectSuspended_l(type, suspend, sessionId); 1240} 1241 1242void AudioFlinger::ThreadBase::setEffectSuspended_l( 1243 const effect_uuid_t *type, bool suspend, int sessionId) 1244{ 1245 sp<EffectChain> chain; 1246 chain = getEffectChain_l(sessionId); 1247 if (chain != 0) { 1248 if (type != NULL) { 1249 chain->setEffectSuspended_l(type, suspend); 1250 } else { 1251 chain->setEffectSuspendedAll_l(suspend); 1252 } 1253 } 1254 1255 updateSuspendedSessions_l(type, suspend, sessionId); 1256} 1257 1258void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1259{ 1260 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1261 if (index < 0) { 1262 return; 1263 } 1264 1265 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1266 mSuspendedSessions.editValueAt(index); 1267 1268 for (size_t i = 0; i < sessionEffects.size(); i++) { 1269 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1270 for (int j = 0; j < desc->mRefCount; j++) { 1271 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1272 chain->setEffectSuspendedAll_l(true); 1273 } else { 1274 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1275 desc->mType.timeLow); 1276 chain->setEffectSuspended_l(&desc->mType, true); 1277 } 1278 } 1279 } 1280} 1281 1282void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1283 bool suspend, 1284 int sessionId) 1285{ 1286 int index = mSuspendedSessions.indexOfKey(sessionId); 1287 1288 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1289 1290 if (suspend) { 1291 if (index >= 0) { 1292 sessionEffects = mSuspendedSessions.editValueAt(index); 1293 } else { 1294 mSuspendedSessions.add(sessionId, sessionEffects); 1295 } 1296 } else { 1297 if (index < 0) { 1298 return; 1299 } 1300 sessionEffects = mSuspendedSessions.editValueAt(index); 1301 } 1302 1303 1304 int key = EffectChain::kKeyForSuspendAll; 1305 if (type != NULL) { 1306 key = type->timeLow; 1307 } 1308 index = sessionEffects.indexOfKey(key); 1309 1310 sp <SuspendedSessionDesc> desc; 1311 if (suspend) { 1312 if (index >= 0) { 1313 desc = sessionEffects.valueAt(index); 1314 } else { 1315 desc = new SuspendedSessionDesc(); 1316 if (type != NULL) { 1317 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1318 } 1319 sessionEffects.add(key, desc); 1320 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1321 } 1322 desc->mRefCount++; 1323 } else { 1324 if (index < 0) { 1325 return; 1326 } 1327 desc = sessionEffects.valueAt(index); 1328 if (--desc->mRefCount == 0) { 1329 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1330 sessionEffects.removeItemsAt(index); 1331 if (sessionEffects.isEmpty()) { 1332 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1333 sessionId); 1334 mSuspendedSessions.removeItem(sessionId); 1335 } 1336 } 1337 } 1338 if (!sessionEffects.isEmpty()) { 1339 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1340 } 1341} 1342 1343void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1344 bool enabled, 1345 int sessionId) 1346{ 1347 Mutex::Autolock _l(mLock); 1348 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1349} 1350 1351void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1352 bool enabled, 1353 int sessionId) 1354{ 1355 if (mType != RECORD) { 1356 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1357 // another session. This gives the priority to well behaved effect control panels 1358 // and applications not using global effects. 1359 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1360 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1361 } 1362 } 1363 1364 sp<EffectChain> chain = getEffectChain_l(sessionId); 1365 if (chain != 0) { 1366 chain->checkSuspendOnEffectEnabled(effect, enabled); 1367 } 1368} 1369 1370// ---------------------------------------------------------------------------- 1371 1372AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1373 AudioStreamOut* output, 1374 int id, 1375 uint32_t device) 1376 : ThreadBase(audioFlinger, id, device), 1377 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1378 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1379{ 1380 snprintf(mName, kNameLength, "AudioOut_%d", id); 1381 1382 readOutputParameters(); 1383 1384 // Assumes constructor is called by AudioFlinger with it's mLock held, 1385 // but it would be safer to explicitly pass these as parameters 1386 mMasterVolume = mAudioFlinger->masterVolume_l(); 1387 mMasterMute = mAudioFlinger->masterMute_l(); 1388 1389 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1390 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1391 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1392 stream = (audio_stream_type_t) (stream + 1)) { 1393 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1394 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1395 // initialized by stream_type_t default constructor 1396 // mStreamTypes[stream].valid = true; 1397 } 1398} 1399 1400AudioFlinger::PlaybackThread::~PlaybackThread() 1401{ 1402 delete [] mMixBuffer; 1403} 1404 1405status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1406{ 1407 dumpInternals(fd, args); 1408 dumpTracks(fd, args); 1409 dumpEffectChains(fd, args); 1410 return NO_ERROR; 1411} 1412 1413status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1414{ 1415 const size_t SIZE = 256; 1416 char buffer[SIZE]; 1417 String8 result; 1418 1419 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1420 result.append(buffer); 1421 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1422 for (size_t i = 0; i < mTracks.size(); ++i) { 1423 sp<Track> track = mTracks[i]; 1424 if (track != 0) { 1425 track->dump(buffer, SIZE); 1426 result.append(buffer); 1427 } 1428 } 1429 1430 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1431 result.append(buffer); 1432 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1433 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1434 wp<Track> wTrack = mActiveTracks[i]; 1435 if (wTrack != 0) { 1436 sp<Track> track = wTrack.promote(); 1437 if (track != 0) { 1438 track->dump(buffer, SIZE); 1439 result.append(buffer); 1440 } 1441 } 1442 } 1443 write(fd, result.string(), result.size()); 1444 return NO_ERROR; 1445} 1446 1447status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1448{ 1449 const size_t SIZE = 256; 1450 char buffer[SIZE]; 1451 String8 result; 1452 1453 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1454 result.append(buffer); 1455 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1456 result.append(buffer); 1457 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1458 result.append(buffer); 1459 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1460 result.append(buffer); 1461 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1462 result.append(buffer); 1463 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1464 result.append(buffer); 1465 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1466 result.append(buffer); 1467 write(fd, result.string(), result.size()); 1468 1469 dumpBase(fd, args); 1470 1471 return NO_ERROR; 1472} 1473 1474// Thread virtuals 1475status_t AudioFlinger::PlaybackThread::readyToRun() 1476{ 1477 status_t status = initCheck(); 1478 if (status == NO_ERROR) { 1479 ALOGI("AudioFlinger's thread %p ready to run", this); 1480 } else { 1481 ALOGE("No working audio driver found."); 1482 } 1483 return status; 1484} 1485 1486void AudioFlinger::PlaybackThread::onFirstRef() 1487{ 1488 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1489} 1490 1491// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1492sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1493 const sp<AudioFlinger::Client>& client, 1494 audio_stream_type_t streamType, 1495 uint32_t sampleRate, 1496 audio_format_t format, 1497 uint32_t channelMask, 1498 int frameCount, 1499 const sp<IMemory>& sharedBuffer, 1500 int sessionId, 1501 status_t *status) 1502{ 1503 sp<Track> track; 1504 status_t lStatus; 1505 1506 if (mType == DIRECT) { 1507 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1508 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1509 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1510 "for output %p with format %d", 1511 sampleRate, format, channelMask, mOutput, mFormat); 1512 lStatus = BAD_VALUE; 1513 goto Exit; 1514 } 1515 } 1516 } else { 1517 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1518 if (sampleRate > mSampleRate*2) { 1519 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1520 lStatus = BAD_VALUE; 1521 goto Exit; 1522 } 1523 } 1524 1525 lStatus = initCheck(); 1526 if (lStatus != NO_ERROR) { 1527 ALOGE("Audio driver not initialized."); 1528 goto Exit; 1529 } 1530 1531 { // scope for mLock 1532 Mutex::Autolock _l(mLock); 1533 1534 // all tracks in same audio session must share the same routing strategy otherwise 1535 // conflicts will happen when tracks are moved from one output to another by audio policy 1536 // manager 1537 uint32_t strategy = 1538 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1539 for (size_t i = 0; i < mTracks.size(); ++i) { 1540 sp<Track> t = mTracks[i]; 1541 if (t != 0) { 1542 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1543 if (sessionId == t->sessionId() && strategy != actual) { 1544 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1545 strategy, actual); 1546 lStatus = BAD_VALUE; 1547 goto Exit; 1548 } 1549 } 1550 } 1551 1552 track = new Track(this, client, streamType, sampleRate, format, 1553 channelMask, frameCount, sharedBuffer, sessionId); 1554 if (track->getCblk() == NULL || track->name() < 0) { 1555 lStatus = NO_MEMORY; 1556 goto Exit; 1557 } 1558 mTracks.add(track); 1559 1560 sp<EffectChain> chain = getEffectChain_l(sessionId); 1561 if (chain != 0) { 1562 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1563 track->setMainBuffer(chain->inBuffer()); 1564 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1565 chain->incTrackCnt(); 1566 } 1567 1568 // invalidate track immediately if the stream type was moved to another thread since 1569 // createTrack() was called by the client process. 1570 if (!mStreamTypes[streamType].valid) { 1571 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1572 this, streamType); 1573 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1574 } 1575 } 1576 lStatus = NO_ERROR; 1577 1578Exit: 1579 if(status) { 1580 *status = lStatus; 1581 } 1582 return track; 1583} 1584 1585uint32_t AudioFlinger::PlaybackThread::latency() const 1586{ 1587 Mutex::Autolock _l(mLock); 1588 if (initCheck() == NO_ERROR) { 1589 return mOutput->stream->get_latency(mOutput->stream); 1590 } else { 1591 return 0; 1592 } 1593} 1594 1595status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1596{ 1597 mMasterVolume = value; 1598 return NO_ERROR; 1599} 1600 1601status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1602{ 1603 mMasterMute = muted; 1604 return NO_ERROR; 1605} 1606 1607float AudioFlinger::PlaybackThread::masterVolume() const 1608{ 1609 return mMasterVolume; 1610} 1611 1612bool AudioFlinger::PlaybackThread::masterMute() const 1613{ 1614 return mMasterMute; 1615} 1616 1617status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1618{ 1619 mStreamTypes[stream].volume = value; 1620 return NO_ERROR; 1621} 1622 1623status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1624{ 1625 mStreamTypes[stream].mute = muted; 1626 return NO_ERROR; 1627} 1628 1629float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1630{ 1631 return mStreamTypes[stream].volume; 1632} 1633 1634bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1635{ 1636 return mStreamTypes[stream].mute; 1637} 1638 1639// addTrack_l() must be called with ThreadBase::mLock held 1640status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1641{ 1642 status_t status = ALREADY_EXISTS; 1643 1644 // set retry count for buffer fill 1645 track->mRetryCount = kMaxTrackStartupRetries; 1646 if (mActiveTracks.indexOf(track) < 0) { 1647 // the track is newly added, make sure it fills up all its 1648 // buffers before playing. This is to ensure the client will 1649 // effectively get the latency it requested. 1650 track->mFillingUpStatus = Track::FS_FILLING; 1651 track->mResetDone = false; 1652 mActiveTracks.add(track); 1653 if (track->mainBuffer() != mMixBuffer) { 1654 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1655 if (chain != 0) { 1656 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1657 chain->incActiveTrackCnt(); 1658 } 1659 } 1660 1661 status = NO_ERROR; 1662 } 1663 1664 ALOGV("mWaitWorkCV.broadcast"); 1665 mWaitWorkCV.broadcast(); 1666 1667 return status; 1668} 1669 1670// destroyTrack_l() must be called with ThreadBase::mLock held 1671void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1672{ 1673 track->mState = TrackBase::TERMINATED; 1674 if (mActiveTracks.indexOf(track) < 0) { 1675 removeTrack_l(track); 1676 } 1677} 1678 1679void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1680{ 1681 mTracks.remove(track); 1682 deleteTrackName_l(track->name()); 1683 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1684 if (chain != 0) { 1685 chain->decTrackCnt(); 1686 } 1687} 1688 1689String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1690{ 1691 String8 out_s8 = String8(""); 1692 char *s; 1693 1694 Mutex::Autolock _l(mLock); 1695 if (initCheck() != NO_ERROR) { 1696 return out_s8; 1697 } 1698 1699 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1700 out_s8 = String8(s); 1701 free(s); 1702 return out_s8; 1703} 1704 1705// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1706void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1707 AudioSystem::OutputDescriptor desc; 1708 void *param2 = 0; 1709 1710 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1711 1712 switch (event) { 1713 case AudioSystem::OUTPUT_OPENED: 1714 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1715 desc.channels = mChannelMask; 1716 desc.samplingRate = mSampleRate; 1717 desc.format = mFormat; 1718 desc.frameCount = mFrameCount; 1719 desc.latency = latency(); 1720 param2 = &desc; 1721 break; 1722 1723 case AudioSystem::STREAM_CONFIG_CHANGED: 1724 param2 = ¶m; 1725 case AudioSystem::OUTPUT_CLOSED: 1726 default: 1727 break; 1728 } 1729 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1730} 1731 1732void AudioFlinger::PlaybackThread::readOutputParameters() 1733{ 1734 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1735 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1736 mChannelCount = (uint16_t)popcount(mChannelMask); 1737 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1738 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1739 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1740 1741 // FIXME - Current mixer implementation only supports stereo output: Always 1742 // Allocate a stereo buffer even if HW output is mono. 1743 if (mMixBuffer != NULL) delete[] mMixBuffer; 1744 mMixBuffer = new int16_t[mFrameCount * 2]; 1745 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1746 1747 // force reconfiguration of effect chains and engines to take new buffer size and audio 1748 // parameters into account 1749 // Note that mLock is not held when readOutputParameters() is called from the constructor 1750 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1751 // matter. 1752 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1753 Vector< sp<EffectChain> > effectChains = mEffectChains; 1754 for (size_t i = 0; i < effectChains.size(); i ++) { 1755 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1756 } 1757} 1758 1759status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1760{ 1761 if (halFrames == 0 || dspFrames == 0) { 1762 return BAD_VALUE; 1763 } 1764 Mutex::Autolock _l(mLock); 1765 if (initCheck() != NO_ERROR) { 1766 return INVALID_OPERATION; 1767 } 1768 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1769 1770 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1771} 1772 1773uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1774{ 1775 Mutex::Autolock _l(mLock); 1776 uint32_t result = 0; 1777 if (getEffectChain_l(sessionId) != 0) { 1778 result = EFFECT_SESSION; 1779 } 1780 1781 for (size_t i = 0; i < mTracks.size(); ++i) { 1782 sp<Track> track = mTracks[i]; 1783 if (sessionId == track->sessionId() && 1784 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1785 result |= TRACK_SESSION; 1786 break; 1787 } 1788 } 1789 1790 return result; 1791} 1792 1793uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1794{ 1795 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1796 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1797 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1798 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1799 } 1800 for (size_t i = 0; i < mTracks.size(); i++) { 1801 sp<Track> track = mTracks[i]; 1802 if (sessionId == track->sessionId() && 1803 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1804 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1805 } 1806 } 1807 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1808} 1809 1810 1811AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() 1812{ 1813 Mutex::Autolock _l(mLock); 1814 return mOutput; 1815} 1816 1817AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1818{ 1819 Mutex::Autolock _l(mLock); 1820 AudioStreamOut *output = mOutput; 1821 mOutput = NULL; 1822 return output; 1823} 1824 1825// this method must always be called either with ThreadBase mLock held or inside the thread loop 1826audio_stream_t* AudioFlinger::PlaybackThread::stream() 1827{ 1828 if (mOutput == NULL) { 1829 return NULL; 1830 } 1831 return &mOutput->stream->common; 1832} 1833 1834uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1835{ 1836 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1837 // decoding and transfer time. So sleeping for half of the latency would likely cause 1838 // underruns 1839 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1840 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1841 } else { 1842 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1843 } 1844} 1845 1846// ---------------------------------------------------------------------------- 1847 1848AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1849 : PlaybackThread(audioFlinger, output, id, device), 1850 mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE) 1851{ 1852 mType = ThreadBase::MIXER; 1853 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1854 1855 // FIXME - Current mixer implementation only supports stereo output 1856 if (mChannelCount == 1) { 1857 ALOGE("Invalid audio hardware channel count"); 1858 } 1859} 1860 1861AudioFlinger::MixerThread::~MixerThread() 1862{ 1863 delete mAudioMixer; 1864} 1865 1866bool AudioFlinger::MixerThread::threadLoop() 1867{ 1868 Vector< sp<Track> > tracksToRemove; 1869 uint32_t mixerStatus = MIXER_IDLE; 1870 nsecs_t standbyTime = systemTime(); 1871 size_t mixBufferSize = mFrameCount * mFrameSize; 1872 // FIXME: Relaxed timing because of a certain device that can't meet latency 1873 // Should be reduced to 2x after the vendor fixes the driver issue 1874 // increase threshold again due to low power audio mode. The way this warning threshold is 1875 // calculated and its usefulness should be reconsidered anyway. 1876 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1877 nsecs_t lastWarning = 0; 1878 bool longStandbyExit = false; 1879 uint32_t activeSleepTime = activeSleepTimeUs(); 1880 uint32_t idleSleepTime = idleSleepTimeUs(); 1881 uint32_t sleepTime = idleSleepTime; 1882 uint32_t sleepTimeShift = 0; 1883 Vector< sp<EffectChain> > effectChains; 1884#ifdef DEBUG_CPU_USAGE 1885 ThreadCpuUsage cpu; 1886 const CentralTendencyStatistics& stats = cpu.statistics(); 1887#endif 1888 1889 acquireWakeLock(); 1890 1891 while (!exitPending()) 1892 { 1893#ifdef DEBUG_CPU_USAGE 1894 cpu.sampleAndEnable(); 1895 unsigned n = stats.n(); 1896 // cpu.elapsed() is expensive, so don't call it every loop 1897 if ((n & 127) == 1) { 1898 long long elapsed = cpu.elapsed(); 1899 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1900 double perLoop = elapsed / (double) n; 1901 double perLoop100 = perLoop * 0.01; 1902 double mean = stats.mean(); 1903 double stddev = stats.stddev(); 1904 double minimum = stats.minimum(); 1905 double maximum = stats.maximum(); 1906 cpu.resetStatistics(); 1907 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1908 elapsed * .000000001, n, perLoop * .000001, 1909 mean * .001, 1910 stddev * .001, 1911 minimum * .001, 1912 maximum * .001, 1913 mean / perLoop100, 1914 stddev / perLoop100, 1915 minimum / perLoop100, 1916 maximum / perLoop100); 1917 } 1918 } 1919#endif 1920 processConfigEvents(); 1921 1922 mixerStatus = MIXER_IDLE; 1923 { // scope for mLock 1924 1925 Mutex::Autolock _l(mLock); 1926 1927 if (checkForNewParameters_l()) { 1928 mixBufferSize = mFrameCount * mFrameSize; 1929 // FIXME: Relaxed timing because of a certain device that can't meet latency 1930 // Should be reduced to 2x after the vendor fixes the driver issue 1931 // increase threshold again due to low power audio mode. The way this warning 1932 // threshold is calculated and its usefulness should be reconsidered anyway. 1933 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1934 activeSleepTime = activeSleepTimeUs(); 1935 idleSleepTime = idleSleepTimeUs(); 1936 } 1937 1938 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1939 1940 // put audio hardware into standby after short delay 1941 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1942 mSuspended)) { 1943 if (!mStandby) { 1944 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1945 mOutput->stream->common.standby(&mOutput->stream->common); 1946 mStandby = true; 1947 mBytesWritten = 0; 1948 } 1949 1950 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1951 // we're about to wait, flush the binder command buffer 1952 IPCThreadState::self()->flushCommands(); 1953 1954 if (exitPending()) break; 1955 1956 releaseWakeLock_l(); 1957 // wait until we have something to do... 1958 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1959 mWaitWorkCV.wait(mLock); 1960 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1961 acquireWakeLock_l(); 1962 1963 mPrevMixerStatus = MIXER_IDLE; 1964 if (!mMasterMute) { 1965 char value[PROPERTY_VALUE_MAX]; 1966 property_get("ro.audio.silent", value, "0"); 1967 if (atoi(value)) { 1968 ALOGD("Silence is golden"); 1969 setMasterMute(true); 1970 } 1971 } 1972 1973 standbyTime = systemTime() + kStandbyTimeInNsecs; 1974 sleepTime = idleSleepTime; 1975 sleepTimeShift = 0; 1976 continue; 1977 } 1978 } 1979 1980 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1981 1982 // prevent any changes in effect chain list and in each effect chain 1983 // during mixing and effect process as the audio buffers could be deleted 1984 // or modified if an effect is created or deleted 1985 lockEffectChains_l(effectChains); 1986 } 1987 1988 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1989 // mix buffers... 1990 mAudioMixer->process(); 1991 sleepTime = 0; 1992 // increase sleep time progressively when application underrun condition clears 1993 if (sleepTimeShift > 0) { 1994 sleepTimeShift--; 1995 } 1996 standbyTime = systemTime() + kStandbyTimeInNsecs; 1997 //TODO: delay standby when effects have a tail 1998 } else { 1999 // If no tracks are ready, sleep once for the duration of an output 2000 // buffer size, then write 0s to the output 2001 if (sleepTime == 0) { 2002 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2003 sleepTime = activeSleepTime >> sleepTimeShift; 2004 if (sleepTime < kMinThreadSleepTimeUs) { 2005 sleepTime = kMinThreadSleepTimeUs; 2006 } 2007 // reduce sleep time in case of consecutive application underruns to avoid 2008 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2009 // duration we would end up writing less data than needed by the audio HAL if 2010 // the condition persists. 2011 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2012 sleepTimeShift++; 2013 } 2014 } else { 2015 sleepTime = idleSleepTime; 2016 } 2017 } else if (mBytesWritten != 0 || 2018 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2019 memset (mMixBuffer, 0, mixBufferSize); 2020 sleepTime = 0; 2021 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2022 } 2023 // TODO add standby time extension fct of effect tail 2024 } 2025 2026 if (mSuspended) { 2027 sleepTime = suspendSleepTimeUs(); 2028 } 2029 // sleepTime == 0 means we must write to audio hardware 2030 if (sleepTime == 0) { 2031 for (size_t i = 0; i < effectChains.size(); i ++) { 2032 effectChains[i]->process_l(); 2033 } 2034 // enable changes in effect chain 2035 unlockEffectChains(effectChains); 2036 mLastWriteTime = systemTime(); 2037 mInWrite = true; 2038 mBytesWritten += mixBufferSize; 2039 2040 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2041 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2042 mNumWrites++; 2043 mInWrite = false; 2044 nsecs_t now = systemTime(); 2045 nsecs_t delta = now - mLastWriteTime; 2046 if (!mStandby && delta > maxPeriod) { 2047 mNumDelayedWrites++; 2048 if ((now - lastWarning) > kWarningThrottleNs) { 2049 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2050 ns2ms(delta), mNumDelayedWrites, this); 2051 lastWarning = now; 2052 } 2053 if (mStandby) { 2054 longStandbyExit = true; 2055 } 2056 } 2057 mStandby = false; 2058 } else { 2059 // enable changes in effect chain 2060 unlockEffectChains(effectChains); 2061 usleep(sleepTime); 2062 } 2063 2064 // finally let go of all our tracks, without the lock held 2065 // since we can't guarantee the destructors won't acquire that 2066 // same lock. 2067 tracksToRemove.clear(); 2068 2069 // Effect chains will be actually deleted here if they were removed from 2070 // mEffectChains list during mixing or effects processing 2071 effectChains.clear(); 2072 } 2073 2074 if (!mStandby) { 2075 mOutput->stream->common.standby(&mOutput->stream->common); 2076 } 2077 2078 releaseWakeLock(); 2079 2080 ALOGV("MixerThread %p exiting", this); 2081 return false; 2082} 2083 2084// prepareTracks_l() must be called with ThreadBase::mLock held 2085uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2086{ 2087 2088 uint32_t mixerStatus = MIXER_IDLE; 2089 // find out which tracks need to be processed 2090 size_t count = activeTracks.size(); 2091 size_t mixedTracks = 0; 2092 size_t tracksWithEffect = 0; 2093 2094 float masterVolume = mMasterVolume; 2095 bool masterMute = mMasterMute; 2096 2097 if (masterMute) { 2098 masterVolume = 0; 2099 } 2100 // Delegate master volume control to effect in output mix effect chain if needed 2101 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2102 if (chain != 0) { 2103 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2104 chain->setVolume_l(&v, &v); 2105 masterVolume = (float)((v + (1 << 23)) >> 24); 2106 chain.clear(); 2107 } 2108 2109 for (size_t i=0 ; i<count ; i++) { 2110 sp<Track> t = activeTracks[i].promote(); 2111 if (t == 0) continue; 2112 2113 // this const just means the local variable doesn't change 2114 Track* const track = t.get(); 2115 audio_track_cblk_t* cblk = track->cblk(); 2116 2117 // The first time a track is added we wait 2118 // for all its buffers to be filled before processing it 2119 int name = track->name(); 2120 // make sure that we have enough frames to mix one full buffer. 2121 // enforce this condition only once to enable draining the buffer in case the client 2122 // app does not call stop() and relies on underrun to stop: 2123 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2124 // during last round 2125 uint32_t minFrames = 1; 2126 if (!track->isStopped() && !track->isPausing() && 2127 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2128 if (t->sampleRate() == (int)mSampleRate) { 2129 minFrames = mFrameCount; 2130 } else { 2131 // +1 for rounding and +1 for additional sample needed for interpolation 2132 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2133 // add frames already consumed but not yet released by the resampler 2134 // because cblk->framesReady() will include these frames 2135 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2136 // the minimum track buffer size is normally twice the number of frames necessary 2137 // to fill one buffer and the resampler should not leave more than one buffer worth 2138 // of unreleased frames after each pass, but just in case... 2139 ALOG_ASSERT(minFrames <= cblk->frameCount); 2140 } 2141 } 2142 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2143 !track->isPaused() && !track->isTerminated()) 2144 { 2145 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2146 2147 mixedTracks++; 2148 2149 // track->mainBuffer() != mMixBuffer means there is an effect chain 2150 // connected to the track 2151 chain.clear(); 2152 if (track->mainBuffer() != mMixBuffer) { 2153 chain = getEffectChain_l(track->sessionId()); 2154 // Delegate volume control to effect in track effect chain if needed 2155 if (chain != 0) { 2156 tracksWithEffect++; 2157 } else { 2158 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2159 name, track->sessionId()); 2160 } 2161 } 2162 2163 2164 int param = AudioMixer::VOLUME; 2165 if (track->mFillingUpStatus == Track::FS_FILLED) { 2166 // no ramp for the first volume setting 2167 track->mFillingUpStatus = Track::FS_ACTIVE; 2168 if (track->mState == TrackBase::RESUMING) { 2169 track->mState = TrackBase::ACTIVE; 2170 param = AudioMixer::RAMP_VOLUME; 2171 } 2172 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2173 } else if (cblk->server != 0) { 2174 // If the track is stopped before the first frame was mixed, 2175 // do not apply ramp 2176 param = AudioMixer::RAMP_VOLUME; 2177 } 2178 2179 // compute volume for this track 2180 uint32_t vl, vr, va; 2181 if (track->isMuted() || track->isPausing() || 2182 mStreamTypes[track->type()].mute) { 2183 vl = vr = va = 0; 2184 if (track->isPausing()) { 2185 track->setPaused(); 2186 } 2187 } else { 2188 2189 // read original volumes with volume control 2190 float typeVolume = mStreamTypes[track->type()].volume; 2191 float v = masterVolume * typeVolume; 2192 uint32_t vlr = cblk->volumeLR; 2193 vl = vlr & 0xFFFF; 2194 vr = vlr >> 16; 2195 // track volumes come from shared memory, so can't be trusted and must be clamped 2196 if (vl > MAX_GAIN_INT) { 2197 ALOGV("Track left volume out of range: %04X", vl); 2198 vl = MAX_GAIN_INT; 2199 } 2200 if (vr > MAX_GAIN_INT) { 2201 ALOGV("Track right volume out of range: %04X", vr); 2202 vr = MAX_GAIN_INT; 2203 } 2204 // now apply the master volume and stream type volume 2205 vl = (uint32_t)(v * vl) << 12; 2206 vr = (uint32_t)(v * vr) << 12; 2207 // assuming master volume and stream type volume each go up to 1.0, 2208 // vl and vr are now in 8.24 format 2209 2210 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2211 // send level comes from shared memory and so may be corrupt 2212 if (sendLevel >= MAX_GAIN_INT) { 2213 ALOGV("Track send level out of range: %04X", sendLevel); 2214 sendLevel = MAX_GAIN_INT; 2215 } 2216 va = (uint32_t)(v * sendLevel); 2217 } 2218 // Delegate volume control to effect in track effect chain if needed 2219 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2220 // Do not ramp volume if volume is controlled by effect 2221 param = AudioMixer::VOLUME; 2222 track->mHasVolumeController = true; 2223 } else { 2224 // force no volume ramp when volume controller was just disabled or removed 2225 // from effect chain to avoid volume spike 2226 if (track->mHasVolumeController) { 2227 param = AudioMixer::VOLUME; 2228 } 2229 track->mHasVolumeController = false; 2230 } 2231 2232 // Convert volumes from 8.24 to 4.12 format 2233 int16_t left, right, aux; 2234 // This additional clamping is needed in case chain->setVolume_l() overshot 2235 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2236 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2237 left = int16_t(v_clamped); 2238 v_clamped = (vr + (1 << 11)) >> 12; 2239 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2240 right = int16_t(v_clamped); 2241 2242 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2243 aux = int16_t(va); 2244 2245 // XXX: these things DON'T need to be done each time 2246 mAudioMixer->setBufferProvider(name, track); 2247 mAudioMixer->enable(name); 2248 2249 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2250 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2251 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2252 mAudioMixer->setParameter( 2253 name, 2254 AudioMixer::TRACK, 2255 AudioMixer::FORMAT, (void *)track->format()); 2256 mAudioMixer->setParameter( 2257 name, 2258 AudioMixer::TRACK, 2259 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2260 mAudioMixer->setParameter( 2261 name, 2262 AudioMixer::RESAMPLE, 2263 AudioMixer::SAMPLE_RATE, 2264 (void *)(cblk->sampleRate)); 2265 mAudioMixer->setParameter( 2266 name, 2267 AudioMixer::TRACK, 2268 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2269 mAudioMixer->setParameter( 2270 name, 2271 AudioMixer::TRACK, 2272 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2273 2274 // reset retry count 2275 track->mRetryCount = kMaxTrackRetries; 2276 // If one track is ready, set the mixer ready if: 2277 // - the mixer was not ready during previous round OR 2278 // - no other track is not ready 2279 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2280 mixerStatus != MIXER_TRACKS_ENABLED) { 2281 mixerStatus = MIXER_TRACKS_READY; 2282 } 2283 } else { 2284 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2285 if (track->isStopped()) { 2286 track->reset(); 2287 } 2288 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2289 // We have consumed all the buffers of this track. 2290 // Remove it from the list of active tracks. 2291 tracksToRemove->add(track); 2292 } else { 2293 // No buffers for this track. Give it a few chances to 2294 // fill a buffer, then remove it from active list. 2295 if (--(track->mRetryCount) <= 0) { 2296 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2297 tracksToRemove->add(track); 2298 // indicate to client process that the track was disabled because of underrun 2299 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2300 // If one track is not ready, mark the mixer also not ready if: 2301 // - the mixer was ready during previous round OR 2302 // - no other track is ready 2303 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2304 mixerStatus != MIXER_TRACKS_READY) { 2305 mixerStatus = MIXER_TRACKS_ENABLED; 2306 } 2307 } 2308 mAudioMixer->disable(name); 2309 } 2310 } 2311 2312 // remove all the tracks that need to be... 2313 count = tracksToRemove->size(); 2314 if (CC_UNLIKELY(count)) { 2315 for (size_t i=0 ; i<count ; i++) { 2316 const sp<Track>& track = tracksToRemove->itemAt(i); 2317 mActiveTracks.remove(track); 2318 if (track->mainBuffer() != mMixBuffer) { 2319 chain = getEffectChain_l(track->sessionId()); 2320 if (chain != 0) { 2321 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2322 chain->decActiveTrackCnt(); 2323 } 2324 } 2325 if (track->isTerminated()) { 2326 removeTrack_l(track); 2327 } 2328 } 2329 } 2330 2331 // mix buffer must be cleared if all tracks are connected to an 2332 // effect chain as in this case the mixer will not write to 2333 // mix buffer and track effects will accumulate into it 2334 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2335 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2336 } 2337 2338 mPrevMixerStatus = mixerStatus; 2339 return mixerStatus; 2340} 2341 2342void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2343{ 2344 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2345 this, streamType, mTracks.size()); 2346 Mutex::Autolock _l(mLock); 2347 2348 size_t size = mTracks.size(); 2349 for (size_t i = 0; i < size; i++) { 2350 sp<Track> t = mTracks[i]; 2351 if (t->type() == streamType) { 2352 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2353 t->mCblk->cv.signal(); 2354 } 2355 } 2356} 2357 2358void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2359{ 2360 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2361 this, streamType, valid); 2362 Mutex::Autolock _l(mLock); 2363 2364 mStreamTypes[streamType].valid = valid; 2365} 2366 2367// getTrackName_l() must be called with ThreadBase::mLock held 2368int AudioFlinger::MixerThread::getTrackName_l() 2369{ 2370 return mAudioMixer->getTrackName(); 2371} 2372 2373// deleteTrackName_l() must be called with ThreadBase::mLock held 2374void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2375{ 2376 ALOGV("remove track (%d) and delete from mixer", name); 2377 mAudioMixer->deleteTrackName(name); 2378} 2379 2380// checkForNewParameters_l() must be called with ThreadBase::mLock held 2381bool AudioFlinger::MixerThread::checkForNewParameters_l() 2382{ 2383 bool reconfig = false; 2384 2385 while (!mNewParameters.isEmpty()) { 2386 status_t status = NO_ERROR; 2387 String8 keyValuePair = mNewParameters[0]; 2388 AudioParameter param = AudioParameter(keyValuePair); 2389 int value; 2390 2391 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2392 reconfig = true; 2393 } 2394 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2395 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2396 status = BAD_VALUE; 2397 } else { 2398 reconfig = true; 2399 } 2400 } 2401 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2402 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2403 status = BAD_VALUE; 2404 } else { 2405 reconfig = true; 2406 } 2407 } 2408 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2409 // do not accept frame count changes if tracks are open as the track buffer 2410 // size depends on frame count and correct behavior would not be guaranteed 2411 // if frame count is changed after track creation 2412 if (!mTracks.isEmpty()) { 2413 status = INVALID_OPERATION; 2414 } else { 2415 reconfig = true; 2416 } 2417 } 2418 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2419 // when changing the audio output device, call addBatteryData to notify 2420 // the change 2421 if ((int)mDevice != value) { 2422 uint32_t params = 0; 2423 // check whether speaker is on 2424 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2425 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2426 } 2427 2428 int deviceWithoutSpeaker 2429 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2430 // check if any other device (except speaker) is on 2431 if (value & deviceWithoutSpeaker ) { 2432 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2433 } 2434 2435 if (params != 0) { 2436 addBatteryData(params); 2437 } 2438 } 2439 2440 // forward device change to effects that have requested to be 2441 // aware of attached audio device. 2442 mDevice = (uint32_t)value; 2443 for (size_t i = 0; i < mEffectChains.size(); i++) { 2444 mEffectChains[i]->setDevice_l(mDevice); 2445 } 2446 } 2447 2448 if (status == NO_ERROR) { 2449 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2450 keyValuePair.string()); 2451 if (!mStandby && status == INVALID_OPERATION) { 2452 mOutput->stream->common.standby(&mOutput->stream->common); 2453 mStandby = true; 2454 mBytesWritten = 0; 2455 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2456 keyValuePair.string()); 2457 } 2458 if (status == NO_ERROR && reconfig) { 2459 delete mAudioMixer; 2460 readOutputParameters(); 2461 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2462 for (size_t i = 0; i < mTracks.size() ; i++) { 2463 int name = getTrackName_l(); 2464 if (name < 0) break; 2465 mTracks[i]->mName = name; 2466 // limit track sample rate to 2 x new output sample rate 2467 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2468 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2469 } 2470 } 2471 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2472 } 2473 } 2474 2475 mNewParameters.removeAt(0); 2476 2477 mParamStatus = status; 2478 mParamCond.signal(); 2479 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2480 // already timed out waiting for the status and will never signal the condition. 2481 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2482 } 2483 return reconfig; 2484} 2485 2486status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2487{ 2488 const size_t SIZE = 256; 2489 char buffer[SIZE]; 2490 String8 result; 2491 2492 PlaybackThread::dumpInternals(fd, args); 2493 2494 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2495 result.append(buffer); 2496 write(fd, result.string(), result.size()); 2497 return NO_ERROR; 2498} 2499 2500uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2501{ 2502 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2503} 2504 2505uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2506{ 2507 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2508} 2509 2510// ---------------------------------------------------------------------------- 2511AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2512 : PlaybackThread(audioFlinger, output, id, device) 2513{ 2514 mType = ThreadBase::DIRECT; 2515} 2516 2517AudioFlinger::DirectOutputThread::~DirectOutputThread() 2518{ 2519} 2520 2521static inline 2522int32_t mul(int16_t in, int16_t v) 2523{ 2524#if defined(__arm__) && !defined(__thumb__) 2525 int32_t out; 2526 asm( "smulbb %[out], %[in], %[v] \n" 2527 : [out]"=r"(out) 2528 : [in]"%r"(in), [v]"r"(v) 2529 : ); 2530 return out; 2531#else 2532 return in * int32_t(v); 2533#endif 2534} 2535 2536void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2537{ 2538 // Do not apply volume on compressed audio 2539 if (!audio_is_linear_pcm(mFormat)) { 2540 return; 2541 } 2542 2543 // convert to signed 16 bit before volume calculation 2544 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2545 size_t count = mFrameCount * mChannelCount; 2546 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2547 int16_t *dst = mMixBuffer + count-1; 2548 while(count--) { 2549 *dst-- = (int16_t)(*src--^0x80) << 8; 2550 } 2551 } 2552 2553 size_t frameCount = mFrameCount; 2554 int16_t *out = mMixBuffer; 2555 if (ramp) { 2556 if (mChannelCount == 1) { 2557 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2558 int32_t vlInc = d / (int32_t)frameCount; 2559 int32_t vl = ((int32_t)mLeftVolShort << 16); 2560 do { 2561 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2562 out++; 2563 vl += vlInc; 2564 } while (--frameCount); 2565 2566 } else { 2567 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2568 int32_t vlInc = d / (int32_t)frameCount; 2569 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2570 int32_t vrInc = d / (int32_t)frameCount; 2571 int32_t vl = ((int32_t)mLeftVolShort << 16); 2572 int32_t vr = ((int32_t)mRightVolShort << 16); 2573 do { 2574 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2575 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2576 out += 2; 2577 vl += vlInc; 2578 vr += vrInc; 2579 } while (--frameCount); 2580 } 2581 } else { 2582 if (mChannelCount == 1) { 2583 do { 2584 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2585 out++; 2586 } while (--frameCount); 2587 } else { 2588 do { 2589 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2590 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2591 out += 2; 2592 } while (--frameCount); 2593 } 2594 } 2595 2596 // convert back to unsigned 8 bit after volume calculation 2597 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2598 size_t count = mFrameCount * mChannelCount; 2599 int16_t *src = mMixBuffer; 2600 uint8_t *dst = (uint8_t *)mMixBuffer; 2601 while(count--) { 2602 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2603 } 2604 } 2605 2606 mLeftVolShort = leftVol; 2607 mRightVolShort = rightVol; 2608} 2609 2610bool AudioFlinger::DirectOutputThread::threadLoop() 2611{ 2612 uint32_t mixerStatus = MIXER_IDLE; 2613 sp<Track> trackToRemove; 2614 sp<Track> activeTrack; 2615 nsecs_t standbyTime = systemTime(); 2616 int8_t *curBuf; 2617 size_t mixBufferSize = mFrameCount*mFrameSize; 2618 uint32_t activeSleepTime = activeSleepTimeUs(); 2619 uint32_t idleSleepTime = idleSleepTimeUs(); 2620 uint32_t sleepTime = idleSleepTime; 2621 // use shorter standby delay as on normal output to release 2622 // hardware resources as soon as possible 2623 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2624 2625 acquireWakeLock(); 2626 2627 while (!exitPending()) 2628 { 2629 bool rampVolume; 2630 uint16_t leftVol; 2631 uint16_t rightVol; 2632 Vector< sp<EffectChain> > effectChains; 2633 2634 processConfigEvents(); 2635 2636 mixerStatus = MIXER_IDLE; 2637 2638 { // scope for the mLock 2639 2640 Mutex::Autolock _l(mLock); 2641 2642 if (checkForNewParameters_l()) { 2643 mixBufferSize = mFrameCount*mFrameSize; 2644 activeSleepTime = activeSleepTimeUs(); 2645 idleSleepTime = idleSleepTimeUs(); 2646 standbyDelay = microseconds(activeSleepTime*2); 2647 } 2648 2649 // put audio hardware into standby after short delay 2650 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2651 mSuspended)) { 2652 // wait until we have something to do... 2653 if (!mStandby) { 2654 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2655 mOutput->stream->common.standby(&mOutput->stream->common); 2656 mStandby = true; 2657 mBytesWritten = 0; 2658 } 2659 2660 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2661 // we're about to wait, flush the binder command buffer 2662 IPCThreadState::self()->flushCommands(); 2663 2664 if (exitPending()) break; 2665 2666 releaseWakeLock_l(); 2667 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2668 mWaitWorkCV.wait(mLock); 2669 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2670 acquireWakeLock_l(); 2671 2672 if (!mMasterMute) { 2673 char value[PROPERTY_VALUE_MAX]; 2674 property_get("ro.audio.silent", value, "0"); 2675 if (atoi(value)) { 2676 ALOGD("Silence is golden"); 2677 setMasterMute(true); 2678 } 2679 } 2680 2681 standbyTime = systemTime() + standbyDelay; 2682 sleepTime = idleSleepTime; 2683 continue; 2684 } 2685 } 2686 2687 effectChains = mEffectChains; 2688 2689 // find out which tracks need to be processed 2690 if (mActiveTracks.size() != 0) { 2691 sp<Track> t = mActiveTracks[0].promote(); 2692 if (t == 0) continue; 2693 2694 Track* const track = t.get(); 2695 audio_track_cblk_t* cblk = track->cblk(); 2696 2697 // The first time a track is added we wait 2698 // for all its buffers to be filled before processing it 2699 if (cblk->framesReady() && track->isReady() && 2700 !track->isPaused() && !track->isTerminated()) 2701 { 2702 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2703 2704 if (track->mFillingUpStatus == Track::FS_FILLED) { 2705 track->mFillingUpStatus = Track::FS_ACTIVE; 2706 mLeftVolFloat = mRightVolFloat = 0; 2707 mLeftVolShort = mRightVolShort = 0; 2708 if (track->mState == TrackBase::RESUMING) { 2709 track->mState = TrackBase::ACTIVE; 2710 rampVolume = true; 2711 } 2712 } else if (cblk->server != 0) { 2713 // If the track is stopped before the first frame was mixed, 2714 // do not apply ramp 2715 rampVolume = true; 2716 } 2717 // compute volume for this track 2718 float left, right; 2719 if (track->isMuted() || mMasterMute || track->isPausing() || 2720 mStreamTypes[track->type()].mute) { 2721 left = right = 0; 2722 if (track->isPausing()) { 2723 track->setPaused(); 2724 } 2725 } else { 2726 float typeVolume = mStreamTypes[track->type()].volume; 2727 float v = mMasterVolume * typeVolume; 2728 uint32_t vlr = cblk->volumeLR; 2729 float v_clamped = v * (vlr & 0xFFFF); 2730 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2731 left = v_clamped/MAX_GAIN; 2732 v_clamped = v * (vlr >> 16); 2733 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2734 right = v_clamped/MAX_GAIN; 2735 } 2736 2737 if (left != mLeftVolFloat || right != mRightVolFloat) { 2738 mLeftVolFloat = left; 2739 mRightVolFloat = right; 2740 2741 // If audio HAL implements volume control, 2742 // force software volume to nominal value 2743 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2744 left = 1.0f; 2745 right = 1.0f; 2746 } 2747 2748 // Convert volumes from float to 8.24 2749 uint32_t vl = (uint32_t)(left * (1 << 24)); 2750 uint32_t vr = (uint32_t)(right * (1 << 24)); 2751 2752 // Delegate volume control to effect in track effect chain if needed 2753 // only one effect chain can be present on DirectOutputThread, so if 2754 // there is one, the track is connected to it 2755 if (!effectChains.isEmpty()) { 2756 // Do not ramp volume if volume is controlled by effect 2757 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2758 rampVolume = false; 2759 } 2760 } 2761 2762 // Convert volumes from 8.24 to 4.12 format 2763 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2764 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2765 leftVol = (uint16_t)v_clamped; 2766 v_clamped = (vr + (1 << 11)) >> 12; 2767 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2768 rightVol = (uint16_t)v_clamped; 2769 } else { 2770 leftVol = mLeftVolShort; 2771 rightVol = mRightVolShort; 2772 rampVolume = false; 2773 } 2774 2775 // reset retry count 2776 track->mRetryCount = kMaxTrackRetriesDirect; 2777 activeTrack = t; 2778 mixerStatus = MIXER_TRACKS_READY; 2779 } else { 2780 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2781 if (track->isStopped()) { 2782 track->reset(); 2783 } 2784 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2785 // We have consumed all the buffers of this track. 2786 // Remove it from the list of active tracks. 2787 trackToRemove = track; 2788 } else { 2789 // No buffers for this track. Give it a few chances to 2790 // fill a buffer, then remove it from active list. 2791 if (--(track->mRetryCount) <= 0) { 2792 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2793 trackToRemove = track; 2794 } else { 2795 mixerStatus = MIXER_TRACKS_ENABLED; 2796 } 2797 } 2798 } 2799 } 2800 2801 // remove all the tracks that need to be... 2802 if (CC_UNLIKELY(trackToRemove != 0)) { 2803 mActiveTracks.remove(trackToRemove); 2804 if (!effectChains.isEmpty()) { 2805 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2806 trackToRemove->sessionId()); 2807 effectChains[0]->decActiveTrackCnt(); 2808 } 2809 if (trackToRemove->isTerminated()) { 2810 removeTrack_l(trackToRemove); 2811 } 2812 } 2813 2814 lockEffectChains_l(effectChains); 2815 } 2816 2817 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2818 AudioBufferProvider::Buffer buffer; 2819 size_t frameCount = mFrameCount; 2820 curBuf = (int8_t *)mMixBuffer; 2821 // output audio to hardware 2822 while (frameCount) { 2823 buffer.frameCount = frameCount; 2824 activeTrack->getNextBuffer(&buffer); 2825 if (CC_UNLIKELY(buffer.raw == NULL)) { 2826 memset(curBuf, 0, frameCount * mFrameSize); 2827 break; 2828 } 2829 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2830 frameCount -= buffer.frameCount; 2831 curBuf += buffer.frameCount * mFrameSize; 2832 activeTrack->releaseBuffer(&buffer); 2833 } 2834 sleepTime = 0; 2835 standbyTime = systemTime() + standbyDelay; 2836 } else { 2837 if (sleepTime == 0) { 2838 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2839 sleepTime = activeSleepTime; 2840 } else { 2841 sleepTime = idleSleepTime; 2842 } 2843 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2844 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2845 sleepTime = 0; 2846 } 2847 } 2848 2849 if (mSuspended) { 2850 sleepTime = suspendSleepTimeUs(); 2851 } 2852 // sleepTime == 0 means we must write to audio hardware 2853 if (sleepTime == 0) { 2854 if (mixerStatus == MIXER_TRACKS_READY) { 2855 applyVolume(leftVol, rightVol, rampVolume); 2856 } 2857 for (size_t i = 0; i < effectChains.size(); i ++) { 2858 effectChains[i]->process_l(); 2859 } 2860 unlockEffectChains(effectChains); 2861 2862 mLastWriteTime = systemTime(); 2863 mInWrite = true; 2864 mBytesWritten += mixBufferSize; 2865 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2866 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2867 mNumWrites++; 2868 mInWrite = false; 2869 mStandby = false; 2870 } else { 2871 unlockEffectChains(effectChains); 2872 usleep(sleepTime); 2873 } 2874 2875 // finally let go of removed track, without the lock held 2876 // since we can't guarantee the destructors won't acquire that 2877 // same lock. 2878 trackToRemove.clear(); 2879 activeTrack.clear(); 2880 2881 // Effect chains will be actually deleted here if they were removed from 2882 // mEffectChains list during mixing or effects processing 2883 effectChains.clear(); 2884 } 2885 2886 if (!mStandby) { 2887 mOutput->stream->common.standby(&mOutput->stream->common); 2888 } 2889 2890 releaseWakeLock(); 2891 2892 ALOGV("DirectOutputThread %p exiting", this); 2893 return false; 2894} 2895 2896// getTrackName_l() must be called with ThreadBase::mLock held 2897int AudioFlinger::DirectOutputThread::getTrackName_l() 2898{ 2899 return 0; 2900} 2901 2902// deleteTrackName_l() must be called with ThreadBase::mLock held 2903void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2904{ 2905} 2906 2907// checkForNewParameters_l() must be called with ThreadBase::mLock held 2908bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2909{ 2910 bool reconfig = false; 2911 2912 while (!mNewParameters.isEmpty()) { 2913 status_t status = NO_ERROR; 2914 String8 keyValuePair = mNewParameters[0]; 2915 AudioParameter param = AudioParameter(keyValuePair); 2916 int value; 2917 2918 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2919 // do not accept frame count changes if tracks are open as the track buffer 2920 // size depends on frame count and correct behavior would not be garantied 2921 // if frame count is changed after track creation 2922 if (!mTracks.isEmpty()) { 2923 status = INVALID_OPERATION; 2924 } else { 2925 reconfig = true; 2926 } 2927 } 2928 if (status == NO_ERROR) { 2929 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2930 keyValuePair.string()); 2931 if (!mStandby && status == INVALID_OPERATION) { 2932 mOutput->stream->common.standby(&mOutput->stream->common); 2933 mStandby = true; 2934 mBytesWritten = 0; 2935 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2936 keyValuePair.string()); 2937 } 2938 if (status == NO_ERROR && reconfig) { 2939 readOutputParameters(); 2940 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2941 } 2942 } 2943 2944 mNewParameters.removeAt(0); 2945 2946 mParamStatus = status; 2947 mParamCond.signal(); 2948 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2949 // already timed out waiting for the status and will never signal the condition. 2950 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2951 } 2952 return reconfig; 2953} 2954 2955uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2956{ 2957 uint32_t time; 2958 if (audio_is_linear_pcm(mFormat)) { 2959 time = PlaybackThread::activeSleepTimeUs(); 2960 } else { 2961 time = 10000; 2962 } 2963 return time; 2964} 2965 2966uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2967{ 2968 uint32_t time; 2969 if (audio_is_linear_pcm(mFormat)) { 2970 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2971 } else { 2972 time = 10000; 2973 } 2974 return time; 2975} 2976 2977uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2978{ 2979 uint32_t time; 2980 if (audio_is_linear_pcm(mFormat)) { 2981 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2982 } else { 2983 time = 10000; 2984 } 2985 return time; 2986} 2987 2988 2989// ---------------------------------------------------------------------------- 2990 2991AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2992 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2993{ 2994 mType = ThreadBase::DUPLICATING; 2995 addOutputTrack(mainThread); 2996} 2997 2998AudioFlinger::DuplicatingThread::~DuplicatingThread() 2999{ 3000 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3001 mOutputTracks[i]->destroy(); 3002 } 3003 mOutputTracks.clear(); 3004} 3005 3006bool AudioFlinger::DuplicatingThread::threadLoop() 3007{ 3008 Vector< sp<Track> > tracksToRemove; 3009 uint32_t mixerStatus = MIXER_IDLE; 3010 nsecs_t standbyTime = systemTime(); 3011 size_t mixBufferSize = mFrameCount*mFrameSize; 3012 SortedVector< sp<OutputTrack> > outputTracks; 3013 uint32_t writeFrames = 0; 3014 uint32_t activeSleepTime = activeSleepTimeUs(); 3015 uint32_t idleSleepTime = idleSleepTimeUs(); 3016 uint32_t sleepTime = idleSleepTime; 3017 Vector< sp<EffectChain> > effectChains; 3018 3019 acquireWakeLock(); 3020 3021 while (!exitPending()) 3022 { 3023 processConfigEvents(); 3024 3025 mixerStatus = MIXER_IDLE; 3026 { // scope for the mLock 3027 3028 Mutex::Autolock _l(mLock); 3029 3030 if (checkForNewParameters_l()) { 3031 mixBufferSize = mFrameCount*mFrameSize; 3032 updateWaitTime(); 3033 activeSleepTime = activeSleepTimeUs(); 3034 idleSleepTime = idleSleepTimeUs(); 3035 } 3036 3037 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3038 3039 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3040 outputTracks.add(mOutputTracks[i]); 3041 } 3042 3043 // put audio hardware into standby after short delay 3044 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3045 mSuspended)) { 3046 if (!mStandby) { 3047 for (size_t i = 0; i < outputTracks.size(); i++) { 3048 outputTracks[i]->stop(); 3049 } 3050 mStandby = true; 3051 mBytesWritten = 0; 3052 } 3053 3054 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3055 // we're about to wait, flush the binder command buffer 3056 IPCThreadState::self()->flushCommands(); 3057 outputTracks.clear(); 3058 3059 if (exitPending()) break; 3060 3061 releaseWakeLock_l(); 3062 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3063 mWaitWorkCV.wait(mLock); 3064 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3065 acquireWakeLock_l(); 3066 3067 mPrevMixerStatus = MIXER_IDLE; 3068 if (!mMasterMute) { 3069 char value[PROPERTY_VALUE_MAX]; 3070 property_get("ro.audio.silent", value, "0"); 3071 if (atoi(value)) { 3072 ALOGD("Silence is golden"); 3073 setMasterMute(true); 3074 } 3075 } 3076 3077 standbyTime = systemTime() + kStandbyTimeInNsecs; 3078 sleepTime = idleSleepTime; 3079 continue; 3080 } 3081 } 3082 3083 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3084 3085 // prevent any changes in effect chain list and in each effect chain 3086 // during mixing and effect process as the audio buffers could be deleted 3087 // or modified if an effect is created or deleted 3088 lockEffectChains_l(effectChains); 3089 } 3090 3091 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3092 // mix buffers... 3093 if (outputsReady(outputTracks)) { 3094 mAudioMixer->process(); 3095 } else { 3096 memset(mMixBuffer, 0, mixBufferSize); 3097 } 3098 sleepTime = 0; 3099 writeFrames = mFrameCount; 3100 } else { 3101 if (sleepTime == 0) { 3102 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3103 sleepTime = activeSleepTime; 3104 } else { 3105 sleepTime = idleSleepTime; 3106 } 3107 } else if (mBytesWritten != 0) { 3108 // flush remaining overflow buffers in output tracks 3109 for (size_t i = 0; i < outputTracks.size(); i++) { 3110 if (outputTracks[i]->isActive()) { 3111 sleepTime = 0; 3112 writeFrames = 0; 3113 memset(mMixBuffer, 0, mixBufferSize); 3114 break; 3115 } 3116 } 3117 } 3118 } 3119 3120 if (mSuspended) { 3121 sleepTime = suspendSleepTimeUs(); 3122 } 3123 // sleepTime == 0 means we must write to audio hardware 3124 if (sleepTime == 0) { 3125 for (size_t i = 0; i < effectChains.size(); i ++) { 3126 effectChains[i]->process_l(); 3127 } 3128 // enable changes in effect chain 3129 unlockEffectChains(effectChains); 3130 3131 standbyTime = systemTime() + kStandbyTimeInNsecs; 3132 for (size_t i = 0; i < outputTracks.size(); i++) { 3133 outputTracks[i]->write(mMixBuffer, writeFrames); 3134 } 3135 mStandby = false; 3136 mBytesWritten += mixBufferSize; 3137 } else { 3138 // enable changes in effect chain 3139 unlockEffectChains(effectChains); 3140 usleep(sleepTime); 3141 } 3142 3143 // finally let go of all our tracks, without the lock held 3144 // since we can't guarantee the destructors won't acquire that 3145 // same lock. 3146 tracksToRemove.clear(); 3147 outputTracks.clear(); 3148 3149 // Effect chains will be actually deleted here if they were removed from 3150 // mEffectChains list during mixing or effects processing 3151 effectChains.clear(); 3152 } 3153 3154 releaseWakeLock(); 3155 3156 return false; 3157} 3158 3159void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3160{ 3161 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3162 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3163 this, 3164 mSampleRate, 3165 mFormat, 3166 mChannelMask, 3167 frameCount); 3168 if (outputTrack->cblk() != NULL) { 3169 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3170 mOutputTracks.add(outputTrack); 3171 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3172 updateWaitTime(); 3173 } 3174} 3175 3176void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3177{ 3178 Mutex::Autolock _l(mLock); 3179 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3180 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3181 mOutputTracks[i]->destroy(); 3182 mOutputTracks.removeAt(i); 3183 updateWaitTime(); 3184 return; 3185 } 3186 } 3187 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3188} 3189 3190void AudioFlinger::DuplicatingThread::updateWaitTime() 3191{ 3192 mWaitTimeMs = UINT_MAX; 3193 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3194 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3195 if (strong != NULL) { 3196 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3197 if (waitTimeMs < mWaitTimeMs) { 3198 mWaitTimeMs = waitTimeMs; 3199 } 3200 } 3201 } 3202} 3203 3204 3205bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3206{ 3207 for (size_t i = 0; i < outputTracks.size(); i++) { 3208 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3209 if (thread == 0) { 3210 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3211 return false; 3212 } 3213 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3214 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3215 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3216 return false; 3217 } 3218 } 3219 return true; 3220} 3221 3222uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3223{ 3224 return (mWaitTimeMs * 1000) / 2; 3225} 3226 3227// ---------------------------------------------------------------------------- 3228 3229// TrackBase constructor must be called with AudioFlinger::mLock held 3230AudioFlinger::ThreadBase::TrackBase::TrackBase( 3231 const wp<ThreadBase>& thread, 3232 const sp<Client>& client, 3233 uint32_t sampleRate, 3234 audio_format_t format, 3235 uint32_t channelMask, 3236 int frameCount, 3237 uint32_t flags, 3238 const sp<IMemory>& sharedBuffer, 3239 int sessionId) 3240 : RefBase(), 3241 mThread(thread), 3242 mClient(client), 3243 mCblk(0), 3244 mFrameCount(0), 3245 mState(IDLE), 3246 mClientTid(-1), 3247 mFormat(format), 3248 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3249 mSessionId(sessionId) 3250{ 3251 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3252 3253 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3254 size_t size = sizeof(audio_track_cblk_t); 3255 uint8_t channelCount = popcount(channelMask); 3256 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3257 if (sharedBuffer == 0) { 3258 size += bufferSize; 3259 } 3260 3261 if (client != NULL) { 3262 mCblkMemory = client->heap()->allocate(size); 3263 if (mCblkMemory != 0) { 3264 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3265 if (mCblk) { // construct the shared structure in-place. 3266 new(mCblk) audio_track_cblk_t(); 3267 // clear all buffers 3268 mCblk->frameCount = frameCount; 3269 mCblk->sampleRate = sampleRate; 3270 mChannelCount = channelCount; 3271 mChannelMask = channelMask; 3272 if (sharedBuffer == 0) { 3273 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3274 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3275 // Force underrun condition to avoid false underrun callback until first data is 3276 // written to buffer (other flags are cleared) 3277 mCblk->flags = CBLK_UNDERRUN_ON; 3278 } else { 3279 mBuffer = sharedBuffer->pointer(); 3280 } 3281 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3282 } 3283 } else { 3284 ALOGE("not enough memory for AudioTrack size=%u", size); 3285 client->heap()->dump("AudioTrack"); 3286 return; 3287 } 3288 } else { 3289 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3290 // construct the shared structure in-place. 3291 new(mCblk) audio_track_cblk_t(); 3292 // clear all buffers 3293 mCblk->frameCount = frameCount; 3294 mCblk->sampleRate = sampleRate; 3295 mChannelCount = channelCount; 3296 mChannelMask = channelMask; 3297 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3298 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3299 // Force underrun condition to avoid false underrun callback until first data is 3300 // written to buffer (other flags are cleared) 3301 mCblk->flags = CBLK_UNDERRUN_ON; 3302 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3303 } 3304} 3305 3306AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3307{ 3308 if (mCblk) { 3309 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3310 if (mClient == NULL) { 3311 delete mCblk; 3312 } 3313 } 3314 mCblkMemory.clear(); // and free the shared memory 3315 if (mClient != NULL) { 3316 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3317 mClient.clear(); 3318 } 3319} 3320 3321void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3322{ 3323 buffer->raw = NULL; 3324 mFrameCount = buffer->frameCount; 3325 step(); 3326 buffer->frameCount = 0; 3327} 3328 3329bool AudioFlinger::ThreadBase::TrackBase::step() { 3330 bool result; 3331 audio_track_cblk_t* cblk = this->cblk(); 3332 3333 result = cblk->stepServer(mFrameCount); 3334 if (!result) { 3335 ALOGV("stepServer failed acquiring cblk mutex"); 3336 mFlags |= STEPSERVER_FAILED; 3337 } 3338 return result; 3339} 3340 3341void AudioFlinger::ThreadBase::TrackBase::reset() { 3342 audio_track_cblk_t* cblk = this->cblk(); 3343 3344 cblk->user = 0; 3345 cblk->server = 0; 3346 cblk->userBase = 0; 3347 cblk->serverBase = 0; 3348 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3349 ALOGV("TrackBase::reset"); 3350} 3351 3352sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3353{ 3354 return mCblkMemory; 3355} 3356 3357int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3358 return (int)mCblk->sampleRate; 3359} 3360 3361int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3362 return (const int)mChannelCount; 3363} 3364 3365uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3366 return mChannelMask; 3367} 3368 3369void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3370 audio_track_cblk_t* cblk = this->cblk(); 3371 size_t frameSize = cblk->frameSize; 3372 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3373 int8_t *bufferEnd = bufferStart + frames * frameSize; 3374 3375 // Check validity of returned pointer in case the track control block would have been corrupted. 3376 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3377 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3378 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3379 server %d, serverBase %d, user %d, userBase %d", 3380 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3381 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3382 return 0; 3383 } 3384 3385 return bufferStart; 3386} 3387 3388// ---------------------------------------------------------------------------- 3389 3390// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3391AudioFlinger::PlaybackThread::Track::Track( 3392 const wp<ThreadBase>& thread, 3393 const sp<Client>& client, 3394 audio_stream_type_t streamType, 3395 uint32_t sampleRate, 3396 audio_format_t format, 3397 uint32_t channelMask, 3398 int frameCount, 3399 const sp<IMemory>& sharedBuffer, 3400 int sessionId) 3401 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3402 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3403 mAuxEffectId(0), mHasVolumeController(false) 3404{ 3405 if (mCblk != NULL) { 3406 sp<ThreadBase> baseThread = thread.promote(); 3407 if (baseThread != 0) { 3408 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3409 mName = playbackThread->getTrackName_l(); 3410 mMainBuffer = playbackThread->mixBuffer(); 3411 } 3412 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3413 if (mName < 0) { 3414 ALOGE("no more track names available"); 3415 } 3416 mStreamType = streamType; 3417 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3418 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3419 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3420 } 3421} 3422 3423AudioFlinger::PlaybackThread::Track::~Track() 3424{ 3425 ALOGV("PlaybackThread::Track destructor"); 3426 sp<ThreadBase> thread = mThread.promote(); 3427 if (thread != 0) { 3428 Mutex::Autolock _l(thread->mLock); 3429 mState = TERMINATED; 3430 } 3431} 3432 3433void AudioFlinger::PlaybackThread::Track::destroy() 3434{ 3435 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3436 // by removing it from mTracks vector, so there is a risk that this Tracks's 3437 // desctructor is called. As the destructor needs to lock mLock, 3438 // we must acquire a strong reference on this Track before locking mLock 3439 // here so that the destructor is called only when exiting this function. 3440 // On the other hand, as long as Track::destroy() is only called by 3441 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3442 // this Track with its member mTrack. 3443 sp<Track> keep(this); 3444 { // scope for mLock 3445 sp<ThreadBase> thread = mThread.promote(); 3446 if (thread != 0) { 3447 if (!isOutputTrack()) { 3448 if (mState == ACTIVE || mState == RESUMING) { 3449 AudioSystem::stopOutput(thread->id(), 3450 (audio_stream_type_t)mStreamType, 3451 mSessionId); 3452 3453 // to track the speaker usage 3454 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3455 } 3456 AudioSystem::releaseOutput(thread->id()); 3457 } 3458 Mutex::Autolock _l(thread->mLock); 3459 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3460 playbackThread->destroyTrack_l(this); 3461 } 3462 } 3463} 3464 3465void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3466{ 3467 uint32_t vlr = mCblk->volumeLR; 3468 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", 3469 mName - AudioMixer::TRACK0, 3470 (mClient == NULL) ? getpid() : mClient->pid(), 3471 mStreamType, 3472 mFormat, 3473 mChannelMask, 3474 mSessionId, 3475 mFrameCount, 3476 mState, 3477 mMute, 3478 mFillingUpStatus, 3479 mCblk->sampleRate, 3480 vlr & 0xFFFF, 3481 vlr >> 16, 3482 mCblk->server, 3483 mCblk->user, 3484 (int)mMainBuffer, 3485 (int)mAuxBuffer); 3486} 3487 3488status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3489{ 3490 audio_track_cblk_t* cblk = this->cblk(); 3491 uint32_t framesReady; 3492 uint32_t framesReq = buffer->frameCount; 3493 3494 // Check if last stepServer failed, try to step now 3495 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3496 if (!step()) goto getNextBuffer_exit; 3497 ALOGV("stepServer recovered"); 3498 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3499 } 3500 3501 framesReady = cblk->framesReady(); 3502 3503 if (CC_LIKELY(framesReady)) { 3504 uint32_t s = cblk->server; 3505 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3506 3507 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3508 if (framesReq > framesReady) { 3509 framesReq = framesReady; 3510 } 3511 if (s + framesReq > bufferEnd) { 3512 framesReq = bufferEnd - s; 3513 } 3514 3515 buffer->raw = getBuffer(s, framesReq); 3516 if (buffer->raw == NULL) goto getNextBuffer_exit; 3517 3518 buffer->frameCount = framesReq; 3519 return NO_ERROR; 3520 } 3521 3522getNextBuffer_exit: 3523 buffer->raw = NULL; 3524 buffer->frameCount = 0; 3525 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3526 return NOT_ENOUGH_DATA; 3527} 3528 3529bool AudioFlinger::PlaybackThread::Track::isReady() const { 3530 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3531 3532 if (mCblk->framesReady() >= mCblk->frameCount || 3533 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3534 mFillingUpStatus = FS_FILLED; 3535 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3536 return true; 3537 } 3538 return false; 3539} 3540 3541status_t AudioFlinger::PlaybackThread::Track::start() 3542{ 3543 status_t status = NO_ERROR; 3544 ALOGV("start(%d), calling thread %d session %d", 3545 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3546 sp<ThreadBase> thread = mThread.promote(); 3547 if (thread != 0) { 3548 Mutex::Autolock _l(thread->mLock); 3549 int state = mState; 3550 // here the track could be either new, or restarted 3551 // in both cases "unstop" the track 3552 if (mState == PAUSED) { 3553 mState = TrackBase::RESUMING; 3554 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3555 } else { 3556 mState = TrackBase::ACTIVE; 3557 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3558 } 3559 3560 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3561 thread->mLock.unlock(); 3562 status = AudioSystem::startOutput(thread->id(), 3563 (audio_stream_type_t)mStreamType, 3564 mSessionId); 3565 thread->mLock.lock(); 3566 3567 // to track the speaker usage 3568 if (status == NO_ERROR) { 3569 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3570 } 3571 } 3572 if (status == NO_ERROR) { 3573 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3574 playbackThread->addTrack_l(this); 3575 } else { 3576 mState = state; 3577 } 3578 } else { 3579 status = BAD_VALUE; 3580 } 3581 return status; 3582} 3583 3584void AudioFlinger::PlaybackThread::Track::stop() 3585{ 3586 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3587 sp<ThreadBase> thread = mThread.promote(); 3588 if (thread != 0) { 3589 Mutex::Autolock _l(thread->mLock); 3590 int state = mState; 3591 if (mState > STOPPED) { 3592 mState = STOPPED; 3593 // If the track is not active (PAUSED and buffers full), flush buffers 3594 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3595 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3596 reset(); 3597 } 3598 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3599 } 3600 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3601 thread->mLock.unlock(); 3602 AudioSystem::stopOutput(thread->id(), 3603 (audio_stream_type_t)mStreamType, 3604 mSessionId); 3605 thread->mLock.lock(); 3606 3607 // to track the speaker usage 3608 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3609 } 3610 } 3611} 3612 3613void AudioFlinger::PlaybackThread::Track::pause() 3614{ 3615 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3616 sp<ThreadBase> thread = mThread.promote(); 3617 if (thread != 0) { 3618 Mutex::Autolock _l(thread->mLock); 3619 if (mState == ACTIVE || mState == RESUMING) { 3620 mState = PAUSING; 3621 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3622 if (!isOutputTrack()) { 3623 thread->mLock.unlock(); 3624 AudioSystem::stopOutput(thread->id(), 3625 (audio_stream_type_t)mStreamType, 3626 mSessionId); 3627 thread->mLock.lock(); 3628 3629 // to track the speaker usage 3630 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3631 } 3632 } 3633 } 3634} 3635 3636void AudioFlinger::PlaybackThread::Track::flush() 3637{ 3638 ALOGV("flush(%d)", mName); 3639 sp<ThreadBase> thread = mThread.promote(); 3640 if (thread != 0) { 3641 Mutex::Autolock _l(thread->mLock); 3642 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3643 return; 3644 } 3645 // No point remaining in PAUSED state after a flush => go to 3646 // STOPPED state 3647 mState = STOPPED; 3648 3649 // do not reset the track if it is still in the process of being stopped or paused. 3650 // this will be done by prepareTracks_l() when the track is stopped. 3651 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3652 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3653 reset(); 3654 } 3655 } 3656} 3657 3658void AudioFlinger::PlaybackThread::Track::reset() 3659{ 3660 // Do not reset twice to avoid discarding data written just after a flush and before 3661 // the audioflinger thread detects the track is stopped. 3662 if (!mResetDone) { 3663 TrackBase::reset(); 3664 // Force underrun condition to avoid false underrun callback until first data is 3665 // written to buffer 3666 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3667 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3668 mFillingUpStatus = FS_FILLING; 3669 mResetDone = true; 3670 } 3671} 3672 3673void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3674{ 3675 mMute = muted; 3676} 3677 3678status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3679{ 3680 status_t status = DEAD_OBJECT; 3681 sp<ThreadBase> thread = mThread.promote(); 3682 if (thread != 0) { 3683 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3684 status = playbackThread->attachAuxEffect(this, EffectId); 3685 } 3686 return status; 3687} 3688 3689void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3690{ 3691 mAuxEffectId = EffectId; 3692 mAuxBuffer = buffer; 3693} 3694 3695// ---------------------------------------------------------------------------- 3696 3697// RecordTrack constructor must be called with AudioFlinger::mLock held 3698AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3699 const wp<ThreadBase>& thread, 3700 const sp<Client>& client, 3701 uint32_t sampleRate, 3702 audio_format_t format, 3703 uint32_t channelMask, 3704 int frameCount, 3705 uint32_t flags, 3706 int sessionId) 3707 : TrackBase(thread, client, sampleRate, format, 3708 channelMask, frameCount, flags, 0, sessionId), 3709 mOverflow(false) 3710{ 3711 if (mCblk != NULL) { 3712 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3713 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3714 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3715 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3716 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3717 } else { 3718 mCblk->frameSize = sizeof(int8_t); 3719 } 3720 } 3721} 3722 3723AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3724{ 3725 sp<ThreadBase> thread = mThread.promote(); 3726 if (thread != 0) { 3727 AudioSystem::releaseInput(thread->id()); 3728 } 3729} 3730 3731status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3732{ 3733 audio_track_cblk_t* cblk = this->cblk(); 3734 uint32_t framesAvail; 3735 uint32_t framesReq = buffer->frameCount; 3736 3737 // Check if last stepServer failed, try to step now 3738 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3739 if (!step()) goto getNextBuffer_exit; 3740 ALOGV("stepServer recovered"); 3741 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3742 } 3743 3744 framesAvail = cblk->framesAvailable_l(); 3745 3746 if (CC_LIKELY(framesAvail)) { 3747 uint32_t s = cblk->server; 3748 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3749 3750 if (framesReq > framesAvail) { 3751 framesReq = framesAvail; 3752 } 3753 if (s + framesReq > bufferEnd) { 3754 framesReq = bufferEnd - s; 3755 } 3756 3757 buffer->raw = getBuffer(s, framesReq); 3758 if (buffer->raw == NULL) goto getNextBuffer_exit; 3759 3760 buffer->frameCount = framesReq; 3761 return NO_ERROR; 3762 } 3763 3764getNextBuffer_exit: 3765 buffer->raw = NULL; 3766 buffer->frameCount = 0; 3767 return NOT_ENOUGH_DATA; 3768} 3769 3770status_t AudioFlinger::RecordThread::RecordTrack::start() 3771{ 3772 sp<ThreadBase> thread = mThread.promote(); 3773 if (thread != 0) { 3774 RecordThread *recordThread = (RecordThread *)thread.get(); 3775 return recordThread->start(this); 3776 } else { 3777 return BAD_VALUE; 3778 } 3779} 3780 3781void AudioFlinger::RecordThread::RecordTrack::stop() 3782{ 3783 sp<ThreadBase> thread = mThread.promote(); 3784 if (thread != 0) { 3785 RecordThread *recordThread = (RecordThread *)thread.get(); 3786 recordThread->stop(this); 3787 TrackBase::reset(); 3788 // Force overerrun condition to avoid false overrun callback until first data is 3789 // read from buffer 3790 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3791 } 3792} 3793 3794void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3795{ 3796 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3797 (mClient == NULL) ? getpid() : mClient->pid(), 3798 mFormat, 3799 mChannelMask, 3800 mSessionId, 3801 mFrameCount, 3802 mState, 3803 mCblk->sampleRate, 3804 mCblk->server, 3805 mCblk->user); 3806} 3807 3808 3809// ---------------------------------------------------------------------------- 3810 3811AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3812 const wp<ThreadBase>& thread, 3813 DuplicatingThread *sourceThread, 3814 uint32_t sampleRate, 3815 audio_format_t format, 3816 uint32_t channelMask, 3817 int frameCount) 3818 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3819 mActive(false), mSourceThread(sourceThread) 3820{ 3821 3822 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3823 if (mCblk != NULL) { 3824 mCblk->flags |= CBLK_DIRECTION_OUT; 3825 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3826 mCblk->volumeLR = (MAX_GAIN_INT << 16) | MAX_GAIN_INT; 3827 mOutBuffer.frameCount = 0; 3828 playbackThread->mTracks.add(this); 3829 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3830 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3831 mCblk, mBuffer, mCblk->buffers, 3832 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3833 } else { 3834 ALOGW("Error creating output track on thread %p", playbackThread); 3835 } 3836} 3837 3838AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3839{ 3840 clearBufferQueue(); 3841} 3842 3843status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3844{ 3845 status_t status = Track::start(); 3846 if (status != NO_ERROR) { 3847 return status; 3848 } 3849 3850 mActive = true; 3851 mRetryCount = 127; 3852 return status; 3853} 3854 3855void AudioFlinger::PlaybackThread::OutputTrack::stop() 3856{ 3857 Track::stop(); 3858 clearBufferQueue(); 3859 mOutBuffer.frameCount = 0; 3860 mActive = false; 3861} 3862 3863bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3864{ 3865 Buffer *pInBuffer; 3866 Buffer inBuffer; 3867 uint32_t channelCount = mChannelCount; 3868 bool outputBufferFull = false; 3869 inBuffer.frameCount = frames; 3870 inBuffer.i16 = data; 3871 3872 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3873 3874 if (!mActive && frames != 0) { 3875 start(); 3876 sp<ThreadBase> thread = mThread.promote(); 3877 if (thread != 0) { 3878 MixerThread *mixerThread = (MixerThread *)thread.get(); 3879 if (mCblk->frameCount > frames){ 3880 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3881 uint32_t startFrames = (mCblk->frameCount - frames); 3882 pInBuffer = new Buffer; 3883 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3884 pInBuffer->frameCount = startFrames; 3885 pInBuffer->i16 = pInBuffer->mBuffer; 3886 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3887 mBufferQueue.add(pInBuffer); 3888 } else { 3889 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3890 } 3891 } 3892 } 3893 } 3894 3895 while (waitTimeLeftMs) { 3896 // First write pending buffers, then new data 3897 if (mBufferQueue.size()) { 3898 pInBuffer = mBufferQueue.itemAt(0); 3899 } else { 3900 pInBuffer = &inBuffer; 3901 } 3902 3903 if (pInBuffer->frameCount == 0) { 3904 break; 3905 } 3906 3907 if (mOutBuffer.frameCount == 0) { 3908 mOutBuffer.frameCount = pInBuffer->frameCount; 3909 nsecs_t startTime = systemTime(); 3910 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3911 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3912 outputBufferFull = true; 3913 break; 3914 } 3915 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3916 if (waitTimeLeftMs >= waitTimeMs) { 3917 waitTimeLeftMs -= waitTimeMs; 3918 } else { 3919 waitTimeLeftMs = 0; 3920 } 3921 } 3922 3923 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3924 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3925 mCblk->stepUser(outFrames); 3926 pInBuffer->frameCount -= outFrames; 3927 pInBuffer->i16 += outFrames * channelCount; 3928 mOutBuffer.frameCount -= outFrames; 3929 mOutBuffer.i16 += outFrames * channelCount; 3930 3931 if (pInBuffer->frameCount == 0) { 3932 if (mBufferQueue.size()) { 3933 mBufferQueue.removeAt(0); 3934 delete [] pInBuffer->mBuffer; 3935 delete pInBuffer; 3936 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3937 } else { 3938 break; 3939 } 3940 } 3941 } 3942 3943 // If we could not write all frames, allocate a buffer and queue it for next time. 3944 if (inBuffer.frameCount) { 3945 sp<ThreadBase> thread = mThread.promote(); 3946 if (thread != 0 && !thread->standby()) { 3947 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3948 pInBuffer = new Buffer; 3949 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3950 pInBuffer->frameCount = inBuffer.frameCount; 3951 pInBuffer->i16 = pInBuffer->mBuffer; 3952 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3953 mBufferQueue.add(pInBuffer); 3954 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3955 } else { 3956 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3957 } 3958 } 3959 } 3960 3961 // Calling write() with a 0 length buffer, means that no more data will be written: 3962 // If no more buffers are pending, fill output track buffer to make sure it is started 3963 // by output mixer. 3964 if (frames == 0 && mBufferQueue.size() == 0) { 3965 if (mCblk->user < mCblk->frameCount) { 3966 frames = mCblk->frameCount - mCblk->user; 3967 pInBuffer = new Buffer; 3968 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3969 pInBuffer->frameCount = frames; 3970 pInBuffer->i16 = pInBuffer->mBuffer; 3971 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3972 mBufferQueue.add(pInBuffer); 3973 } else if (mActive) { 3974 stop(); 3975 } 3976 } 3977 3978 return outputBufferFull; 3979} 3980 3981status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3982{ 3983 int active; 3984 status_t result; 3985 audio_track_cblk_t* cblk = mCblk; 3986 uint32_t framesReq = buffer->frameCount; 3987 3988// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3989 buffer->frameCount = 0; 3990 3991 uint32_t framesAvail = cblk->framesAvailable(); 3992 3993 3994 if (framesAvail == 0) { 3995 Mutex::Autolock _l(cblk->lock); 3996 goto start_loop_here; 3997 while (framesAvail == 0) { 3998 active = mActive; 3999 if (CC_UNLIKELY(!active)) { 4000 ALOGV("Not active and NO_MORE_BUFFERS"); 4001 return NO_MORE_BUFFERS; 4002 } 4003 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4004 if (result != NO_ERROR) { 4005 return NO_MORE_BUFFERS; 4006 } 4007 // read the server count again 4008 start_loop_here: 4009 framesAvail = cblk->framesAvailable_l(); 4010 } 4011 } 4012 4013// if (framesAvail < framesReq) { 4014// return NO_MORE_BUFFERS; 4015// } 4016 4017 if (framesReq > framesAvail) { 4018 framesReq = framesAvail; 4019 } 4020 4021 uint32_t u = cblk->user; 4022 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4023 4024 if (u + framesReq > bufferEnd) { 4025 framesReq = bufferEnd - u; 4026 } 4027 4028 buffer->frameCount = framesReq; 4029 buffer->raw = (void *)cblk->buffer(u); 4030 return NO_ERROR; 4031} 4032 4033 4034void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4035{ 4036 size_t size = mBufferQueue.size(); 4037 Buffer *pBuffer; 4038 4039 for (size_t i = 0; i < size; i++) { 4040 pBuffer = mBufferQueue.itemAt(i); 4041 delete [] pBuffer->mBuffer; 4042 delete pBuffer; 4043 } 4044 mBufferQueue.clear(); 4045} 4046 4047// ---------------------------------------------------------------------------- 4048 4049AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4050 : RefBase(), 4051 mAudioFlinger(audioFlinger), 4052 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4053 mPid(pid) 4054{ 4055 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4056} 4057 4058// Client destructor must be called with AudioFlinger::mLock held 4059AudioFlinger::Client::~Client() 4060{ 4061 mAudioFlinger->removeClient_l(mPid); 4062} 4063 4064const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4065{ 4066 return mMemoryDealer; 4067} 4068 4069// ---------------------------------------------------------------------------- 4070 4071AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4072 const sp<IAudioFlingerClient>& client, 4073 pid_t pid) 4074 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4075{ 4076} 4077 4078AudioFlinger::NotificationClient::~NotificationClient() 4079{ 4080 mClient.clear(); 4081} 4082 4083void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4084{ 4085 sp<NotificationClient> keep(this); 4086 { 4087 mAudioFlinger->removeNotificationClient(mPid); 4088 } 4089} 4090 4091// ---------------------------------------------------------------------------- 4092 4093AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4094 : BnAudioTrack(), 4095 mTrack(track) 4096{ 4097} 4098 4099AudioFlinger::TrackHandle::~TrackHandle() { 4100 // just stop the track on deletion, associated resources 4101 // will be freed from the main thread once all pending buffers have 4102 // been played. Unless it's not in the active track list, in which 4103 // case we free everything now... 4104 mTrack->destroy(); 4105} 4106 4107status_t AudioFlinger::TrackHandle::start() { 4108 return mTrack->start(); 4109} 4110 4111void AudioFlinger::TrackHandle::stop() { 4112 mTrack->stop(); 4113} 4114 4115void AudioFlinger::TrackHandle::flush() { 4116 mTrack->flush(); 4117} 4118 4119void AudioFlinger::TrackHandle::mute(bool e) { 4120 mTrack->mute(e); 4121} 4122 4123void AudioFlinger::TrackHandle::pause() { 4124 mTrack->pause(); 4125} 4126 4127sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4128 return mTrack->getCblk(); 4129} 4130 4131status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4132{ 4133 return mTrack->attachAuxEffect(EffectId); 4134} 4135 4136status_t AudioFlinger::TrackHandle::onTransact( 4137 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4138{ 4139 return BnAudioTrack::onTransact(code, data, reply, flags); 4140} 4141 4142// ---------------------------------------------------------------------------- 4143 4144sp<IAudioRecord> AudioFlinger::openRecord( 4145 pid_t pid, 4146 int input, 4147 uint32_t sampleRate, 4148 audio_format_t format, 4149 uint32_t channelMask, 4150 int frameCount, 4151 uint32_t flags, 4152 int *sessionId, 4153 status_t *status) 4154{ 4155 sp<RecordThread::RecordTrack> recordTrack; 4156 sp<RecordHandle> recordHandle; 4157 sp<Client> client; 4158 wp<Client> wclient; 4159 status_t lStatus; 4160 RecordThread *thread; 4161 size_t inFrameCount; 4162 int lSessionId; 4163 4164 // check calling permissions 4165 if (!recordingAllowed()) { 4166 lStatus = PERMISSION_DENIED; 4167 goto Exit; 4168 } 4169 4170 // add client to list 4171 { // scope for mLock 4172 Mutex::Autolock _l(mLock); 4173 thread = checkRecordThread_l(input); 4174 if (thread == NULL) { 4175 lStatus = BAD_VALUE; 4176 goto Exit; 4177 } 4178 4179 wclient = mClients.valueFor(pid); 4180 if (wclient != NULL) { 4181 client = wclient.promote(); 4182 } else { 4183 client = new Client(this, pid); 4184 mClients.add(pid, client); 4185 } 4186 4187 // If no audio session id is provided, create one here 4188 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4189 lSessionId = *sessionId; 4190 } else { 4191 lSessionId = nextUniqueId(); 4192 if (sessionId != NULL) { 4193 *sessionId = lSessionId; 4194 } 4195 } 4196 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4197 recordTrack = thread->createRecordTrack_l(client, 4198 sampleRate, 4199 format, 4200 channelMask, 4201 frameCount, 4202 flags, 4203 lSessionId, 4204 &lStatus); 4205 } 4206 if (lStatus != NO_ERROR) { 4207 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4208 // destructor is called by the TrackBase destructor with mLock held 4209 client.clear(); 4210 recordTrack.clear(); 4211 goto Exit; 4212 } 4213 4214 // return to handle to client 4215 recordHandle = new RecordHandle(recordTrack); 4216 lStatus = NO_ERROR; 4217 4218Exit: 4219 if (status) { 4220 *status = lStatus; 4221 } 4222 return recordHandle; 4223} 4224 4225// ---------------------------------------------------------------------------- 4226 4227AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4228 : BnAudioRecord(), 4229 mRecordTrack(recordTrack) 4230{ 4231} 4232 4233AudioFlinger::RecordHandle::~RecordHandle() { 4234 stop(); 4235} 4236 4237status_t AudioFlinger::RecordHandle::start() { 4238 ALOGV("RecordHandle::start()"); 4239 return mRecordTrack->start(); 4240} 4241 4242void AudioFlinger::RecordHandle::stop() { 4243 ALOGV("RecordHandle::stop()"); 4244 mRecordTrack->stop(); 4245} 4246 4247sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4248 return mRecordTrack->getCblk(); 4249} 4250 4251status_t AudioFlinger::RecordHandle::onTransact( 4252 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4253{ 4254 return BnAudioRecord::onTransact(code, data, reply, flags); 4255} 4256 4257// ---------------------------------------------------------------------------- 4258 4259AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4260 AudioStreamIn *input, 4261 uint32_t sampleRate, 4262 uint32_t channels, 4263 int id, 4264 uint32_t device) : 4265 ThreadBase(audioFlinger, id, device), 4266 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4267{ 4268 mType = ThreadBase::RECORD; 4269 4270 snprintf(mName, kNameLength, "AudioIn_%d", id); 4271 4272 mReqChannelCount = popcount(channels); 4273 mReqSampleRate = sampleRate; 4274 readInputParameters(); 4275} 4276 4277 4278AudioFlinger::RecordThread::~RecordThread() 4279{ 4280 delete[] mRsmpInBuffer; 4281 if (mResampler != NULL) { 4282 delete mResampler; 4283 delete[] mRsmpOutBuffer; 4284 } 4285} 4286 4287void AudioFlinger::RecordThread::onFirstRef() 4288{ 4289 run(mName, PRIORITY_URGENT_AUDIO); 4290} 4291 4292status_t AudioFlinger::RecordThread::readyToRun() 4293{ 4294 status_t status = initCheck(); 4295 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4296 return status; 4297} 4298 4299bool AudioFlinger::RecordThread::threadLoop() 4300{ 4301 AudioBufferProvider::Buffer buffer; 4302 sp<RecordTrack> activeTrack; 4303 Vector< sp<EffectChain> > effectChains; 4304 4305 nsecs_t lastWarning = 0; 4306 4307 acquireWakeLock(); 4308 4309 // start recording 4310 while (!exitPending()) { 4311 4312 processConfigEvents(); 4313 4314 { // scope for mLock 4315 Mutex::Autolock _l(mLock); 4316 checkForNewParameters_l(); 4317 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4318 if (!mStandby) { 4319 mInput->stream->common.standby(&mInput->stream->common); 4320 mStandby = true; 4321 } 4322 4323 if (exitPending()) break; 4324 4325 releaseWakeLock_l(); 4326 ALOGV("RecordThread: loop stopping"); 4327 // go to sleep 4328 mWaitWorkCV.wait(mLock); 4329 ALOGV("RecordThread: loop starting"); 4330 acquireWakeLock_l(); 4331 continue; 4332 } 4333 if (mActiveTrack != 0) { 4334 if (mActiveTrack->mState == TrackBase::PAUSING) { 4335 if (!mStandby) { 4336 mInput->stream->common.standby(&mInput->stream->common); 4337 mStandby = true; 4338 } 4339 mActiveTrack.clear(); 4340 mStartStopCond.broadcast(); 4341 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4342 if (mReqChannelCount != mActiveTrack->channelCount()) { 4343 mActiveTrack.clear(); 4344 mStartStopCond.broadcast(); 4345 } else if (mBytesRead != 0) { 4346 // record start succeeds only if first read from audio input 4347 // succeeds 4348 if (mBytesRead > 0) { 4349 mActiveTrack->mState = TrackBase::ACTIVE; 4350 } else { 4351 mActiveTrack.clear(); 4352 } 4353 mStartStopCond.broadcast(); 4354 } 4355 mStandby = false; 4356 } 4357 } 4358 lockEffectChains_l(effectChains); 4359 } 4360 4361 if (mActiveTrack != 0) { 4362 if (mActiveTrack->mState != TrackBase::ACTIVE && 4363 mActiveTrack->mState != TrackBase::RESUMING) { 4364 unlockEffectChains(effectChains); 4365 usleep(kRecordThreadSleepUs); 4366 continue; 4367 } 4368 for (size_t i = 0; i < effectChains.size(); i ++) { 4369 effectChains[i]->process_l(); 4370 } 4371 4372 buffer.frameCount = mFrameCount; 4373 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4374 size_t framesOut = buffer.frameCount; 4375 if (mResampler == NULL) { 4376 // no resampling 4377 while (framesOut) { 4378 size_t framesIn = mFrameCount - mRsmpInIndex; 4379 if (framesIn) { 4380 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4381 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4382 if (framesIn > framesOut) 4383 framesIn = framesOut; 4384 mRsmpInIndex += framesIn; 4385 framesOut -= framesIn; 4386 if ((int)mChannelCount == mReqChannelCount || 4387 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4388 memcpy(dst, src, framesIn * mFrameSize); 4389 } else { 4390 int16_t *src16 = (int16_t *)src; 4391 int16_t *dst16 = (int16_t *)dst; 4392 if (mChannelCount == 1) { 4393 while (framesIn--) { 4394 *dst16++ = *src16; 4395 *dst16++ = *src16++; 4396 } 4397 } else { 4398 while (framesIn--) { 4399 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4400 src16 += 2; 4401 } 4402 } 4403 } 4404 } 4405 if (framesOut && mFrameCount == mRsmpInIndex) { 4406 if (framesOut == mFrameCount && 4407 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4408 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4409 framesOut = 0; 4410 } else { 4411 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4412 mRsmpInIndex = 0; 4413 } 4414 if (mBytesRead < 0) { 4415 ALOGE("Error reading audio input"); 4416 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4417 // Force input into standby so that it tries to 4418 // recover at next read attempt 4419 mInput->stream->common.standby(&mInput->stream->common); 4420 usleep(kRecordThreadSleepUs); 4421 } 4422 mRsmpInIndex = mFrameCount; 4423 framesOut = 0; 4424 buffer.frameCount = 0; 4425 } 4426 } 4427 } 4428 } else { 4429 // resampling 4430 4431 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4432 // alter output frame count as if we were expecting stereo samples 4433 if (mChannelCount == 1 && mReqChannelCount == 1) { 4434 framesOut >>= 1; 4435 } 4436 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4437 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4438 // are 32 bit aligned which should be always true. 4439 if (mChannelCount == 2 && mReqChannelCount == 1) { 4440 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4441 // the resampler always outputs stereo samples: do post stereo to mono conversion 4442 int16_t *src = (int16_t *)mRsmpOutBuffer; 4443 int16_t *dst = buffer.i16; 4444 while (framesOut--) { 4445 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4446 src += 2; 4447 } 4448 } else { 4449 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4450 } 4451 4452 } 4453 mActiveTrack->releaseBuffer(&buffer); 4454 mActiveTrack->overflow(); 4455 } 4456 // client isn't retrieving buffers fast enough 4457 else { 4458 if (!mActiveTrack->setOverflow()) { 4459 nsecs_t now = systemTime(); 4460 if ((now - lastWarning) > kWarningThrottleNs) { 4461 ALOGW("RecordThread: buffer overflow"); 4462 lastWarning = now; 4463 } 4464 } 4465 // Release the processor for a while before asking for a new buffer. 4466 // This will give the application more chance to read from the buffer and 4467 // clear the overflow. 4468 usleep(kRecordThreadSleepUs); 4469 } 4470 } 4471 // enable changes in effect chain 4472 unlockEffectChains(effectChains); 4473 effectChains.clear(); 4474 } 4475 4476 if (!mStandby) { 4477 mInput->stream->common.standby(&mInput->stream->common); 4478 } 4479 mActiveTrack.clear(); 4480 4481 mStartStopCond.broadcast(); 4482 4483 releaseWakeLock(); 4484 4485 ALOGV("RecordThread %p exiting", this); 4486 return false; 4487} 4488 4489 4490sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4491 const sp<AudioFlinger::Client>& client, 4492 uint32_t sampleRate, 4493 audio_format_t format, 4494 int channelMask, 4495 int frameCount, 4496 uint32_t flags, 4497 int sessionId, 4498 status_t *status) 4499{ 4500 sp<RecordTrack> track; 4501 status_t lStatus; 4502 4503 lStatus = initCheck(); 4504 if (lStatus != NO_ERROR) { 4505 ALOGE("Audio driver not initialized."); 4506 goto Exit; 4507 } 4508 4509 { // scope for mLock 4510 Mutex::Autolock _l(mLock); 4511 4512 track = new RecordTrack(this, client, sampleRate, 4513 format, channelMask, frameCount, flags, sessionId); 4514 4515 if (track->getCblk() == NULL) { 4516 lStatus = NO_MEMORY; 4517 goto Exit; 4518 } 4519 4520 mTrack = track.get(); 4521 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4522 bool suspend = audio_is_bluetooth_sco_device( 4523 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4524 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4525 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4526 } 4527 lStatus = NO_ERROR; 4528 4529Exit: 4530 if (status) { 4531 *status = lStatus; 4532 } 4533 return track; 4534} 4535 4536status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4537{ 4538 ALOGV("RecordThread::start"); 4539 sp <ThreadBase> strongMe = this; 4540 status_t status = NO_ERROR; 4541 { 4542 AutoMutex lock(mLock); 4543 if (mActiveTrack != 0) { 4544 if (recordTrack != mActiveTrack.get()) { 4545 status = -EBUSY; 4546 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4547 mActiveTrack->mState = TrackBase::ACTIVE; 4548 } 4549 return status; 4550 } 4551 4552 recordTrack->mState = TrackBase::IDLE; 4553 mActiveTrack = recordTrack; 4554 mLock.unlock(); 4555 status_t status = AudioSystem::startInput(mId); 4556 mLock.lock(); 4557 if (status != NO_ERROR) { 4558 mActiveTrack.clear(); 4559 return status; 4560 } 4561 mRsmpInIndex = mFrameCount; 4562 mBytesRead = 0; 4563 if (mResampler != NULL) { 4564 mResampler->reset(); 4565 } 4566 mActiveTrack->mState = TrackBase::RESUMING; 4567 // signal thread to start 4568 ALOGV("Signal record thread"); 4569 mWaitWorkCV.signal(); 4570 // do not wait for mStartStopCond if exiting 4571 if (mExiting) { 4572 mActiveTrack.clear(); 4573 status = INVALID_OPERATION; 4574 goto startError; 4575 } 4576 mStartStopCond.wait(mLock); 4577 if (mActiveTrack == 0) { 4578 ALOGV("Record failed to start"); 4579 status = BAD_VALUE; 4580 goto startError; 4581 } 4582 ALOGV("Record started OK"); 4583 return status; 4584 } 4585startError: 4586 AudioSystem::stopInput(mId); 4587 return status; 4588} 4589 4590void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4591 ALOGV("RecordThread::stop"); 4592 sp <ThreadBase> strongMe = this; 4593 { 4594 AutoMutex lock(mLock); 4595 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4596 mActiveTrack->mState = TrackBase::PAUSING; 4597 // do not wait for mStartStopCond if exiting 4598 if (mExiting) { 4599 return; 4600 } 4601 mStartStopCond.wait(mLock); 4602 // if we have been restarted, recordTrack == mActiveTrack.get() here 4603 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4604 mLock.unlock(); 4605 AudioSystem::stopInput(mId); 4606 mLock.lock(); 4607 ALOGV("Record stopped OK"); 4608 } 4609 } 4610 } 4611} 4612 4613status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4614{ 4615 const size_t SIZE = 256; 4616 char buffer[SIZE]; 4617 String8 result; 4618 pid_t pid = 0; 4619 4620 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4621 result.append(buffer); 4622 4623 if (mActiveTrack != 0) { 4624 result.append("Active Track:\n"); 4625 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4626 mActiveTrack->dump(buffer, SIZE); 4627 result.append(buffer); 4628 4629 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4630 result.append(buffer); 4631 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4632 result.append(buffer); 4633 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4634 result.append(buffer); 4635 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4636 result.append(buffer); 4637 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4638 result.append(buffer); 4639 4640 4641 } else { 4642 result.append("No record client\n"); 4643 } 4644 write(fd, result.string(), result.size()); 4645 4646 dumpBase(fd, args); 4647 dumpEffectChains(fd, args); 4648 4649 return NO_ERROR; 4650} 4651 4652status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4653{ 4654 size_t framesReq = buffer->frameCount; 4655 size_t framesReady = mFrameCount - mRsmpInIndex; 4656 int channelCount; 4657 4658 if (framesReady == 0) { 4659 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4660 if (mBytesRead < 0) { 4661 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4662 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4663 // Force input into standby so that it tries to 4664 // recover at next read attempt 4665 mInput->stream->common.standby(&mInput->stream->common); 4666 usleep(kRecordThreadSleepUs); 4667 } 4668 buffer->raw = NULL; 4669 buffer->frameCount = 0; 4670 return NOT_ENOUGH_DATA; 4671 } 4672 mRsmpInIndex = 0; 4673 framesReady = mFrameCount; 4674 } 4675 4676 if (framesReq > framesReady) { 4677 framesReq = framesReady; 4678 } 4679 4680 if (mChannelCount == 1 && mReqChannelCount == 2) { 4681 channelCount = 1; 4682 } else { 4683 channelCount = 2; 4684 } 4685 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4686 buffer->frameCount = framesReq; 4687 return NO_ERROR; 4688} 4689 4690void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4691{ 4692 mRsmpInIndex += buffer->frameCount; 4693 buffer->frameCount = 0; 4694} 4695 4696bool AudioFlinger::RecordThread::checkForNewParameters_l() 4697{ 4698 bool reconfig = false; 4699 4700 while (!mNewParameters.isEmpty()) { 4701 status_t status = NO_ERROR; 4702 String8 keyValuePair = mNewParameters[0]; 4703 AudioParameter param = AudioParameter(keyValuePair); 4704 int value; 4705 audio_format_t reqFormat = mFormat; 4706 int reqSamplingRate = mReqSampleRate; 4707 int reqChannelCount = mReqChannelCount; 4708 4709 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4710 reqSamplingRate = value; 4711 reconfig = true; 4712 } 4713 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4714 reqFormat = (audio_format_t) value; 4715 reconfig = true; 4716 } 4717 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4718 reqChannelCount = popcount(value); 4719 reconfig = true; 4720 } 4721 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4722 // do not accept frame count changes if tracks are open as the track buffer 4723 // size depends on frame count and correct behavior would not be garantied 4724 // if frame count is changed after track creation 4725 if (mActiveTrack != 0) { 4726 status = INVALID_OPERATION; 4727 } else { 4728 reconfig = true; 4729 } 4730 } 4731 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4732 // forward device change to effects that have requested to be 4733 // aware of attached audio device. 4734 for (size_t i = 0; i < mEffectChains.size(); i++) { 4735 mEffectChains[i]->setDevice_l(value); 4736 } 4737 // store input device and output device but do not forward output device to audio HAL. 4738 // Note that status is ignored by the caller for output device 4739 // (see AudioFlinger::setParameters() 4740 if (value & AUDIO_DEVICE_OUT_ALL) { 4741 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4742 status = BAD_VALUE; 4743 } else { 4744 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4745 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4746 if (mTrack != NULL) { 4747 bool suspend = audio_is_bluetooth_sco_device( 4748 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4749 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4750 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4751 } 4752 } 4753 mDevice |= (uint32_t)value; 4754 } 4755 if (status == NO_ERROR) { 4756 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4757 if (status == INVALID_OPERATION) { 4758 mInput->stream->common.standby(&mInput->stream->common); 4759 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4760 } 4761 if (reconfig) { 4762 if (status == BAD_VALUE && 4763 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4764 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4765 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4766 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4767 (reqChannelCount < 3)) { 4768 status = NO_ERROR; 4769 } 4770 if (status == NO_ERROR) { 4771 readInputParameters(); 4772 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4773 } 4774 } 4775 } 4776 4777 mNewParameters.removeAt(0); 4778 4779 mParamStatus = status; 4780 mParamCond.signal(); 4781 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4782 // already timed out waiting for the status and will never signal the condition. 4783 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4784 } 4785 return reconfig; 4786} 4787 4788String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4789{ 4790 char *s; 4791 String8 out_s8 = String8(); 4792 4793 Mutex::Autolock _l(mLock); 4794 if (initCheck() != NO_ERROR) { 4795 return out_s8; 4796 } 4797 4798 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4799 out_s8 = String8(s); 4800 free(s); 4801 return out_s8; 4802} 4803 4804void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4805 AudioSystem::OutputDescriptor desc; 4806 void *param2 = 0; 4807 4808 switch (event) { 4809 case AudioSystem::INPUT_OPENED: 4810 case AudioSystem::INPUT_CONFIG_CHANGED: 4811 desc.channels = mChannelMask; 4812 desc.samplingRate = mSampleRate; 4813 desc.format = mFormat; 4814 desc.frameCount = mFrameCount; 4815 desc.latency = 0; 4816 param2 = &desc; 4817 break; 4818 4819 case AudioSystem::INPUT_CLOSED: 4820 default: 4821 break; 4822 } 4823 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4824} 4825 4826void AudioFlinger::RecordThread::readInputParameters() 4827{ 4828 if (mRsmpInBuffer) delete mRsmpInBuffer; 4829 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4830 if (mResampler) delete mResampler; 4831 mResampler = NULL; 4832 4833 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4834 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4835 mChannelCount = (uint16_t)popcount(mChannelMask); 4836 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4837 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4838 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4839 mFrameCount = mInputBytes / mFrameSize; 4840 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4841 4842 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4843 { 4844 int channelCount; 4845 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4846 // stereo to mono post process as the resampler always outputs stereo. 4847 if (mChannelCount == 1 && mReqChannelCount == 2) { 4848 channelCount = 1; 4849 } else { 4850 channelCount = 2; 4851 } 4852 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4853 mResampler->setSampleRate(mSampleRate); 4854 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4855 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4856 4857 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4858 if (mChannelCount == 1 && mReqChannelCount == 1) { 4859 mFrameCount >>= 1; 4860 } 4861 4862 } 4863 mRsmpInIndex = mFrameCount; 4864} 4865 4866unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4867{ 4868 Mutex::Autolock _l(mLock); 4869 if (initCheck() != NO_ERROR) { 4870 return 0; 4871 } 4872 4873 return mInput->stream->get_input_frames_lost(mInput->stream); 4874} 4875 4876uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4877{ 4878 Mutex::Autolock _l(mLock); 4879 uint32_t result = 0; 4880 if (getEffectChain_l(sessionId) != 0) { 4881 result = EFFECT_SESSION; 4882 } 4883 4884 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4885 result |= TRACK_SESSION; 4886 } 4887 4888 return result; 4889} 4890 4891AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4892{ 4893 Mutex::Autolock _l(mLock); 4894 return mTrack; 4895} 4896 4897AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() 4898{ 4899 Mutex::Autolock _l(mLock); 4900 return mInput; 4901} 4902 4903AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4904{ 4905 Mutex::Autolock _l(mLock); 4906 AudioStreamIn *input = mInput; 4907 mInput = NULL; 4908 return input; 4909} 4910 4911// this method must always be called either with ThreadBase mLock held or inside the thread loop 4912audio_stream_t* AudioFlinger::RecordThread::stream() 4913{ 4914 if (mInput == NULL) { 4915 return NULL; 4916 } 4917 return &mInput->stream->common; 4918} 4919 4920 4921// ---------------------------------------------------------------------------- 4922 4923int AudioFlinger::openOutput(uint32_t *pDevices, 4924 uint32_t *pSamplingRate, 4925 audio_format_t *pFormat, 4926 uint32_t *pChannels, 4927 uint32_t *pLatencyMs, 4928 uint32_t flags) 4929{ 4930 status_t status; 4931 PlaybackThread *thread = NULL; 4932 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4933 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4934 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4935 uint32_t channels = pChannels ? *pChannels : 0; 4936 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4937 audio_stream_out_t *outStream; 4938 audio_hw_device_t *outHwDev; 4939 4940 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4941 pDevices ? *pDevices : 0, 4942 samplingRate, 4943 format, 4944 channels, 4945 flags); 4946 4947 if (pDevices == NULL || *pDevices == 0) { 4948 return 0; 4949 } 4950 4951 Mutex::Autolock _l(mLock); 4952 4953 outHwDev = findSuitableHwDev_l(*pDevices); 4954 if (outHwDev == NULL) 4955 return 0; 4956 4957 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4958 &channels, &samplingRate, &outStream); 4959 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4960 outStream, 4961 samplingRate, 4962 format, 4963 channels, 4964 status); 4965 4966 mHardwareStatus = AUDIO_HW_IDLE; 4967 if (outStream != NULL) { 4968 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4969 int id = nextUniqueId(); 4970 4971 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4972 (format != AUDIO_FORMAT_PCM_16_BIT) || 4973 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4974 thread = new DirectOutputThread(this, output, id, *pDevices); 4975 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4976 } else { 4977 thread = new MixerThread(this, output, id, *pDevices); 4978 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4979 } 4980 mPlaybackThreads.add(id, thread); 4981 4982 if (pSamplingRate) *pSamplingRate = samplingRate; 4983 if (pFormat) *pFormat = format; 4984 if (pChannels) *pChannels = channels; 4985 if (pLatencyMs) *pLatencyMs = thread->latency(); 4986 4987 // notify client processes of the new output creation 4988 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4989 return id; 4990 } 4991 4992 return 0; 4993} 4994 4995int AudioFlinger::openDuplicateOutput(int output1, int output2) 4996{ 4997 Mutex::Autolock _l(mLock); 4998 MixerThread *thread1 = checkMixerThread_l(output1); 4999 MixerThread *thread2 = checkMixerThread_l(output2); 5000 5001 if (thread1 == NULL || thread2 == NULL) { 5002 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5003 return 0; 5004 } 5005 5006 int id = nextUniqueId(); 5007 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5008 thread->addOutputTrack(thread2); 5009 mPlaybackThreads.add(id, thread); 5010 // notify client processes of the new output creation 5011 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5012 return id; 5013} 5014 5015status_t AudioFlinger::closeOutput(int output) 5016{ 5017 // keep strong reference on the playback thread so that 5018 // it is not destroyed while exit() is executed 5019 sp <PlaybackThread> thread; 5020 { 5021 Mutex::Autolock _l(mLock); 5022 thread = checkPlaybackThread_l(output); 5023 if (thread == NULL) { 5024 return BAD_VALUE; 5025 } 5026 5027 ALOGV("closeOutput() %d", output); 5028 5029 if (thread->type() == ThreadBase::MIXER) { 5030 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5031 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5032 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5033 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5034 } 5035 } 5036 } 5037 void *param2 = 0; 5038 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5039 mPlaybackThreads.removeItem(output); 5040 } 5041 thread->exit(); 5042 5043 if (thread->type() != ThreadBase::DUPLICATING) { 5044 AudioStreamOut *out = thread->clearOutput(); 5045 // from now on thread->mOutput is NULL 5046 out->hwDev->close_output_stream(out->hwDev, out->stream); 5047 delete out; 5048 } 5049 return NO_ERROR; 5050} 5051 5052status_t AudioFlinger::suspendOutput(int output) 5053{ 5054 Mutex::Autolock _l(mLock); 5055 PlaybackThread *thread = checkPlaybackThread_l(output); 5056 5057 if (thread == NULL) { 5058 return BAD_VALUE; 5059 } 5060 5061 ALOGV("suspendOutput() %d", output); 5062 thread->suspend(); 5063 5064 return NO_ERROR; 5065} 5066 5067status_t AudioFlinger::restoreOutput(int output) 5068{ 5069 Mutex::Autolock _l(mLock); 5070 PlaybackThread *thread = checkPlaybackThread_l(output); 5071 5072 if (thread == NULL) { 5073 return BAD_VALUE; 5074 } 5075 5076 ALOGV("restoreOutput() %d", output); 5077 5078 thread->restore(); 5079 5080 return NO_ERROR; 5081} 5082 5083int AudioFlinger::openInput(uint32_t *pDevices, 5084 uint32_t *pSamplingRate, 5085 audio_format_t *pFormat, 5086 uint32_t *pChannels, 5087 uint32_t acoustics) 5088{ 5089 status_t status; 5090 RecordThread *thread = NULL; 5091 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5092 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5093 uint32_t channels = pChannels ? *pChannels : 0; 5094 uint32_t reqSamplingRate = samplingRate; 5095 audio_format_t reqFormat = format; 5096 uint32_t reqChannels = channels; 5097 audio_stream_in_t *inStream; 5098 audio_hw_device_t *inHwDev; 5099 5100 if (pDevices == NULL || *pDevices == 0) { 5101 return 0; 5102 } 5103 5104 Mutex::Autolock _l(mLock); 5105 5106 inHwDev = findSuitableHwDev_l(*pDevices); 5107 if (inHwDev == NULL) 5108 return 0; 5109 5110 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5111 &channels, &samplingRate, 5112 (audio_in_acoustics_t)acoustics, 5113 &inStream); 5114 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5115 inStream, 5116 samplingRate, 5117 format, 5118 channels, 5119 acoustics, 5120 status); 5121 5122 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5123 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5124 // or stereo to mono conversions on 16 bit PCM inputs. 5125 if (inStream == NULL && status == BAD_VALUE && 5126 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5127 (samplingRate <= 2 * reqSamplingRate) && 5128 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5129 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5130 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5131 &channels, &samplingRate, 5132 (audio_in_acoustics_t)acoustics, 5133 &inStream); 5134 } 5135 5136 if (inStream != NULL) { 5137 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5138 5139 int id = nextUniqueId(); 5140 // Start record thread 5141 // RecorThread require both input and output device indication to forward to audio 5142 // pre processing modules 5143 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5144 thread = new RecordThread(this, 5145 input, 5146 reqSamplingRate, 5147 reqChannels, 5148 id, 5149 device); 5150 mRecordThreads.add(id, thread); 5151 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5152 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5153 if (pFormat) *pFormat = format; 5154 if (pChannels) *pChannels = reqChannels; 5155 5156 input->stream->common.standby(&input->stream->common); 5157 5158 // notify client processes of the new input creation 5159 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5160 return id; 5161 } 5162 5163 return 0; 5164} 5165 5166status_t AudioFlinger::closeInput(int input) 5167{ 5168 // keep strong reference on the record thread so that 5169 // it is not destroyed while exit() is executed 5170 sp <RecordThread> thread; 5171 { 5172 Mutex::Autolock _l(mLock); 5173 thread = checkRecordThread_l(input); 5174 if (thread == NULL) { 5175 return BAD_VALUE; 5176 } 5177 5178 ALOGV("closeInput() %d", input); 5179 void *param2 = 0; 5180 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5181 mRecordThreads.removeItem(input); 5182 } 5183 thread->exit(); 5184 5185 AudioStreamIn *in = thread->clearInput(); 5186 // from now on thread->mInput is NULL 5187 in->hwDev->close_input_stream(in->hwDev, in->stream); 5188 delete in; 5189 5190 return NO_ERROR; 5191} 5192 5193status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5194{ 5195 Mutex::Autolock _l(mLock); 5196 MixerThread *dstThread = checkMixerThread_l(output); 5197 if (dstThread == NULL) { 5198 ALOGW("setStreamOutput() bad output id %d", output); 5199 return BAD_VALUE; 5200 } 5201 5202 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5203 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5204 5205 dstThread->setStreamValid(stream, true); 5206 5207 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5208 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5209 if (thread != dstThread && 5210 thread->type() != ThreadBase::DIRECT) { 5211 MixerThread *srcThread = (MixerThread *)thread; 5212 srcThread->setStreamValid(stream, false); 5213 srcThread->invalidateTracks(stream); 5214 } 5215 } 5216 5217 return NO_ERROR; 5218} 5219 5220 5221int AudioFlinger::newAudioSessionId() 5222{ 5223 return nextUniqueId(); 5224} 5225 5226void AudioFlinger::acquireAudioSessionId(int audioSession) 5227{ 5228 Mutex::Autolock _l(mLock); 5229 int caller = IPCThreadState::self()->getCallingPid(); 5230 ALOGV("acquiring %d from %d", audioSession, caller); 5231 int num = mAudioSessionRefs.size(); 5232 for (int i = 0; i< num; i++) { 5233 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5234 if (ref->sessionid == audioSession && ref->pid == caller) { 5235 ref->cnt++; 5236 ALOGV(" incremented refcount to %d", ref->cnt); 5237 return; 5238 } 5239 } 5240 AudioSessionRef *ref = new AudioSessionRef(); 5241 ref->sessionid = audioSession; 5242 ref->pid = caller; 5243 ref->cnt = 1; 5244 mAudioSessionRefs.push(ref); 5245 ALOGV(" added new entry for %d", ref->sessionid); 5246} 5247 5248void AudioFlinger::releaseAudioSessionId(int audioSession) 5249{ 5250 Mutex::Autolock _l(mLock); 5251 int caller = IPCThreadState::self()->getCallingPid(); 5252 ALOGV("releasing %d from %d", audioSession, caller); 5253 int num = mAudioSessionRefs.size(); 5254 for (int i = 0; i< num; i++) { 5255 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5256 if (ref->sessionid == audioSession && ref->pid == caller) { 5257 ref->cnt--; 5258 ALOGV(" decremented refcount to %d", ref->cnt); 5259 if (ref->cnt == 0) { 5260 mAudioSessionRefs.removeAt(i); 5261 delete ref; 5262 purgeStaleEffects_l(); 5263 } 5264 return; 5265 } 5266 } 5267 ALOGW("session id %d not found for pid %d", audioSession, caller); 5268} 5269 5270void AudioFlinger::purgeStaleEffects_l() { 5271 5272 ALOGV("purging stale effects"); 5273 5274 Vector< sp<EffectChain> > chains; 5275 5276 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5277 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5278 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5279 sp<EffectChain> ec = t->mEffectChains[j]; 5280 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5281 chains.push(ec); 5282 } 5283 } 5284 } 5285 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5286 sp<RecordThread> t = mRecordThreads.valueAt(i); 5287 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5288 sp<EffectChain> ec = t->mEffectChains[j]; 5289 chains.push(ec); 5290 } 5291 } 5292 5293 for (size_t i = 0; i < chains.size(); i++) { 5294 sp<EffectChain> ec = chains[i]; 5295 int sessionid = ec->sessionId(); 5296 sp<ThreadBase> t = ec->mThread.promote(); 5297 if (t == 0) { 5298 continue; 5299 } 5300 size_t numsessionrefs = mAudioSessionRefs.size(); 5301 bool found = false; 5302 for (size_t k = 0; k < numsessionrefs; k++) { 5303 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5304 if (ref->sessionid == sessionid) { 5305 ALOGV(" session %d still exists for %d with %d refs", 5306 sessionid, ref->pid, ref->cnt); 5307 found = true; 5308 break; 5309 } 5310 } 5311 if (!found) { 5312 // remove all effects from the chain 5313 while (ec->mEffects.size()) { 5314 sp<EffectModule> effect = ec->mEffects[0]; 5315 effect->unPin(); 5316 Mutex::Autolock _l (t->mLock); 5317 t->removeEffect_l(effect); 5318 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5319 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5320 if (handle != 0) { 5321 handle->mEffect.clear(); 5322 if (handle->mHasControl && handle->mEnabled) { 5323 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5324 } 5325 } 5326 } 5327 AudioSystem::unregisterEffect(effect->id()); 5328 } 5329 } 5330 } 5331 return; 5332} 5333 5334// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5335AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5336{ 5337 PlaybackThread *thread = NULL; 5338 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5339 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5340 } 5341 return thread; 5342} 5343 5344// checkMixerThread_l() must be called with AudioFlinger::mLock held 5345AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5346{ 5347 PlaybackThread *thread = checkPlaybackThread_l(output); 5348 if (thread != NULL) { 5349 if (thread->type() == ThreadBase::DIRECT) { 5350 thread = NULL; 5351 } 5352 } 5353 return (MixerThread *)thread; 5354} 5355 5356// checkRecordThread_l() must be called with AudioFlinger::mLock held 5357AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5358{ 5359 RecordThread *thread = NULL; 5360 if (mRecordThreads.indexOfKey(input) >= 0) { 5361 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5362 } 5363 return thread; 5364} 5365 5366uint32_t AudioFlinger::nextUniqueId() 5367{ 5368 return android_atomic_inc(&mNextUniqueId); 5369} 5370 5371AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5372{ 5373 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5374 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5375 AudioStreamOut *output = thread->getOutput(); 5376 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5377 return thread; 5378 } 5379 } 5380 return NULL; 5381} 5382 5383uint32_t AudioFlinger::primaryOutputDevice_l() 5384{ 5385 PlaybackThread *thread = primaryPlaybackThread_l(); 5386 5387 if (thread == NULL) { 5388 return 0; 5389 } 5390 5391 return thread->device(); 5392} 5393 5394 5395// ---------------------------------------------------------------------------- 5396// Effect management 5397// ---------------------------------------------------------------------------- 5398 5399 5400status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5401{ 5402 Mutex::Autolock _l(mLock); 5403 return EffectQueryNumberEffects(numEffects); 5404} 5405 5406status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5407{ 5408 Mutex::Autolock _l(mLock); 5409 return EffectQueryEffect(index, descriptor); 5410} 5411 5412status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5413{ 5414 Mutex::Autolock _l(mLock); 5415 return EffectGetDescriptor(pUuid, descriptor); 5416} 5417 5418 5419sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5420 effect_descriptor_t *pDesc, 5421 const sp<IEffectClient>& effectClient, 5422 int32_t priority, 5423 int io, 5424 int sessionId, 5425 status_t *status, 5426 int *id, 5427 int *enabled) 5428{ 5429 status_t lStatus = NO_ERROR; 5430 sp<EffectHandle> handle; 5431 effect_descriptor_t desc; 5432 sp<Client> client; 5433 wp<Client> wclient; 5434 5435 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5436 pid, effectClient.get(), priority, sessionId, io); 5437 5438 if (pDesc == NULL) { 5439 lStatus = BAD_VALUE; 5440 goto Exit; 5441 } 5442 5443 // check audio settings permission for global effects 5444 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5445 lStatus = PERMISSION_DENIED; 5446 goto Exit; 5447 } 5448 5449 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5450 // that can only be created by audio policy manager (running in same process) 5451 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5452 lStatus = PERMISSION_DENIED; 5453 goto Exit; 5454 } 5455 5456 if (io == 0) { 5457 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5458 // output must be specified by AudioPolicyManager when using session 5459 // AUDIO_SESSION_OUTPUT_STAGE 5460 lStatus = BAD_VALUE; 5461 goto Exit; 5462 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5463 // if the output returned by getOutputForEffect() is removed before we lock the 5464 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5465 // and we will exit safely 5466 io = AudioSystem::getOutputForEffect(&desc); 5467 } 5468 } 5469 5470 { 5471 Mutex::Autolock _l(mLock); 5472 5473 5474 if (!EffectIsNullUuid(&pDesc->uuid)) { 5475 // if uuid is specified, request effect descriptor 5476 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5477 if (lStatus < 0) { 5478 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5479 goto Exit; 5480 } 5481 } else { 5482 // if uuid is not specified, look for an available implementation 5483 // of the required type in effect factory 5484 if (EffectIsNullUuid(&pDesc->type)) { 5485 ALOGW("createEffect() no effect type"); 5486 lStatus = BAD_VALUE; 5487 goto Exit; 5488 } 5489 uint32_t numEffects = 0; 5490 effect_descriptor_t d; 5491 d.flags = 0; // prevent compiler warning 5492 bool found = false; 5493 5494 lStatus = EffectQueryNumberEffects(&numEffects); 5495 if (lStatus < 0) { 5496 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5497 goto Exit; 5498 } 5499 for (uint32_t i = 0; i < numEffects; i++) { 5500 lStatus = EffectQueryEffect(i, &desc); 5501 if (lStatus < 0) { 5502 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5503 continue; 5504 } 5505 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5506 // If matching type found save effect descriptor. If the session is 5507 // 0 and the effect is not auxiliary, continue enumeration in case 5508 // an auxiliary version of this effect type is available 5509 found = true; 5510 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5511 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5512 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5513 break; 5514 } 5515 } 5516 } 5517 if (!found) { 5518 lStatus = BAD_VALUE; 5519 ALOGW("createEffect() effect not found"); 5520 goto Exit; 5521 } 5522 // For same effect type, chose auxiliary version over insert version if 5523 // connect to output mix (Compliance to OpenSL ES) 5524 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5525 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5526 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5527 } 5528 } 5529 5530 // Do not allow auxiliary effects on a session different from 0 (output mix) 5531 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5532 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5533 lStatus = INVALID_OPERATION; 5534 goto Exit; 5535 } 5536 5537 // check recording permission for visualizer 5538 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5539 !recordingAllowed()) { 5540 lStatus = PERMISSION_DENIED; 5541 goto Exit; 5542 } 5543 5544 // return effect descriptor 5545 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5546 5547 // If output is not specified try to find a matching audio session ID in one of the 5548 // output threads. 5549 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5550 // because of code checking output when entering the function. 5551 // Note: io is never 0 when creating an effect on an input 5552 if (io == 0) { 5553 // look for the thread where the specified audio session is present 5554 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5555 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5556 io = mPlaybackThreads.keyAt(i); 5557 break; 5558 } 5559 } 5560 if (io == 0) { 5561 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5562 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5563 io = mRecordThreads.keyAt(i); 5564 break; 5565 } 5566 } 5567 } 5568 // If no output thread contains the requested session ID, default to 5569 // first output. The effect chain will be moved to the correct output 5570 // thread when a track with the same session ID is created 5571 if (io == 0 && mPlaybackThreads.size()) { 5572 io = mPlaybackThreads.keyAt(0); 5573 } 5574 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5575 } 5576 ThreadBase *thread = checkRecordThread_l(io); 5577 if (thread == NULL) { 5578 thread = checkPlaybackThread_l(io); 5579 if (thread == NULL) { 5580 ALOGE("createEffect() unknown output thread"); 5581 lStatus = BAD_VALUE; 5582 goto Exit; 5583 } 5584 } 5585 5586 wclient = mClients.valueFor(pid); 5587 5588 if (wclient != NULL) { 5589 client = wclient.promote(); 5590 } else { 5591 client = new Client(this, pid); 5592 mClients.add(pid, client); 5593 } 5594 5595 // create effect on selected output thread 5596 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5597 &desc, enabled, &lStatus); 5598 if (handle != 0 && id != NULL) { 5599 *id = handle->id(); 5600 } 5601 } 5602 5603Exit: 5604 if(status) { 5605 *status = lStatus; 5606 } 5607 return handle; 5608} 5609 5610status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5611{ 5612 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5613 sessionId, srcOutput, dstOutput); 5614 Mutex::Autolock _l(mLock); 5615 if (srcOutput == dstOutput) { 5616 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5617 return NO_ERROR; 5618 } 5619 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5620 if (srcThread == NULL) { 5621 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5622 return BAD_VALUE; 5623 } 5624 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5625 if (dstThread == NULL) { 5626 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5627 return BAD_VALUE; 5628 } 5629 5630 Mutex::Autolock _dl(dstThread->mLock); 5631 Mutex::Autolock _sl(srcThread->mLock); 5632 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5633 5634 return NO_ERROR; 5635} 5636 5637// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5638status_t AudioFlinger::moveEffectChain_l(int sessionId, 5639 AudioFlinger::PlaybackThread *srcThread, 5640 AudioFlinger::PlaybackThread *dstThread, 5641 bool reRegister) 5642{ 5643 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5644 sessionId, srcThread, dstThread); 5645 5646 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5647 if (chain == 0) { 5648 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5649 sessionId, srcThread); 5650 return INVALID_OPERATION; 5651 } 5652 5653 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5654 // so that a new chain is created with correct parameters when first effect is added. This is 5655 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5656 // removed. 5657 srcThread->removeEffectChain_l(chain); 5658 5659 // transfer all effects one by one so that new effect chain is created on new thread with 5660 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5661 int dstOutput = dstThread->id(); 5662 sp<EffectChain> dstChain; 5663 uint32_t strategy = 0; // prevent compiler warning 5664 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5665 while (effect != 0) { 5666 srcThread->removeEffect_l(effect); 5667 dstThread->addEffect_l(effect); 5668 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5669 if (effect->state() == EffectModule::ACTIVE || 5670 effect->state() == EffectModule::STOPPING) { 5671 effect->start(); 5672 } 5673 // if the move request is not received from audio policy manager, the effect must be 5674 // re-registered with the new strategy and output 5675 if (dstChain == 0) { 5676 dstChain = effect->chain().promote(); 5677 if (dstChain == 0) { 5678 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5679 srcThread->addEffect_l(effect); 5680 return NO_INIT; 5681 } 5682 strategy = dstChain->strategy(); 5683 } 5684 if (reRegister) { 5685 AudioSystem::unregisterEffect(effect->id()); 5686 AudioSystem::registerEffect(&effect->desc(), 5687 dstOutput, 5688 strategy, 5689 sessionId, 5690 effect->id()); 5691 } 5692 effect = chain->getEffectFromId_l(0); 5693 } 5694 5695 return NO_ERROR; 5696} 5697 5698 5699// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5700sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5701 const sp<AudioFlinger::Client>& client, 5702 const sp<IEffectClient>& effectClient, 5703 int32_t priority, 5704 int sessionId, 5705 effect_descriptor_t *desc, 5706 int *enabled, 5707 status_t *status 5708 ) 5709{ 5710 sp<EffectModule> effect; 5711 sp<EffectHandle> handle; 5712 status_t lStatus; 5713 sp<EffectChain> chain; 5714 bool chainCreated = false; 5715 bool effectCreated = false; 5716 bool effectRegistered = false; 5717 5718 lStatus = initCheck(); 5719 if (lStatus != NO_ERROR) { 5720 ALOGW("createEffect_l() Audio driver not initialized."); 5721 goto Exit; 5722 } 5723 5724 // Do not allow effects with session ID 0 on direct output or duplicating threads 5725 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5726 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5727 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5728 desc->name, sessionId); 5729 lStatus = BAD_VALUE; 5730 goto Exit; 5731 } 5732 // Only Pre processor effects are allowed on input threads and only on input threads 5733 if ((mType == RECORD && 5734 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5735 (mType != RECORD && 5736 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5737 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5738 desc->name, desc->flags, mType); 5739 lStatus = BAD_VALUE; 5740 goto Exit; 5741 } 5742 5743 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5744 5745 { // scope for mLock 5746 Mutex::Autolock _l(mLock); 5747 5748 // check for existing effect chain with the requested audio session 5749 chain = getEffectChain_l(sessionId); 5750 if (chain == 0) { 5751 // create a new chain for this session 5752 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5753 chain = new EffectChain(this, sessionId); 5754 addEffectChain_l(chain); 5755 chain->setStrategy(getStrategyForSession_l(sessionId)); 5756 chainCreated = true; 5757 } else { 5758 effect = chain->getEffectFromDesc_l(desc); 5759 } 5760 5761 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5762 5763 if (effect == 0) { 5764 int id = mAudioFlinger->nextUniqueId(); 5765 // Check CPU and memory usage 5766 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5767 if (lStatus != NO_ERROR) { 5768 goto Exit; 5769 } 5770 effectRegistered = true; 5771 // create a new effect module if none present in the chain 5772 effect = new EffectModule(this, chain, desc, id, sessionId); 5773 lStatus = effect->status(); 5774 if (lStatus != NO_ERROR) { 5775 goto Exit; 5776 } 5777 lStatus = chain->addEffect_l(effect); 5778 if (lStatus != NO_ERROR) { 5779 goto Exit; 5780 } 5781 effectCreated = true; 5782 5783 effect->setDevice(mDevice); 5784 effect->setMode(mAudioFlinger->getMode()); 5785 } 5786 // create effect handle and connect it to effect module 5787 handle = new EffectHandle(effect, client, effectClient, priority); 5788 lStatus = effect->addHandle(handle); 5789 if (enabled) { 5790 *enabled = (int)effect->isEnabled(); 5791 } 5792 } 5793 5794Exit: 5795 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5796 Mutex::Autolock _l(mLock); 5797 if (effectCreated) { 5798 chain->removeEffect_l(effect); 5799 } 5800 if (effectRegistered) { 5801 AudioSystem::unregisterEffect(effect->id()); 5802 } 5803 if (chainCreated) { 5804 removeEffectChain_l(chain); 5805 } 5806 handle.clear(); 5807 } 5808 5809 if(status) { 5810 *status = lStatus; 5811 } 5812 return handle; 5813} 5814 5815sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5816{ 5817 sp<EffectModule> effect; 5818 5819 sp<EffectChain> chain = getEffectChain_l(sessionId); 5820 if (chain != 0) { 5821 effect = chain->getEffectFromId_l(effectId); 5822 } 5823 return effect; 5824} 5825 5826// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5827// PlaybackThread::mLock held 5828status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5829{ 5830 // check for existing effect chain with the requested audio session 5831 int sessionId = effect->sessionId(); 5832 sp<EffectChain> chain = getEffectChain_l(sessionId); 5833 bool chainCreated = false; 5834 5835 if (chain == 0) { 5836 // create a new chain for this session 5837 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5838 chain = new EffectChain(this, sessionId); 5839 addEffectChain_l(chain); 5840 chain->setStrategy(getStrategyForSession_l(sessionId)); 5841 chainCreated = true; 5842 } 5843 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5844 5845 if (chain->getEffectFromId_l(effect->id()) != 0) { 5846 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5847 this, effect->desc().name, chain.get()); 5848 return BAD_VALUE; 5849 } 5850 5851 status_t status = chain->addEffect_l(effect); 5852 if (status != NO_ERROR) { 5853 if (chainCreated) { 5854 removeEffectChain_l(chain); 5855 } 5856 return status; 5857 } 5858 5859 effect->setDevice(mDevice); 5860 effect->setMode(mAudioFlinger->getMode()); 5861 return NO_ERROR; 5862} 5863 5864void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5865 5866 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5867 effect_descriptor_t desc = effect->desc(); 5868 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5869 detachAuxEffect_l(effect->id()); 5870 } 5871 5872 sp<EffectChain> chain = effect->chain().promote(); 5873 if (chain != 0) { 5874 // remove effect chain if removing last effect 5875 if (chain->removeEffect_l(effect) == 0) { 5876 removeEffectChain_l(chain); 5877 } 5878 } else { 5879 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5880 } 5881} 5882 5883void AudioFlinger::ThreadBase::lockEffectChains_l( 5884 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5885{ 5886 effectChains = mEffectChains; 5887 for (size_t i = 0; i < mEffectChains.size(); i++) { 5888 mEffectChains[i]->lock(); 5889 } 5890} 5891 5892void AudioFlinger::ThreadBase::unlockEffectChains( 5893 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5894{ 5895 for (size_t i = 0; i < effectChains.size(); i++) { 5896 effectChains[i]->unlock(); 5897 } 5898} 5899 5900sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5901{ 5902 Mutex::Autolock _l(mLock); 5903 return getEffectChain_l(sessionId); 5904} 5905 5906sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5907{ 5908 sp<EffectChain> chain; 5909 5910 size_t size = mEffectChains.size(); 5911 for (size_t i = 0; i < size; i++) { 5912 if (mEffectChains[i]->sessionId() == sessionId) { 5913 chain = mEffectChains[i]; 5914 break; 5915 } 5916 } 5917 return chain; 5918} 5919 5920void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5921{ 5922 Mutex::Autolock _l(mLock); 5923 size_t size = mEffectChains.size(); 5924 for (size_t i = 0; i < size; i++) { 5925 mEffectChains[i]->setMode_l(mode); 5926 } 5927} 5928 5929void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5930 const wp<EffectHandle>& handle, 5931 bool unpiniflast) { 5932 5933 Mutex::Autolock _l(mLock); 5934 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5935 // delete the effect module if removing last handle on it 5936 if (effect->removeHandle(handle) == 0) { 5937 if (!effect->isPinned() || unpiniflast) { 5938 removeEffect_l(effect); 5939 AudioSystem::unregisterEffect(effect->id()); 5940 } 5941 } 5942} 5943 5944status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5945{ 5946 int session = chain->sessionId(); 5947 int16_t *buffer = mMixBuffer; 5948 bool ownsBuffer = false; 5949 5950 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5951 if (session > 0) { 5952 // Only one effect chain can be present in direct output thread and it uses 5953 // the mix buffer as input 5954 if (mType != DIRECT) { 5955 size_t numSamples = mFrameCount * mChannelCount; 5956 buffer = new int16_t[numSamples]; 5957 memset(buffer, 0, numSamples * sizeof(int16_t)); 5958 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5959 ownsBuffer = true; 5960 } 5961 5962 // Attach all tracks with same session ID to this chain. 5963 for (size_t i = 0; i < mTracks.size(); ++i) { 5964 sp<Track> track = mTracks[i]; 5965 if (session == track->sessionId()) { 5966 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5967 track->setMainBuffer(buffer); 5968 chain->incTrackCnt(); 5969 } 5970 } 5971 5972 // indicate all active tracks in the chain 5973 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5974 sp<Track> track = mActiveTracks[i].promote(); 5975 if (track == 0) continue; 5976 if (session == track->sessionId()) { 5977 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5978 chain->incActiveTrackCnt(); 5979 } 5980 } 5981 } 5982 5983 chain->setInBuffer(buffer, ownsBuffer); 5984 chain->setOutBuffer(mMixBuffer); 5985 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5986 // chains list in order to be processed last as it contains output stage effects 5987 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5988 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5989 // after track specific effects and before output stage 5990 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5991 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5992 // Effect chain for other sessions are inserted at beginning of effect 5993 // chains list to be processed before output mix effects. Relative order between other 5994 // sessions is not important 5995 size_t size = mEffectChains.size(); 5996 size_t i = 0; 5997 for (i = 0; i < size; i++) { 5998 if (mEffectChains[i]->sessionId() < session) break; 5999 } 6000 mEffectChains.insertAt(chain, i); 6001 checkSuspendOnAddEffectChain_l(chain); 6002 6003 return NO_ERROR; 6004} 6005 6006size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6007{ 6008 int session = chain->sessionId(); 6009 6010 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6011 6012 for (size_t i = 0; i < mEffectChains.size(); i++) { 6013 if (chain == mEffectChains[i]) { 6014 mEffectChains.removeAt(i); 6015 // detach all active tracks from the chain 6016 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6017 sp<Track> track = mActiveTracks[i].promote(); 6018 if (track == 0) continue; 6019 if (session == track->sessionId()) { 6020 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6021 chain.get(), session); 6022 chain->decActiveTrackCnt(); 6023 } 6024 } 6025 6026 // detach all tracks with same session ID from this chain 6027 for (size_t i = 0; i < mTracks.size(); ++i) { 6028 sp<Track> track = mTracks[i]; 6029 if (session == track->sessionId()) { 6030 track->setMainBuffer(mMixBuffer); 6031 chain->decTrackCnt(); 6032 } 6033 } 6034 break; 6035 } 6036 } 6037 return mEffectChains.size(); 6038} 6039 6040status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6041 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6042{ 6043 Mutex::Autolock _l(mLock); 6044 return attachAuxEffect_l(track, EffectId); 6045} 6046 6047status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6048 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6049{ 6050 status_t status = NO_ERROR; 6051 6052 if (EffectId == 0) { 6053 track->setAuxBuffer(0, NULL); 6054 } else { 6055 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6056 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6057 if (effect != 0) { 6058 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6059 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6060 } else { 6061 status = INVALID_OPERATION; 6062 } 6063 } else { 6064 status = BAD_VALUE; 6065 } 6066 } 6067 return status; 6068} 6069 6070void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6071{ 6072 for (size_t i = 0; i < mTracks.size(); ++i) { 6073 sp<Track> track = mTracks[i]; 6074 if (track->auxEffectId() == effectId) { 6075 attachAuxEffect_l(track, 0); 6076 } 6077 } 6078} 6079 6080status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6081{ 6082 // only one chain per input thread 6083 if (mEffectChains.size() != 0) { 6084 return INVALID_OPERATION; 6085 } 6086 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6087 6088 chain->setInBuffer(NULL); 6089 chain->setOutBuffer(NULL); 6090 6091 checkSuspendOnAddEffectChain_l(chain); 6092 6093 mEffectChains.add(chain); 6094 6095 return NO_ERROR; 6096} 6097 6098size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6099{ 6100 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6101 ALOGW_IF(mEffectChains.size() != 1, 6102 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6103 chain.get(), mEffectChains.size(), this); 6104 if (mEffectChains.size() == 1) { 6105 mEffectChains.removeAt(0); 6106 } 6107 return 0; 6108} 6109 6110// ---------------------------------------------------------------------------- 6111// EffectModule implementation 6112// ---------------------------------------------------------------------------- 6113 6114#undef LOG_TAG 6115#define LOG_TAG "AudioFlinger::EffectModule" 6116 6117AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6118 const wp<AudioFlinger::EffectChain>& chain, 6119 effect_descriptor_t *desc, 6120 int id, 6121 int sessionId) 6122 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6123 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6124{ 6125 ALOGV("Constructor %p", this); 6126 int lStatus; 6127 sp<ThreadBase> thread = mThread.promote(); 6128 if (thread == 0) { 6129 return; 6130 } 6131 6132 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6133 6134 // create effect engine from effect factory 6135 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6136 6137 if (mStatus != NO_ERROR) { 6138 return; 6139 } 6140 lStatus = init(); 6141 if (lStatus < 0) { 6142 mStatus = lStatus; 6143 goto Error; 6144 } 6145 6146 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6147 mPinned = true; 6148 } 6149 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6150 return; 6151Error: 6152 EffectRelease(mEffectInterface); 6153 mEffectInterface = NULL; 6154 ALOGV("Constructor Error %d", mStatus); 6155} 6156 6157AudioFlinger::EffectModule::~EffectModule() 6158{ 6159 ALOGV("Destructor %p", this); 6160 if (mEffectInterface != NULL) { 6161 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6162 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6163 sp<ThreadBase> thread = mThread.promote(); 6164 if (thread != 0) { 6165 audio_stream_t *stream = thread->stream(); 6166 if (stream != NULL) { 6167 stream->remove_audio_effect(stream, mEffectInterface); 6168 } 6169 } 6170 } 6171 // release effect engine 6172 EffectRelease(mEffectInterface); 6173 } 6174} 6175 6176status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6177{ 6178 status_t status; 6179 6180 Mutex::Autolock _l(mLock); 6181 // First handle in mHandles has highest priority and controls the effect module 6182 int priority = handle->priority(); 6183 size_t size = mHandles.size(); 6184 sp<EffectHandle> h; 6185 size_t i; 6186 for (i = 0; i < size; i++) { 6187 h = mHandles[i].promote(); 6188 if (h == 0) continue; 6189 if (h->priority() <= priority) break; 6190 } 6191 // if inserted in first place, move effect control from previous owner to this handle 6192 if (i == 0) { 6193 bool enabled = false; 6194 if (h != 0) { 6195 enabled = h->enabled(); 6196 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6197 } 6198 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6199 status = NO_ERROR; 6200 } else { 6201 status = ALREADY_EXISTS; 6202 } 6203 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6204 mHandles.insertAt(handle, i); 6205 return status; 6206} 6207 6208size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6209{ 6210 Mutex::Autolock _l(mLock); 6211 size_t size = mHandles.size(); 6212 size_t i; 6213 for (i = 0; i < size; i++) { 6214 if (mHandles[i] == handle) break; 6215 } 6216 if (i == size) { 6217 return size; 6218 } 6219 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6220 6221 bool enabled = false; 6222 EffectHandle *hdl = handle.unsafe_get(); 6223 if (hdl) { 6224 ALOGV("removeHandle() unsafe_get OK"); 6225 enabled = hdl->enabled(); 6226 } 6227 mHandles.removeAt(i); 6228 size = mHandles.size(); 6229 // if removed from first place, move effect control from this handle to next in line 6230 if (i == 0 && size != 0) { 6231 sp<EffectHandle> h = mHandles[0].promote(); 6232 if (h != 0) { 6233 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6234 } 6235 } 6236 6237 // Prevent calls to process() and other functions on effect interface from now on. 6238 // The effect engine will be released by the destructor when the last strong reference on 6239 // this object is released which can happen after next process is called. 6240 if (size == 0 && !mPinned) { 6241 mState = DESTROYED; 6242 } 6243 6244 return size; 6245} 6246 6247sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6248{ 6249 Mutex::Autolock _l(mLock); 6250 sp<EffectHandle> handle; 6251 if (mHandles.size() != 0) { 6252 handle = mHandles[0].promote(); 6253 } 6254 return handle; 6255} 6256 6257void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6258{ 6259 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6260 // keep a strong reference on this EffectModule to avoid calling the 6261 // destructor before we exit 6262 sp<EffectModule> keep(this); 6263 { 6264 sp<ThreadBase> thread = mThread.promote(); 6265 if (thread != 0) { 6266 thread->disconnectEffect(keep, handle, unpiniflast); 6267 } 6268 } 6269} 6270 6271void AudioFlinger::EffectModule::updateState() { 6272 Mutex::Autolock _l(mLock); 6273 6274 switch (mState) { 6275 case RESTART: 6276 reset_l(); 6277 // FALL THROUGH 6278 6279 case STARTING: 6280 // clear auxiliary effect input buffer for next accumulation 6281 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6282 memset(mConfig.inputCfg.buffer.raw, 6283 0, 6284 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6285 } 6286 start_l(); 6287 mState = ACTIVE; 6288 break; 6289 case STOPPING: 6290 stop_l(); 6291 mDisableWaitCnt = mMaxDisableWaitCnt; 6292 mState = STOPPED; 6293 break; 6294 case STOPPED: 6295 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6296 // turn off sequence. 6297 if (--mDisableWaitCnt == 0) { 6298 reset_l(); 6299 mState = IDLE; 6300 } 6301 break; 6302 default: //IDLE , ACTIVE, DESTROYED 6303 break; 6304 } 6305} 6306 6307void AudioFlinger::EffectModule::process() 6308{ 6309 Mutex::Autolock _l(mLock); 6310 6311 if (mState == DESTROYED || mEffectInterface == NULL || 6312 mConfig.inputCfg.buffer.raw == NULL || 6313 mConfig.outputCfg.buffer.raw == NULL) { 6314 return; 6315 } 6316 6317 if (isProcessEnabled()) { 6318 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6319 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6320 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6321 mConfig.inputCfg.buffer.s32, 6322 mConfig.inputCfg.buffer.frameCount/2); 6323 } 6324 6325 // do the actual processing in the effect engine 6326 int ret = (*mEffectInterface)->process(mEffectInterface, 6327 &mConfig.inputCfg.buffer, 6328 &mConfig.outputCfg.buffer); 6329 6330 // force transition to IDLE state when engine is ready 6331 if (mState == STOPPED && ret == -ENODATA) { 6332 mDisableWaitCnt = 1; 6333 } 6334 6335 // clear auxiliary effect input buffer for next accumulation 6336 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6337 memset(mConfig.inputCfg.buffer.raw, 0, 6338 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6339 } 6340 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6341 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6342 // If an insert effect is idle and input buffer is different from output buffer, 6343 // accumulate input onto output 6344 sp<EffectChain> chain = mChain.promote(); 6345 if (chain != 0 && chain->activeTrackCnt() != 0) { 6346 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6347 int16_t *in = mConfig.inputCfg.buffer.s16; 6348 int16_t *out = mConfig.outputCfg.buffer.s16; 6349 for (size_t i = 0; i < frameCnt; i++) { 6350 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6351 } 6352 } 6353 } 6354} 6355 6356void AudioFlinger::EffectModule::reset_l() 6357{ 6358 if (mEffectInterface == NULL) { 6359 return; 6360 } 6361 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6362} 6363 6364status_t AudioFlinger::EffectModule::configure() 6365{ 6366 uint32_t channels; 6367 if (mEffectInterface == NULL) { 6368 return NO_INIT; 6369 } 6370 6371 sp<ThreadBase> thread = mThread.promote(); 6372 if (thread == 0) { 6373 return DEAD_OBJECT; 6374 } 6375 6376 // TODO: handle configuration of effects replacing track process 6377 if (thread->channelCount() == 1) { 6378 channels = AUDIO_CHANNEL_OUT_MONO; 6379 } else { 6380 channels = AUDIO_CHANNEL_OUT_STEREO; 6381 } 6382 6383 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6384 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6385 } else { 6386 mConfig.inputCfg.channels = channels; 6387 } 6388 mConfig.outputCfg.channels = channels; 6389 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6390 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6391 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6392 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6393 mConfig.inputCfg.bufferProvider.cookie = NULL; 6394 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6395 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6396 mConfig.outputCfg.bufferProvider.cookie = NULL; 6397 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6398 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6399 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6400 // Insert effect: 6401 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6402 // always overwrites output buffer: input buffer == output buffer 6403 // - in other sessions: 6404 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6405 // other effect: overwrites output buffer: input buffer == output buffer 6406 // Auxiliary effect: 6407 // accumulates in output buffer: input buffer != output buffer 6408 // Therefore: accumulate <=> input buffer != output buffer 6409 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6410 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6411 } else { 6412 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6413 } 6414 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6415 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6416 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6417 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6418 6419 ALOGV("configure() %p thread %p buffer %p framecount %d", 6420 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6421 6422 status_t cmdStatus; 6423 uint32_t size = sizeof(int); 6424 status_t status = (*mEffectInterface)->command(mEffectInterface, 6425 EFFECT_CMD_SET_CONFIG, 6426 sizeof(effect_config_t), 6427 &mConfig, 6428 &size, 6429 &cmdStatus); 6430 if (status == 0) { 6431 status = cmdStatus; 6432 } 6433 6434 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6435 (1000 * mConfig.outputCfg.buffer.frameCount); 6436 6437 return status; 6438} 6439 6440status_t AudioFlinger::EffectModule::init() 6441{ 6442 Mutex::Autolock _l(mLock); 6443 if (mEffectInterface == NULL) { 6444 return NO_INIT; 6445 } 6446 status_t cmdStatus; 6447 uint32_t size = sizeof(status_t); 6448 status_t status = (*mEffectInterface)->command(mEffectInterface, 6449 EFFECT_CMD_INIT, 6450 0, 6451 NULL, 6452 &size, 6453 &cmdStatus); 6454 if (status == 0) { 6455 status = cmdStatus; 6456 } 6457 return status; 6458} 6459 6460status_t AudioFlinger::EffectModule::start() 6461{ 6462 Mutex::Autolock _l(mLock); 6463 return start_l(); 6464} 6465 6466status_t AudioFlinger::EffectModule::start_l() 6467{ 6468 if (mEffectInterface == NULL) { 6469 return NO_INIT; 6470 } 6471 status_t cmdStatus; 6472 uint32_t size = sizeof(status_t); 6473 status_t status = (*mEffectInterface)->command(mEffectInterface, 6474 EFFECT_CMD_ENABLE, 6475 0, 6476 NULL, 6477 &size, 6478 &cmdStatus); 6479 if (status == 0) { 6480 status = cmdStatus; 6481 } 6482 if (status == 0 && 6483 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6484 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6485 sp<ThreadBase> thread = mThread.promote(); 6486 if (thread != 0) { 6487 audio_stream_t *stream = thread->stream(); 6488 if (stream != NULL) { 6489 stream->add_audio_effect(stream, mEffectInterface); 6490 } 6491 } 6492 } 6493 return status; 6494} 6495 6496status_t AudioFlinger::EffectModule::stop() 6497{ 6498 Mutex::Autolock _l(mLock); 6499 return stop_l(); 6500} 6501 6502status_t AudioFlinger::EffectModule::stop_l() 6503{ 6504 if (mEffectInterface == NULL) { 6505 return NO_INIT; 6506 } 6507 status_t cmdStatus; 6508 uint32_t size = sizeof(status_t); 6509 status_t status = (*mEffectInterface)->command(mEffectInterface, 6510 EFFECT_CMD_DISABLE, 6511 0, 6512 NULL, 6513 &size, 6514 &cmdStatus); 6515 if (status == 0) { 6516 status = cmdStatus; 6517 } 6518 if (status == 0 && 6519 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6520 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6521 sp<ThreadBase> thread = mThread.promote(); 6522 if (thread != 0) { 6523 audio_stream_t *stream = thread->stream(); 6524 if (stream != NULL) { 6525 stream->remove_audio_effect(stream, mEffectInterface); 6526 } 6527 } 6528 } 6529 return status; 6530} 6531 6532status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6533 uint32_t cmdSize, 6534 void *pCmdData, 6535 uint32_t *replySize, 6536 void *pReplyData) 6537{ 6538 Mutex::Autolock _l(mLock); 6539// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6540 6541 if (mState == DESTROYED || mEffectInterface == NULL) { 6542 return NO_INIT; 6543 } 6544 status_t status = (*mEffectInterface)->command(mEffectInterface, 6545 cmdCode, 6546 cmdSize, 6547 pCmdData, 6548 replySize, 6549 pReplyData); 6550 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6551 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6552 for (size_t i = 1; i < mHandles.size(); i++) { 6553 sp<EffectHandle> h = mHandles[i].promote(); 6554 if (h != 0) { 6555 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6556 } 6557 } 6558 } 6559 return status; 6560} 6561 6562status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6563{ 6564 6565 Mutex::Autolock _l(mLock); 6566 ALOGV("setEnabled %p enabled %d", this, enabled); 6567 6568 if (enabled != isEnabled()) { 6569 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6570 if (enabled && status != NO_ERROR) { 6571 return status; 6572 } 6573 6574 switch (mState) { 6575 // going from disabled to enabled 6576 case IDLE: 6577 mState = STARTING; 6578 break; 6579 case STOPPED: 6580 mState = RESTART; 6581 break; 6582 case STOPPING: 6583 mState = ACTIVE; 6584 break; 6585 6586 // going from enabled to disabled 6587 case RESTART: 6588 mState = STOPPED; 6589 break; 6590 case STARTING: 6591 mState = IDLE; 6592 break; 6593 case ACTIVE: 6594 mState = STOPPING; 6595 break; 6596 case DESTROYED: 6597 return NO_ERROR; // simply ignore as we are being destroyed 6598 } 6599 for (size_t i = 1; i < mHandles.size(); i++) { 6600 sp<EffectHandle> h = mHandles[i].promote(); 6601 if (h != 0) { 6602 h->setEnabled(enabled); 6603 } 6604 } 6605 } 6606 return NO_ERROR; 6607} 6608 6609bool AudioFlinger::EffectModule::isEnabled() 6610{ 6611 switch (mState) { 6612 case RESTART: 6613 case STARTING: 6614 case ACTIVE: 6615 return true; 6616 case IDLE: 6617 case STOPPING: 6618 case STOPPED: 6619 case DESTROYED: 6620 default: 6621 return false; 6622 } 6623} 6624 6625bool AudioFlinger::EffectModule::isProcessEnabled() 6626{ 6627 switch (mState) { 6628 case RESTART: 6629 case ACTIVE: 6630 case STOPPING: 6631 case STOPPED: 6632 return true; 6633 case IDLE: 6634 case STARTING: 6635 case DESTROYED: 6636 default: 6637 return false; 6638 } 6639} 6640 6641status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6642{ 6643 Mutex::Autolock _l(mLock); 6644 status_t status = NO_ERROR; 6645 6646 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6647 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6648 if (isProcessEnabled() && 6649 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6650 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6651 status_t cmdStatus; 6652 uint32_t volume[2]; 6653 uint32_t *pVolume = NULL; 6654 uint32_t size = sizeof(volume); 6655 volume[0] = *left; 6656 volume[1] = *right; 6657 if (controller) { 6658 pVolume = volume; 6659 } 6660 status = (*mEffectInterface)->command(mEffectInterface, 6661 EFFECT_CMD_SET_VOLUME, 6662 size, 6663 volume, 6664 &size, 6665 pVolume); 6666 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6667 *left = volume[0]; 6668 *right = volume[1]; 6669 } 6670 } 6671 return status; 6672} 6673 6674status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6675{ 6676 Mutex::Autolock _l(mLock); 6677 status_t status = NO_ERROR; 6678 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6679 // audio pre processing modules on RecordThread can receive both output and 6680 // input device indication in the same call 6681 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6682 if (dev) { 6683 status_t cmdStatus; 6684 uint32_t size = sizeof(status_t); 6685 6686 status = (*mEffectInterface)->command(mEffectInterface, 6687 EFFECT_CMD_SET_DEVICE, 6688 sizeof(uint32_t), 6689 &dev, 6690 &size, 6691 &cmdStatus); 6692 if (status == NO_ERROR) { 6693 status = cmdStatus; 6694 } 6695 } 6696 dev = device & AUDIO_DEVICE_IN_ALL; 6697 if (dev) { 6698 status_t cmdStatus; 6699 uint32_t size = sizeof(status_t); 6700 6701 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6702 EFFECT_CMD_SET_INPUT_DEVICE, 6703 sizeof(uint32_t), 6704 &dev, 6705 &size, 6706 &cmdStatus); 6707 if (status2 == NO_ERROR) { 6708 status2 = cmdStatus; 6709 } 6710 if (status == NO_ERROR) { 6711 status = status2; 6712 } 6713 } 6714 } 6715 return status; 6716} 6717 6718status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6719{ 6720 Mutex::Autolock _l(mLock); 6721 status_t status = NO_ERROR; 6722 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6723 status_t cmdStatus; 6724 uint32_t size = sizeof(status_t); 6725 status = (*mEffectInterface)->command(mEffectInterface, 6726 EFFECT_CMD_SET_AUDIO_MODE, 6727 sizeof(audio_mode_t), 6728 &mode, 6729 &size, 6730 &cmdStatus); 6731 if (status == NO_ERROR) { 6732 status = cmdStatus; 6733 } 6734 } 6735 return status; 6736} 6737 6738void AudioFlinger::EffectModule::setSuspended(bool suspended) 6739{ 6740 Mutex::Autolock _l(mLock); 6741 mSuspended = suspended; 6742} 6743 6744bool AudioFlinger::EffectModule::suspended() const 6745{ 6746 Mutex::Autolock _l(mLock); 6747 return mSuspended; 6748} 6749 6750status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6751{ 6752 const size_t SIZE = 256; 6753 char buffer[SIZE]; 6754 String8 result; 6755 6756 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6757 result.append(buffer); 6758 6759 bool locked = tryLock(mLock); 6760 // failed to lock - AudioFlinger is probably deadlocked 6761 if (!locked) { 6762 result.append("\t\tCould not lock Fx mutex:\n"); 6763 } 6764 6765 result.append("\t\tSession Status State Engine:\n"); 6766 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6767 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6768 result.append(buffer); 6769 6770 result.append("\t\tDescriptor:\n"); 6771 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6772 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6773 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6774 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6775 result.append(buffer); 6776 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6777 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6778 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6779 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6780 result.append(buffer); 6781 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6782 mDescriptor.apiVersion, 6783 mDescriptor.flags); 6784 result.append(buffer); 6785 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6786 mDescriptor.name); 6787 result.append(buffer); 6788 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6789 mDescriptor.implementor); 6790 result.append(buffer); 6791 6792 result.append("\t\t- Input configuration:\n"); 6793 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6794 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6795 (uint32_t)mConfig.inputCfg.buffer.raw, 6796 mConfig.inputCfg.buffer.frameCount, 6797 mConfig.inputCfg.samplingRate, 6798 mConfig.inputCfg.channels, 6799 mConfig.inputCfg.format); 6800 result.append(buffer); 6801 6802 result.append("\t\t- Output configuration:\n"); 6803 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6804 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6805 (uint32_t)mConfig.outputCfg.buffer.raw, 6806 mConfig.outputCfg.buffer.frameCount, 6807 mConfig.outputCfg.samplingRate, 6808 mConfig.outputCfg.channels, 6809 mConfig.outputCfg.format); 6810 result.append(buffer); 6811 6812 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6813 result.append(buffer); 6814 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6815 for (size_t i = 0; i < mHandles.size(); ++i) { 6816 sp<EffectHandle> handle = mHandles[i].promote(); 6817 if (handle != 0) { 6818 handle->dump(buffer, SIZE); 6819 result.append(buffer); 6820 } 6821 } 6822 6823 result.append("\n"); 6824 6825 write(fd, result.string(), result.length()); 6826 6827 if (locked) { 6828 mLock.unlock(); 6829 } 6830 6831 return NO_ERROR; 6832} 6833 6834// ---------------------------------------------------------------------------- 6835// EffectHandle implementation 6836// ---------------------------------------------------------------------------- 6837 6838#undef LOG_TAG 6839#define LOG_TAG "AudioFlinger::EffectHandle" 6840 6841AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6842 const sp<AudioFlinger::Client>& client, 6843 const sp<IEffectClient>& effectClient, 6844 int32_t priority) 6845 : BnEffect(), 6846 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6847 mPriority(priority), mHasControl(false), mEnabled(false) 6848{ 6849 ALOGV("constructor %p", this); 6850 6851 if (client == 0) { 6852 return; 6853 } 6854 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6855 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6856 if (mCblkMemory != 0) { 6857 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6858 6859 if (mCblk) { 6860 new(mCblk) effect_param_cblk_t(); 6861 mBuffer = (uint8_t *)mCblk + bufOffset; 6862 } 6863 } else { 6864 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6865 return; 6866 } 6867} 6868 6869AudioFlinger::EffectHandle::~EffectHandle() 6870{ 6871 ALOGV("Destructor %p", this); 6872 disconnect(false); 6873 ALOGV("Destructor DONE %p", this); 6874} 6875 6876status_t AudioFlinger::EffectHandle::enable() 6877{ 6878 ALOGV("enable %p", this); 6879 if (!mHasControl) return INVALID_OPERATION; 6880 if (mEffect == 0) return DEAD_OBJECT; 6881 6882 if (mEnabled) { 6883 return NO_ERROR; 6884 } 6885 6886 mEnabled = true; 6887 6888 sp<ThreadBase> thread = mEffect->thread().promote(); 6889 if (thread != 0) { 6890 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6891 } 6892 6893 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6894 if (mEffect->suspended()) { 6895 return NO_ERROR; 6896 } 6897 6898 status_t status = mEffect->setEnabled(true); 6899 if (status != NO_ERROR) { 6900 if (thread != 0) { 6901 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6902 } 6903 mEnabled = false; 6904 } 6905 return status; 6906} 6907 6908status_t AudioFlinger::EffectHandle::disable() 6909{ 6910 ALOGV("disable %p", this); 6911 if (!mHasControl) return INVALID_OPERATION; 6912 if (mEffect == 0) return DEAD_OBJECT; 6913 6914 if (!mEnabled) { 6915 return NO_ERROR; 6916 } 6917 mEnabled = false; 6918 6919 if (mEffect->suspended()) { 6920 return NO_ERROR; 6921 } 6922 6923 status_t status = mEffect->setEnabled(false); 6924 6925 sp<ThreadBase> thread = mEffect->thread().promote(); 6926 if (thread != 0) { 6927 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6928 } 6929 6930 return status; 6931} 6932 6933void AudioFlinger::EffectHandle::disconnect() 6934{ 6935 disconnect(true); 6936} 6937 6938void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6939{ 6940 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6941 if (mEffect == 0) { 6942 return; 6943 } 6944 mEffect->disconnect(this, unpiniflast); 6945 6946 if (mHasControl && mEnabled) { 6947 sp<ThreadBase> thread = mEffect->thread().promote(); 6948 if (thread != 0) { 6949 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6950 } 6951 } 6952 6953 // release sp on module => module destructor can be called now 6954 mEffect.clear(); 6955 if (mClient != 0) { 6956 if (mCblk) { 6957 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6958 } 6959 mCblkMemory.clear(); // and free the shared memory 6960 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6961 mClient.clear(); 6962 } 6963} 6964 6965status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6966 uint32_t cmdSize, 6967 void *pCmdData, 6968 uint32_t *replySize, 6969 void *pReplyData) 6970{ 6971// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6972// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6973 6974 // only get parameter command is permitted for applications not controlling the effect 6975 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6976 return INVALID_OPERATION; 6977 } 6978 if (mEffect == 0) return DEAD_OBJECT; 6979 if (mClient == 0) return INVALID_OPERATION; 6980 6981 // handle commands that are not forwarded transparently to effect engine 6982 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6983 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6984 // no risk to block the whole media server process or mixer threads is we are stuck here 6985 Mutex::Autolock _l(mCblk->lock); 6986 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6987 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6988 mCblk->serverIndex = 0; 6989 mCblk->clientIndex = 0; 6990 return BAD_VALUE; 6991 } 6992 status_t status = NO_ERROR; 6993 while (mCblk->serverIndex < mCblk->clientIndex) { 6994 int reply; 6995 uint32_t rsize = sizeof(int); 6996 int *p = (int *)(mBuffer + mCblk->serverIndex); 6997 int size = *p++; 6998 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6999 ALOGW("command(): invalid parameter block size"); 7000 break; 7001 } 7002 effect_param_t *param = (effect_param_t *)p; 7003 if (param->psize == 0 || param->vsize == 0) { 7004 ALOGW("command(): null parameter or value size"); 7005 mCblk->serverIndex += size; 7006 continue; 7007 } 7008 uint32_t psize = sizeof(effect_param_t) + 7009 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7010 param->vsize; 7011 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7012 psize, 7013 p, 7014 &rsize, 7015 &reply); 7016 // stop at first error encountered 7017 if (ret != NO_ERROR) { 7018 status = ret; 7019 *(int *)pReplyData = reply; 7020 break; 7021 } else if (reply != NO_ERROR) { 7022 *(int *)pReplyData = reply; 7023 break; 7024 } 7025 mCblk->serverIndex += size; 7026 } 7027 mCblk->serverIndex = 0; 7028 mCblk->clientIndex = 0; 7029 return status; 7030 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7031 *(int *)pReplyData = NO_ERROR; 7032 return enable(); 7033 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7034 *(int *)pReplyData = NO_ERROR; 7035 return disable(); 7036 } 7037 7038 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7039} 7040 7041sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7042 return mCblkMemory; 7043} 7044 7045void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7046{ 7047 ALOGV("setControl %p control %d", this, hasControl); 7048 7049 mHasControl = hasControl; 7050 mEnabled = enabled; 7051 7052 if (signal && mEffectClient != 0) { 7053 mEffectClient->controlStatusChanged(hasControl); 7054 } 7055} 7056 7057void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7058 uint32_t cmdSize, 7059 void *pCmdData, 7060 uint32_t replySize, 7061 void *pReplyData) 7062{ 7063 if (mEffectClient != 0) { 7064 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7065 } 7066} 7067 7068 7069 7070void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7071{ 7072 if (mEffectClient != 0) { 7073 mEffectClient->enableStatusChanged(enabled); 7074 } 7075} 7076 7077status_t AudioFlinger::EffectHandle::onTransact( 7078 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7079{ 7080 return BnEffect::onTransact(code, data, reply, flags); 7081} 7082 7083 7084void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7085{ 7086 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7087 7088 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7089 (mClient == NULL) ? getpid() : mClient->pid(), 7090 mPriority, 7091 mHasControl, 7092 !locked, 7093 mCblk ? mCblk->clientIndex : 0, 7094 mCblk ? mCblk->serverIndex : 0 7095 ); 7096 7097 if (locked) { 7098 mCblk->lock.unlock(); 7099 } 7100} 7101 7102#undef LOG_TAG 7103#define LOG_TAG "AudioFlinger::EffectChain" 7104 7105AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7106 int sessionId) 7107 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7108 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7109 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7110{ 7111 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7112 sp<ThreadBase> thread = mThread.promote(); 7113 if (thread == 0) { 7114 return; 7115 } 7116 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7117 thread->frameCount(); 7118} 7119 7120AudioFlinger::EffectChain::~EffectChain() 7121{ 7122 if (mOwnInBuffer) { 7123 delete mInBuffer; 7124 } 7125 7126} 7127 7128// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7129sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7130{ 7131 sp<EffectModule> effect; 7132 size_t size = mEffects.size(); 7133 7134 for (size_t i = 0; i < size; i++) { 7135 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7136 effect = mEffects[i]; 7137 break; 7138 } 7139 } 7140 return effect; 7141} 7142 7143// getEffectFromId_l() must be called with ThreadBase::mLock held 7144sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7145{ 7146 sp<EffectModule> effect; 7147 size_t size = mEffects.size(); 7148 7149 for (size_t i = 0; i < size; i++) { 7150 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7151 if (id == 0 || mEffects[i]->id() == id) { 7152 effect = mEffects[i]; 7153 break; 7154 } 7155 } 7156 return effect; 7157} 7158 7159// getEffectFromType_l() must be called with ThreadBase::mLock held 7160sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7161 const effect_uuid_t *type) 7162{ 7163 sp<EffectModule> effect; 7164 size_t size = mEffects.size(); 7165 7166 for (size_t i = 0; i < size; i++) { 7167 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7168 effect = mEffects[i]; 7169 break; 7170 } 7171 } 7172 return effect; 7173} 7174 7175// Must be called with EffectChain::mLock locked 7176void AudioFlinger::EffectChain::process_l() 7177{ 7178 sp<ThreadBase> thread = mThread.promote(); 7179 if (thread == 0) { 7180 ALOGW("process_l(): cannot promote mixer thread"); 7181 return; 7182 } 7183 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7184 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7185 // always process effects unless no more tracks are on the session and the effect tail 7186 // has been rendered 7187 bool doProcess = true; 7188 if (!isGlobalSession) { 7189 bool tracksOnSession = (trackCnt() != 0); 7190 7191 if (!tracksOnSession && mTailBufferCount == 0) { 7192 doProcess = false; 7193 } 7194 7195 if (activeTrackCnt() == 0) { 7196 // if no track is active and the effect tail has not been rendered, 7197 // the input buffer must be cleared here as the mixer process will not do it 7198 if (tracksOnSession || mTailBufferCount > 0) { 7199 size_t numSamples = thread->frameCount() * thread->channelCount(); 7200 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7201 if (mTailBufferCount > 0) { 7202 mTailBufferCount--; 7203 } 7204 } 7205 } 7206 } 7207 7208 size_t size = mEffects.size(); 7209 if (doProcess) { 7210 for (size_t i = 0; i < size; i++) { 7211 mEffects[i]->process(); 7212 } 7213 } 7214 for (size_t i = 0; i < size; i++) { 7215 mEffects[i]->updateState(); 7216 } 7217} 7218 7219// addEffect_l() must be called with PlaybackThread::mLock held 7220status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7221{ 7222 effect_descriptor_t desc = effect->desc(); 7223 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7224 7225 Mutex::Autolock _l(mLock); 7226 effect->setChain(this); 7227 sp<ThreadBase> thread = mThread.promote(); 7228 if (thread == 0) { 7229 return NO_INIT; 7230 } 7231 effect->setThread(thread); 7232 7233 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7234 // Auxiliary effects are inserted at the beginning of mEffects vector as 7235 // they are processed first and accumulated in chain input buffer 7236 mEffects.insertAt(effect, 0); 7237 7238 // the input buffer for auxiliary effect contains mono samples in 7239 // 32 bit format. This is to avoid saturation in AudoMixer 7240 // accumulation stage. Saturation is done in EffectModule::process() before 7241 // calling the process in effect engine 7242 size_t numSamples = thread->frameCount(); 7243 int32_t *buffer = new int32_t[numSamples]; 7244 memset(buffer, 0, numSamples * sizeof(int32_t)); 7245 effect->setInBuffer((int16_t *)buffer); 7246 // auxiliary effects output samples to chain input buffer for further processing 7247 // by insert effects 7248 effect->setOutBuffer(mInBuffer); 7249 } else { 7250 // Insert effects are inserted at the end of mEffects vector as they are processed 7251 // after track and auxiliary effects. 7252 // Insert effect order as a function of indicated preference: 7253 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7254 // another effect is present 7255 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7256 // last effect claiming first position 7257 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7258 // first effect claiming last position 7259 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7260 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7261 // already present 7262 7263 int size = (int)mEffects.size(); 7264 int idx_insert = size; 7265 int idx_insert_first = -1; 7266 int idx_insert_last = -1; 7267 7268 for (int i = 0; i < size; i++) { 7269 effect_descriptor_t d = mEffects[i]->desc(); 7270 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7271 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7272 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7273 // check invalid effect chaining combinations 7274 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7275 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7276 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7277 return INVALID_OPERATION; 7278 } 7279 // remember position of first insert effect and by default 7280 // select this as insert position for new effect 7281 if (idx_insert == size) { 7282 idx_insert = i; 7283 } 7284 // remember position of last insert effect claiming 7285 // first position 7286 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7287 idx_insert_first = i; 7288 } 7289 // remember position of first insert effect claiming 7290 // last position 7291 if (iPref == EFFECT_FLAG_INSERT_LAST && 7292 idx_insert_last == -1) { 7293 idx_insert_last = i; 7294 } 7295 } 7296 } 7297 7298 // modify idx_insert from first position if needed 7299 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7300 if (idx_insert_last != -1) { 7301 idx_insert = idx_insert_last; 7302 } else { 7303 idx_insert = size; 7304 } 7305 } else { 7306 if (idx_insert_first != -1) { 7307 idx_insert = idx_insert_first + 1; 7308 } 7309 } 7310 7311 // always read samples from chain input buffer 7312 effect->setInBuffer(mInBuffer); 7313 7314 // if last effect in the chain, output samples to chain 7315 // output buffer, otherwise to chain input buffer 7316 if (idx_insert == size) { 7317 if (idx_insert != 0) { 7318 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7319 mEffects[idx_insert-1]->configure(); 7320 } 7321 effect->setOutBuffer(mOutBuffer); 7322 } else { 7323 effect->setOutBuffer(mInBuffer); 7324 } 7325 mEffects.insertAt(effect, idx_insert); 7326 7327 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7328 } 7329 effect->configure(); 7330 return NO_ERROR; 7331} 7332 7333// removeEffect_l() must be called with PlaybackThread::mLock held 7334size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7335{ 7336 Mutex::Autolock _l(mLock); 7337 int size = (int)mEffects.size(); 7338 int i; 7339 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7340 7341 for (i = 0; i < size; i++) { 7342 if (effect == mEffects[i]) { 7343 // calling stop here will remove pre-processing effect from the audio HAL. 7344 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7345 // the middle of a read from audio HAL 7346 if (mEffects[i]->state() == EffectModule::ACTIVE || 7347 mEffects[i]->state() == EffectModule::STOPPING) { 7348 mEffects[i]->stop(); 7349 } 7350 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7351 delete[] effect->inBuffer(); 7352 } else { 7353 if (i == size - 1 && i != 0) { 7354 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7355 mEffects[i - 1]->configure(); 7356 } 7357 } 7358 mEffects.removeAt(i); 7359 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7360 break; 7361 } 7362 } 7363 7364 return mEffects.size(); 7365} 7366 7367// setDevice_l() must be called with PlaybackThread::mLock held 7368void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7369{ 7370 size_t size = mEffects.size(); 7371 for (size_t i = 0; i < size; i++) { 7372 mEffects[i]->setDevice(device); 7373 } 7374} 7375 7376// setMode_l() must be called with PlaybackThread::mLock held 7377void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7378{ 7379 size_t size = mEffects.size(); 7380 for (size_t i = 0; i < size; i++) { 7381 mEffects[i]->setMode(mode); 7382 } 7383} 7384 7385// setVolume_l() must be called with PlaybackThread::mLock held 7386bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7387{ 7388 uint32_t newLeft = *left; 7389 uint32_t newRight = *right; 7390 bool hasControl = false; 7391 int ctrlIdx = -1; 7392 size_t size = mEffects.size(); 7393 7394 // first update volume controller 7395 for (size_t i = size; i > 0; i--) { 7396 if (mEffects[i - 1]->isProcessEnabled() && 7397 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7398 ctrlIdx = i - 1; 7399 hasControl = true; 7400 break; 7401 } 7402 } 7403 7404 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7405 if (hasControl) { 7406 *left = mNewLeftVolume; 7407 *right = mNewRightVolume; 7408 } 7409 return hasControl; 7410 } 7411 7412 mVolumeCtrlIdx = ctrlIdx; 7413 mLeftVolume = newLeft; 7414 mRightVolume = newRight; 7415 7416 // second get volume update from volume controller 7417 if (ctrlIdx >= 0) { 7418 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7419 mNewLeftVolume = newLeft; 7420 mNewRightVolume = newRight; 7421 } 7422 // then indicate volume to all other effects in chain. 7423 // Pass altered volume to effects before volume controller 7424 // and requested volume to effects after controller 7425 uint32_t lVol = newLeft; 7426 uint32_t rVol = newRight; 7427 7428 for (size_t i = 0; i < size; i++) { 7429 if ((int)i == ctrlIdx) continue; 7430 // this also works for ctrlIdx == -1 when there is no volume controller 7431 if ((int)i > ctrlIdx) { 7432 lVol = *left; 7433 rVol = *right; 7434 } 7435 mEffects[i]->setVolume(&lVol, &rVol, false); 7436 } 7437 *left = newLeft; 7438 *right = newRight; 7439 7440 return hasControl; 7441} 7442 7443status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7444{ 7445 const size_t SIZE = 256; 7446 char buffer[SIZE]; 7447 String8 result; 7448 7449 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7450 result.append(buffer); 7451 7452 bool locked = tryLock(mLock); 7453 // failed to lock - AudioFlinger is probably deadlocked 7454 if (!locked) { 7455 result.append("\tCould not lock mutex:\n"); 7456 } 7457 7458 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7459 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7460 mEffects.size(), 7461 (uint32_t)mInBuffer, 7462 (uint32_t)mOutBuffer, 7463 mActiveTrackCnt); 7464 result.append(buffer); 7465 write(fd, result.string(), result.size()); 7466 7467 for (size_t i = 0; i < mEffects.size(); ++i) { 7468 sp<EffectModule> effect = mEffects[i]; 7469 if (effect != 0) { 7470 effect->dump(fd, args); 7471 } 7472 } 7473 7474 if (locked) { 7475 mLock.unlock(); 7476 } 7477 7478 return NO_ERROR; 7479} 7480 7481// must be called with ThreadBase::mLock held 7482void AudioFlinger::EffectChain::setEffectSuspended_l( 7483 const effect_uuid_t *type, bool suspend) 7484{ 7485 sp<SuspendedEffectDesc> desc; 7486 // use effect type UUID timelow as key as there is no real risk of identical 7487 // timeLow fields among effect type UUIDs. 7488 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7489 if (suspend) { 7490 if (index >= 0) { 7491 desc = mSuspendedEffects.valueAt(index); 7492 } else { 7493 desc = new SuspendedEffectDesc(); 7494 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7495 mSuspendedEffects.add(type->timeLow, desc); 7496 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7497 } 7498 if (desc->mRefCount++ == 0) { 7499 sp<EffectModule> effect = getEffectIfEnabled(type); 7500 if (effect != 0) { 7501 desc->mEffect = effect; 7502 effect->setSuspended(true); 7503 effect->setEnabled(false); 7504 } 7505 } 7506 } else { 7507 if (index < 0) { 7508 return; 7509 } 7510 desc = mSuspendedEffects.valueAt(index); 7511 if (desc->mRefCount <= 0) { 7512 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7513 desc->mRefCount = 1; 7514 } 7515 if (--desc->mRefCount == 0) { 7516 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7517 if (desc->mEffect != 0) { 7518 sp<EffectModule> effect = desc->mEffect.promote(); 7519 if (effect != 0) { 7520 effect->setSuspended(false); 7521 sp<EffectHandle> handle = effect->controlHandle(); 7522 if (handle != 0) { 7523 effect->setEnabled(handle->enabled()); 7524 } 7525 } 7526 desc->mEffect.clear(); 7527 } 7528 mSuspendedEffects.removeItemsAt(index); 7529 } 7530 } 7531} 7532 7533// must be called with ThreadBase::mLock held 7534void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7535{ 7536 sp<SuspendedEffectDesc> desc; 7537 7538 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7539 if (suspend) { 7540 if (index >= 0) { 7541 desc = mSuspendedEffects.valueAt(index); 7542 } else { 7543 desc = new SuspendedEffectDesc(); 7544 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7545 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7546 } 7547 if (desc->mRefCount++ == 0) { 7548 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7549 for (size_t i = 0; i < effects.size(); i++) { 7550 setEffectSuspended_l(&effects[i]->desc().type, true); 7551 } 7552 } 7553 } else { 7554 if (index < 0) { 7555 return; 7556 } 7557 desc = mSuspendedEffects.valueAt(index); 7558 if (desc->mRefCount <= 0) { 7559 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7560 desc->mRefCount = 1; 7561 } 7562 if (--desc->mRefCount == 0) { 7563 Vector<const effect_uuid_t *> types; 7564 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7565 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7566 continue; 7567 } 7568 types.add(&mSuspendedEffects.valueAt(i)->mType); 7569 } 7570 for (size_t i = 0; i < types.size(); i++) { 7571 setEffectSuspended_l(types[i], false); 7572 } 7573 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7574 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7575 } 7576 } 7577} 7578 7579 7580// The volume effect is used for automated tests only 7581#ifndef OPENSL_ES_H_ 7582static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7583 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7584const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7585#endif //OPENSL_ES_H_ 7586 7587bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7588{ 7589 // auxiliary effects and visualizer are never suspended on output mix 7590 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7591 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7592 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7593 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7594 return false; 7595 } 7596 return true; 7597} 7598 7599Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7600{ 7601 Vector< sp<EffectModule> > effects; 7602 for (size_t i = 0; i < mEffects.size(); i++) { 7603 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7604 continue; 7605 } 7606 effects.add(mEffects[i]); 7607 } 7608 return effects; 7609} 7610 7611sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7612 const effect_uuid_t *type) 7613{ 7614 sp<EffectModule> effect; 7615 effect = getEffectFromType_l(type); 7616 if (effect != 0 && !effect->isEnabled()) { 7617 effect.clear(); 7618 } 7619 return effect; 7620} 7621 7622void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7623 bool enabled) 7624{ 7625 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7626 if (enabled) { 7627 if (index < 0) { 7628 // if the effect is not suspend check if all effects are suspended 7629 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7630 if (index < 0) { 7631 return; 7632 } 7633 if (!isEffectEligibleForSuspend(effect->desc())) { 7634 return; 7635 } 7636 setEffectSuspended_l(&effect->desc().type, enabled); 7637 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7638 if (index < 0) { 7639 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7640 return; 7641 } 7642 } 7643 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7644 effect->desc().type.timeLow); 7645 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7646 // if effect is requested to suspended but was not yet enabled, supend it now. 7647 if (desc->mEffect == 0) { 7648 desc->mEffect = effect; 7649 effect->setEnabled(false); 7650 effect->setSuspended(true); 7651 } 7652 } else { 7653 if (index < 0) { 7654 return; 7655 } 7656 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7657 effect->desc().type.timeLow); 7658 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7659 desc->mEffect.clear(); 7660 effect->setSuspended(false); 7661 } 7662} 7663 7664#undef LOG_TAG 7665#define LOG_TAG "AudioFlinger" 7666 7667// ---------------------------------------------------------------------------- 7668 7669status_t AudioFlinger::onTransact( 7670 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7671{ 7672 return BnAudioFlinger::onTransact(code, data, reply, flags); 7673} 7674 7675}; // namespace android 7676