AudioFlinger.cpp revision 01c4ebf6b794493898114a502ed36de13137f7e5
1/* 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#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_INIT; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_INIT; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 uint32_t flags, 436 const sp<IMemory>& sharedBuffer, 437 audio_io_handle_t output, 438 bool isTimed, 439 int *sessionId, 440 status_t *status) 441{ 442 sp<PlaybackThread::Track> track; 443 sp<TrackHandle> trackHandle; 444 sp<Client> client; 445 status_t lStatus; 446 int lSessionId; 447 448 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 449 // but if someone uses binder directly they could bypass that and cause us to crash 450 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 451 ALOGE("createTrack() invalid stream type %d", streamType); 452 lStatus = BAD_VALUE; 453 goto Exit; 454 } 455 456 { 457 Mutex::Autolock _l(mLock); 458 PlaybackThread *thread = checkPlaybackThread_l(output); 459 PlaybackThread *effectThread = NULL; 460 if (thread == NULL) { 461 ALOGE("unknown output thread"); 462 lStatus = BAD_VALUE; 463 goto Exit; 464 } 465 466 client = registerPid_l(pid); 467 468 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 469 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 470 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 471 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 472 if (mPlaybackThreads.keyAt(i) != output) { 473 // prevent same audio session on different output threads 474 uint32_t sessions = t->hasAudioSession(*sessionId); 475 if (sessions & PlaybackThread::TRACK_SESSION) { 476 ALOGE("createTrack() session ID %d already in use", *sessionId); 477 lStatus = BAD_VALUE; 478 goto Exit; 479 } 480 // check if an effect with same session ID is waiting for a track to be created 481 if (sessions & PlaybackThread::EFFECT_SESSION) { 482 effectThread = t.get(); 483 } 484 } 485 } 486 lSessionId = *sessionId; 487 } else { 488 // if no audio session id is provided, create one here 489 lSessionId = nextUniqueId(); 490 if (sessionId != NULL) { 491 *sessionId = lSessionId; 492 } 493 } 494 ALOGV("createTrack() lSessionId: %d", lSessionId); 495 496 track = thread->createTrack_l(client, streamType, sampleRate, format, 497 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 498 499 // move effect chain to this output thread if an effect on same session was waiting 500 // for a track to be created 501 if (lStatus == NO_ERROR && effectThread != NULL) { 502 Mutex::Autolock _dl(thread->mLock); 503 Mutex::Autolock _sl(effectThread->mLock); 504 moveEffectChain_l(lSessionId, effectThread, thread, true); 505 } 506 } 507 if (lStatus == NO_ERROR) { 508 trackHandle = new TrackHandle(track); 509 } else { 510 // remove local strong reference to Client before deleting the Track so that the Client 511 // destructor is called by the TrackBase destructor with mLock held 512 client.clear(); 513 track.clear(); 514 } 515 516Exit: 517 if(status) { 518 *status = lStatus; 519 } 520 return trackHandle; 521} 522 523uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 524{ 525 Mutex::Autolock _l(mLock); 526 PlaybackThread *thread = checkPlaybackThread_l(output); 527 if (thread == NULL) { 528 ALOGW("sampleRate() unknown thread %d", output); 529 return 0; 530 } 531 return thread->sampleRate(); 532} 533 534int AudioFlinger::channelCount(audio_io_handle_t output) const 535{ 536 Mutex::Autolock _l(mLock); 537 PlaybackThread *thread = checkPlaybackThread_l(output); 538 if (thread == NULL) { 539 ALOGW("channelCount() unknown thread %d", output); 540 return 0; 541 } 542 return thread->channelCount(); 543} 544 545audio_format_t AudioFlinger::format(audio_io_handle_t output) const 546{ 547 Mutex::Autolock _l(mLock); 548 PlaybackThread *thread = checkPlaybackThread_l(output); 549 if (thread == NULL) { 550 ALOGW("format() unknown thread %d", output); 551 return AUDIO_FORMAT_INVALID; 552 } 553 return thread->format(); 554} 555 556size_t AudioFlinger::frameCount(audio_io_handle_t output) const 557{ 558 Mutex::Autolock _l(mLock); 559 PlaybackThread *thread = checkPlaybackThread_l(output); 560 if (thread == NULL) { 561 ALOGW("frameCount() unknown thread %d", output); 562 return 0; 563 } 564 return thread->frameCount(); 565} 566 567uint32_t AudioFlinger::latency(audio_io_handle_t output) const 568{ 569 Mutex::Autolock _l(mLock); 570 PlaybackThread *thread = checkPlaybackThread_l(output); 571 if (thread == NULL) { 572 ALOGW("latency() unknown thread %d", output); 573 return 0; 574 } 575 return thread->latency(); 576} 577 578status_t AudioFlinger::setMasterVolume(float value) 579{ 580 status_t ret = initCheck(); 581 if (ret != NO_ERROR) { 582 return ret; 583 } 584 585 // check calling permissions 586 if (!settingsAllowed()) { 587 return PERMISSION_DENIED; 588 } 589 590 float swmv = value; 591 592 // when hw supports master volume, don't scale in sw mixer 593 if (MVS_NONE != mMasterVolumeSupportLvl) { 594 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 595 AutoMutex lock(mHardwareLock); 596 audio_hw_device_t *dev = mAudioHwDevs[i]; 597 598 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 599 if (NULL != dev->set_master_volume) { 600 dev->set_master_volume(dev, value); 601 } 602 mHardwareStatus = AUDIO_HW_IDLE; 603 } 604 605 swmv = 1.0; 606 } 607 608 Mutex::Autolock _l(mLock); 609 mMasterVolume = value; 610 mMasterVolumeSW = swmv; 611 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 612 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 613 614 return NO_ERROR; 615} 616 617status_t AudioFlinger::setMode(audio_mode_t mode) 618{ 619 status_t ret = initCheck(); 620 if (ret != NO_ERROR) { 621 return ret; 622 } 623 624 // check calling permissions 625 if (!settingsAllowed()) { 626 return PERMISSION_DENIED; 627 } 628 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 629 ALOGW("Illegal value: setMode(%d)", mode); 630 return BAD_VALUE; 631 } 632 633 { // scope for the lock 634 AutoMutex lock(mHardwareLock); 635 mHardwareStatus = AUDIO_HW_SET_MODE; 636 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 637 mHardwareStatus = AUDIO_HW_IDLE; 638 } 639 640 if (NO_ERROR == ret) { 641 Mutex::Autolock _l(mLock); 642 mMode = mode; 643 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 644 mPlaybackThreads.valueAt(i)->setMode(mode); 645 } 646 647 return ret; 648} 649 650status_t AudioFlinger::setMicMute(bool state) 651{ 652 status_t ret = initCheck(); 653 if (ret != NO_ERROR) { 654 return ret; 655 } 656 657 // check calling permissions 658 if (!settingsAllowed()) { 659 return PERMISSION_DENIED; 660 } 661 662 AutoMutex lock(mHardwareLock); 663 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 664 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 665 mHardwareStatus = AUDIO_HW_IDLE; 666 return ret; 667} 668 669bool AudioFlinger::getMicMute() const 670{ 671 status_t ret = initCheck(); 672 if (ret != NO_ERROR) { 673 return false; 674 } 675 676 bool state = AUDIO_MODE_INVALID; 677 AutoMutex lock(mHardwareLock); 678 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 679 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 680 mHardwareStatus = AUDIO_HW_IDLE; 681 return state; 682} 683 684status_t AudioFlinger::setMasterMute(bool muted) 685{ 686 // check calling permissions 687 if (!settingsAllowed()) { 688 return PERMISSION_DENIED; 689 } 690 691 Mutex::Autolock _l(mLock); 692 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 693 mMasterMute = muted; 694 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 695 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 696 697 return NO_ERROR; 698} 699 700float AudioFlinger::masterVolume() const 701{ 702 Mutex::Autolock _l(mLock); 703 return masterVolume_l(); 704} 705 706float AudioFlinger::masterVolumeSW() const 707{ 708 Mutex::Autolock _l(mLock); 709 return masterVolumeSW_l(); 710} 711 712bool AudioFlinger::masterMute() const 713{ 714 Mutex::Autolock _l(mLock); 715 return masterMute_l(); 716} 717 718float AudioFlinger::masterVolume_l() const 719{ 720 if (MVS_FULL == mMasterVolumeSupportLvl) { 721 float ret_val; 722 AutoMutex lock(mHardwareLock); 723 724 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 725 assert(NULL != mPrimaryHardwareDev); 726 assert(NULL != mPrimaryHardwareDev->get_master_volume); 727 728 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 729 mHardwareStatus = AUDIO_HW_IDLE; 730 return ret_val; 731 } 732 733 return mMasterVolume; 734} 735 736status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 737 audio_io_handle_t output) 738{ 739 // check calling permissions 740 if (!settingsAllowed()) { 741 return PERMISSION_DENIED; 742 } 743 744 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 745 ALOGE("setStreamVolume() invalid stream %d", stream); 746 return BAD_VALUE; 747 } 748 749 AutoMutex lock(mLock); 750 PlaybackThread *thread = NULL; 751 if (output) { 752 thread = checkPlaybackThread_l(output); 753 if (thread == NULL) { 754 return BAD_VALUE; 755 } 756 } 757 758 mStreamTypes[stream].volume = value; 759 760 if (thread == NULL) { 761 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 762 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 763 } 764 } else { 765 thread->setStreamVolume(stream, value); 766 } 767 768 return NO_ERROR; 769} 770 771status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 772{ 773 // check calling permissions 774 if (!settingsAllowed()) { 775 return PERMISSION_DENIED; 776 } 777 778 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 779 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 780 ALOGE("setStreamMute() invalid stream %d", stream); 781 return BAD_VALUE; 782 } 783 784 AutoMutex lock(mLock); 785 mStreamTypes[stream].mute = muted; 786 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 787 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 788 789 return NO_ERROR; 790} 791 792float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 793{ 794 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 795 return 0.0f; 796 } 797 798 AutoMutex lock(mLock); 799 float volume; 800 if (output) { 801 PlaybackThread *thread = checkPlaybackThread_l(output); 802 if (thread == NULL) { 803 return 0.0f; 804 } 805 volume = thread->streamVolume(stream); 806 } else { 807 volume = streamVolume_l(stream); 808 } 809 810 return volume; 811} 812 813bool AudioFlinger::streamMute(audio_stream_type_t stream) const 814{ 815 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 816 return true; 817 } 818 819 AutoMutex lock(mLock); 820 return streamMute_l(stream); 821} 822 823status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 824{ 825 status_t result; 826 827 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 828 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 829 // check calling permissions 830 if (!settingsAllowed()) { 831 return PERMISSION_DENIED; 832 } 833 834 // ioHandle == 0 means the parameters are global to the audio hardware interface 835 if (ioHandle == 0) { 836 AutoMutex lock(mHardwareLock); 837 mHardwareStatus = AUDIO_SET_PARAMETER; 838 status_t final_result = NO_ERROR; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs[i]; 841 result = dev->set_parameters(dev, keyValuePairs.string()); 842 final_result = result ?: final_result; 843 } 844 mHardwareStatus = AUDIO_HW_IDLE; 845 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 846 AudioParameter param = AudioParameter(keyValuePairs); 847 String8 value; 848 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 849 Mutex::Autolock _l(mLock); 850 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 851 if (mBtNrecIsOff != btNrecIsOff) { 852 for (size_t i = 0; i < mRecordThreads.size(); i++) { 853 sp<RecordThread> thread = mRecordThreads.valueAt(i); 854 RecordThread::RecordTrack *track = thread->track(); 855 if (track != NULL) { 856 audio_devices_t device = (audio_devices_t)( 857 thread->device() & AUDIO_DEVICE_IN_ALL); 858 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 859 thread->setEffectSuspended(FX_IID_AEC, 860 suspend, 861 track->sessionId()); 862 thread->setEffectSuspended(FX_IID_NS, 863 suspend, 864 track->sessionId()); 865 } 866 } 867 mBtNrecIsOff = btNrecIsOff; 868 } 869 } 870 return final_result; 871 } 872 873 // hold a strong ref on thread in case closeOutput() or closeInput() is called 874 // and the thread is exited once the lock is released 875 sp<ThreadBase> thread; 876 { 877 Mutex::Autolock _l(mLock); 878 thread = checkPlaybackThread_l(ioHandle); 879 if (thread == NULL) { 880 thread = checkRecordThread_l(ioHandle); 881 } else if (thread == primaryPlaybackThread_l()) { 882 // indicate output device change to all input threads for pre processing 883 AudioParameter param = AudioParameter(keyValuePairs); 884 int value; 885 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 886 for (size_t i = 0; i < mRecordThreads.size(); i++) { 887 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 888 } 889 } 890 } 891 } 892 if (thread != 0) { 893 return thread->setParameters(keyValuePairs); 894 } 895 return BAD_VALUE; 896} 897 898String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 899{ 900// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 901// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 902 903 if (ioHandle == 0) { 904 String8 out_s8; 905 906 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 907 audio_hw_device_t *dev = mAudioHwDevs[i]; 908 char *s = dev->get_parameters(dev, keys.string()); 909 out_s8 += String8(s ? s : ""); 910 free(s); 911 } 912 return out_s8; 913 } 914 915 Mutex::Autolock _l(mLock); 916 917 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 918 if (playbackThread != NULL) { 919 return playbackThread->getParameters(keys); 920 } 921 RecordThread *recordThread = checkRecordThread_l(ioHandle); 922 if (recordThread != NULL) { 923 return recordThread->getParameters(keys); 924 } 925 return String8(""); 926} 927 928size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 929{ 930 status_t ret = initCheck(); 931 if (ret != NO_ERROR) { 932 return 0; 933 } 934 935 AutoMutex lock(mHardwareLock); 936 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 937 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 938 mHardwareStatus = AUDIO_HW_IDLE; 939 return size; 940} 941 942unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 943{ 944 if (ioHandle == 0) { 945 return 0; 946 } 947 948 Mutex::Autolock _l(mLock); 949 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getInputFramesLost(); 953 } 954 return 0; 955} 956 957status_t AudioFlinger::setVoiceVolume(float value) 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return ret; 962 } 963 964 // check calling permissions 965 if (!settingsAllowed()) { 966 return PERMISSION_DENIED; 967 } 968 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 971 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 972 mHardwareStatus = AUDIO_HW_IDLE; 973 974 return ret; 975} 976 977status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 978 audio_io_handle_t output) const 979{ 980 status_t status; 981 982 Mutex::Autolock _l(mLock); 983 984 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 985 if (playbackThread != NULL) { 986 return playbackThread->getRenderPosition(halFrames, dspFrames); 987 } 988 989 return BAD_VALUE; 990} 991 992void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 993{ 994 995 Mutex::Autolock _l(mLock); 996 997 pid_t pid = IPCThreadState::self()->getCallingPid(); 998 if (mNotificationClients.indexOfKey(pid) < 0) { 999 sp<NotificationClient> notificationClient = new NotificationClient(this, 1000 client, 1001 pid); 1002 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1003 1004 mNotificationClients.add(pid, notificationClient); 1005 1006 sp<IBinder> binder = client->asBinder(); 1007 binder->linkToDeath(notificationClient); 1008 1009 // the config change is always sent from playback or record threads to avoid deadlock 1010 // with AudioSystem::gLock 1011 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1012 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1013 } 1014 1015 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1016 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1017 } 1018 } 1019} 1020 1021void AudioFlinger::removeNotificationClient(pid_t pid) 1022{ 1023 Mutex::Autolock _l(mLock); 1024 1025 ssize_t index = mNotificationClients.indexOfKey(pid); 1026 if (index >= 0) { 1027 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 1028 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 1029 mNotificationClients.removeItem(pid); 1030 } 1031 1032 ALOGV("%d died, releasing its sessions", pid); 1033 size_t num = mAudioSessionRefs.size(); 1034 bool removed = false; 1035 for (size_t i = 0; i< num; ) { 1036 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1037 ALOGV(" pid %d @ %d", ref->pid, i); 1038 if (ref->pid == pid) { 1039 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1040 mAudioSessionRefs.removeAt(i); 1041 delete ref; 1042 removed = true; 1043 num--; 1044 } else { 1045 i++; 1046 } 1047 } 1048 if (removed) { 1049 purgeStaleEffects_l(); 1050 } 1051} 1052 1053// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1054void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1055{ 1056 size_t size = mNotificationClients.size(); 1057 for (size_t i = 0; i < size; i++) { 1058 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1059 param2); 1060 } 1061} 1062 1063// removeClient_l() must be called with AudioFlinger::mLock held 1064void AudioFlinger::removeClient_l(pid_t pid) 1065{ 1066 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1067 mClients.removeItem(pid); 1068} 1069 1070 1071// ---------------------------------------------------------------------------- 1072 1073AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1074 uint32_t device, type_t type) 1075 : Thread(false), 1076 mType(type), 1077 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1078 // mChannelMask 1079 mChannelCount(0), 1080 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1081 mParamStatus(NO_ERROR), 1082 mStandby(false), mId(id), 1083 mDevice(device), 1084 mDeathRecipient(new PMDeathRecipient(this)) 1085{ 1086} 1087 1088AudioFlinger::ThreadBase::~ThreadBase() 1089{ 1090 mParamCond.broadcast(); 1091 // do not lock the mutex in destructor 1092 releaseWakeLock_l(); 1093 if (mPowerManager != 0) { 1094 sp<IBinder> binder = mPowerManager->asBinder(); 1095 binder->unlinkToDeath(mDeathRecipient); 1096 } 1097} 1098 1099void AudioFlinger::ThreadBase::exit() 1100{ 1101 ALOGV("ThreadBase::exit"); 1102 { 1103 // This lock prevents the following race in thread (uniprocessor for illustration): 1104 // if (!exitPending()) { 1105 // // context switch from here to exit() 1106 // // exit() calls requestExit(), what exitPending() observes 1107 // // exit() calls signal(), which is dropped since no waiters 1108 // // context switch back from exit() to here 1109 // mWaitWorkCV.wait(...); 1110 // // now thread is hung 1111 // } 1112 AutoMutex lock(mLock); 1113 requestExit(); 1114 mWaitWorkCV.signal(); 1115 } 1116 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1117 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1118 requestExitAndWait(); 1119} 1120 1121status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1122{ 1123 status_t status; 1124 1125 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1126 Mutex::Autolock _l(mLock); 1127 1128 mNewParameters.add(keyValuePairs); 1129 mWaitWorkCV.signal(); 1130 // wait condition with timeout in case the thread loop has exited 1131 // before the request could be processed 1132 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1133 status = mParamStatus; 1134 mWaitWorkCV.signal(); 1135 } else { 1136 status = TIMED_OUT; 1137 } 1138 return status; 1139} 1140 1141void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1142{ 1143 Mutex::Autolock _l(mLock); 1144 sendConfigEvent_l(event, param); 1145} 1146 1147// sendConfigEvent_l() must be called with ThreadBase::mLock held 1148void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1149{ 1150 ConfigEvent configEvent; 1151 configEvent.mEvent = event; 1152 configEvent.mParam = param; 1153 mConfigEvents.add(configEvent); 1154 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1155 mWaitWorkCV.signal(); 1156} 1157 1158void AudioFlinger::ThreadBase::processConfigEvents() 1159{ 1160 mLock.lock(); 1161 while(!mConfigEvents.isEmpty()) { 1162 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1163 ConfigEvent configEvent = mConfigEvents[0]; 1164 mConfigEvents.removeAt(0); 1165 // release mLock before locking AudioFlinger mLock: lock order is always 1166 // AudioFlinger then ThreadBase to avoid cross deadlock 1167 mLock.unlock(); 1168 mAudioFlinger->mLock.lock(); 1169 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1170 mAudioFlinger->mLock.unlock(); 1171 mLock.lock(); 1172 } 1173 mLock.unlock(); 1174} 1175 1176status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1177{ 1178 const size_t SIZE = 256; 1179 char buffer[SIZE]; 1180 String8 result; 1181 1182 bool locked = tryLock(mLock); 1183 if (!locked) { 1184 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1185 write(fd, buffer, strlen(buffer)); 1186 } 1187 1188 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1189 result.append(buffer); 1190 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1191 result.append(buffer); 1192 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1193 result.append(buffer); 1194 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1197 result.append(buffer); 1198 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1199 result.append(buffer); 1200 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1201 result.append(buffer); 1202 1203 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1204 result.append(buffer); 1205 result.append(" Index Command"); 1206 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1207 snprintf(buffer, SIZE, "\n %02d ", i); 1208 result.append(buffer); 1209 result.append(mNewParameters[i]); 1210 } 1211 1212 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, " Index event param\n"); 1215 result.append(buffer); 1216 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1217 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1218 result.append(buffer); 1219 } 1220 result.append("\n"); 1221 1222 write(fd, result.string(), result.size()); 1223 1224 if (locked) { 1225 mLock.unlock(); 1226 } 1227 return NO_ERROR; 1228} 1229 1230status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1231{ 1232 const size_t SIZE = 256; 1233 char buffer[SIZE]; 1234 String8 result; 1235 1236 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1237 write(fd, buffer, strlen(buffer)); 1238 1239 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1240 sp<EffectChain> chain = mEffectChains[i]; 1241 if (chain != 0) { 1242 chain->dump(fd, args); 1243 } 1244 } 1245 return NO_ERROR; 1246} 1247 1248void AudioFlinger::ThreadBase::acquireWakeLock() 1249{ 1250 Mutex::Autolock _l(mLock); 1251 acquireWakeLock_l(); 1252} 1253 1254void AudioFlinger::ThreadBase::acquireWakeLock_l() 1255{ 1256 if (mPowerManager == 0) { 1257 // use checkService() to avoid blocking if power service is not up yet 1258 sp<IBinder> binder = 1259 defaultServiceManager()->checkService(String16("power")); 1260 if (binder == 0) { 1261 ALOGW("Thread %s cannot connect to the power manager service", mName); 1262 } else { 1263 mPowerManager = interface_cast<IPowerManager>(binder); 1264 binder->linkToDeath(mDeathRecipient); 1265 } 1266 } 1267 if (mPowerManager != 0) { 1268 sp<IBinder> binder = new BBinder(); 1269 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1270 binder, 1271 String16(mName)); 1272 if (status == NO_ERROR) { 1273 mWakeLockToken = binder; 1274 } 1275 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1276 } 1277} 1278 1279void AudioFlinger::ThreadBase::releaseWakeLock() 1280{ 1281 Mutex::Autolock _l(mLock); 1282 releaseWakeLock_l(); 1283} 1284 1285void AudioFlinger::ThreadBase::releaseWakeLock_l() 1286{ 1287 if (mWakeLockToken != 0) { 1288 ALOGV("releaseWakeLock_l() %s", mName); 1289 if (mPowerManager != 0) { 1290 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1291 } 1292 mWakeLockToken.clear(); 1293 } 1294} 1295 1296void AudioFlinger::ThreadBase::clearPowerManager() 1297{ 1298 Mutex::Autolock _l(mLock); 1299 releaseWakeLock_l(); 1300 mPowerManager.clear(); 1301} 1302 1303void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1304{ 1305 sp<ThreadBase> thread = mThread.promote(); 1306 if (thread != 0) { 1307 thread->clearPowerManager(); 1308 } 1309 ALOGW("power manager service died !!!"); 1310} 1311 1312void AudioFlinger::ThreadBase::setEffectSuspended( 1313 const effect_uuid_t *type, bool suspend, int sessionId) 1314{ 1315 Mutex::Autolock _l(mLock); 1316 setEffectSuspended_l(type, suspend, sessionId); 1317} 1318 1319void AudioFlinger::ThreadBase::setEffectSuspended_l( 1320 const effect_uuid_t *type, bool suspend, int sessionId) 1321{ 1322 sp<EffectChain> chain = getEffectChain_l(sessionId); 1323 if (chain != 0) { 1324 if (type != NULL) { 1325 chain->setEffectSuspended_l(type, suspend); 1326 } else { 1327 chain->setEffectSuspendedAll_l(suspend); 1328 } 1329 } 1330 1331 updateSuspendedSessions_l(type, suspend, sessionId); 1332} 1333 1334void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1335{ 1336 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1337 if (index < 0) { 1338 return; 1339 } 1340 1341 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1342 mSuspendedSessions.editValueAt(index); 1343 1344 for (size_t i = 0; i < sessionEffects.size(); i++) { 1345 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1346 for (int j = 0; j < desc->mRefCount; j++) { 1347 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1348 chain->setEffectSuspendedAll_l(true); 1349 } else { 1350 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1351 desc->mType.timeLow); 1352 chain->setEffectSuspended_l(&desc->mType, true); 1353 } 1354 } 1355 } 1356} 1357 1358void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1359 bool suspend, 1360 int sessionId) 1361{ 1362 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1363 1364 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1365 1366 if (suspend) { 1367 if (index >= 0) { 1368 sessionEffects = mSuspendedSessions.editValueAt(index); 1369 } else { 1370 mSuspendedSessions.add(sessionId, sessionEffects); 1371 } 1372 } else { 1373 if (index < 0) { 1374 return; 1375 } 1376 sessionEffects = mSuspendedSessions.editValueAt(index); 1377 } 1378 1379 1380 int key = EffectChain::kKeyForSuspendAll; 1381 if (type != NULL) { 1382 key = type->timeLow; 1383 } 1384 index = sessionEffects.indexOfKey(key); 1385 1386 sp <SuspendedSessionDesc> desc; 1387 if (suspend) { 1388 if (index >= 0) { 1389 desc = sessionEffects.valueAt(index); 1390 } else { 1391 desc = new SuspendedSessionDesc(); 1392 if (type != NULL) { 1393 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1394 } 1395 sessionEffects.add(key, desc); 1396 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1397 } 1398 desc->mRefCount++; 1399 } else { 1400 if (index < 0) { 1401 return; 1402 } 1403 desc = sessionEffects.valueAt(index); 1404 if (--desc->mRefCount == 0) { 1405 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1406 sessionEffects.removeItemsAt(index); 1407 if (sessionEffects.isEmpty()) { 1408 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1409 sessionId); 1410 mSuspendedSessions.removeItem(sessionId); 1411 } 1412 } 1413 } 1414 if (!sessionEffects.isEmpty()) { 1415 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1416 } 1417} 1418 1419void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1420 bool enabled, 1421 int sessionId) 1422{ 1423 Mutex::Autolock _l(mLock); 1424 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1425} 1426 1427void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1428 bool enabled, 1429 int sessionId) 1430{ 1431 if (mType != RECORD) { 1432 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1433 // another session. This gives the priority to well behaved effect control panels 1434 // and applications not using global effects. 1435 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1436 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1437 } 1438 } 1439 1440 sp<EffectChain> chain = getEffectChain_l(sessionId); 1441 if (chain != 0) { 1442 chain->checkSuspendOnEffectEnabled(effect, enabled); 1443 } 1444} 1445 1446// ---------------------------------------------------------------------------- 1447 1448AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1449 AudioStreamOut* output, 1450 audio_io_handle_t id, 1451 uint32_t device, 1452 type_t type) 1453 : ThreadBase(audioFlinger, id, device, type), 1454 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1455 // Assumes constructor is called by AudioFlinger with it's mLock held, 1456 // but it would be safer to explicitly pass initial masterMute as parameter 1457 mMasterMute(audioFlinger->masterMute_l()), 1458 // mStreamTypes[] initialized in constructor body 1459 mOutput(output), 1460 // Assumes constructor is called by AudioFlinger with it's mLock held, 1461 // but it would be safer to explicitly pass initial masterVolume as parameter 1462 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1463 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1464{ 1465 snprintf(mName, kNameLength, "AudioOut_%d", id); 1466 1467 readOutputParameters(); 1468 1469 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1470 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1471 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1472 stream = (audio_stream_type_t) (stream + 1)) { 1473 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1474 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1475 // initialized by stream_type_t default constructor 1476 // mStreamTypes[stream].valid = true; 1477 } 1478 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1479 // because mAudioFlinger doesn't have one to copy from 1480} 1481 1482AudioFlinger::PlaybackThread::~PlaybackThread() 1483{ 1484 delete [] mMixBuffer; 1485} 1486 1487status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1488{ 1489 dumpInternals(fd, args); 1490 dumpTracks(fd, args); 1491 dumpEffectChains(fd, args); 1492 return NO_ERROR; 1493} 1494 1495status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1496{ 1497 const size_t SIZE = 256; 1498 char buffer[SIZE]; 1499 String8 result; 1500 1501 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1502 result.append(buffer); 1503 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1504 for (size_t i = 0; i < mTracks.size(); ++i) { 1505 sp<Track> track = mTracks[i]; 1506 if (track != 0) { 1507 track->dump(buffer, SIZE); 1508 result.append(buffer); 1509 } 1510 } 1511 1512 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1513 result.append(buffer); 1514 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1515 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1516 sp<Track> track = mActiveTracks[i].promote(); 1517 if (track != 0) { 1518 track->dump(buffer, SIZE); 1519 result.append(buffer); 1520 } 1521 } 1522 write(fd, result.string(), result.size()); 1523 return NO_ERROR; 1524} 1525 1526status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1527{ 1528 const size_t SIZE = 256; 1529 char buffer[SIZE]; 1530 String8 result; 1531 1532 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1533 result.append(buffer); 1534 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1535 result.append(buffer); 1536 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1537 result.append(buffer); 1538 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1541 result.append(buffer); 1542 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1545 result.append(buffer); 1546 write(fd, result.string(), result.size()); 1547 1548 dumpBase(fd, args); 1549 1550 return NO_ERROR; 1551} 1552 1553// Thread virtuals 1554status_t AudioFlinger::PlaybackThread::readyToRun() 1555{ 1556 status_t status = initCheck(); 1557 if (status == NO_ERROR) { 1558 ALOGI("AudioFlinger's thread %p ready to run", this); 1559 } else { 1560 ALOGE("No working audio driver found."); 1561 } 1562 return status; 1563} 1564 1565void AudioFlinger::PlaybackThread::onFirstRef() 1566{ 1567 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1568} 1569 1570// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1571sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1572 const sp<AudioFlinger::Client>& client, 1573 audio_stream_type_t streamType, 1574 uint32_t sampleRate, 1575 audio_format_t format, 1576 uint32_t channelMask, 1577 int frameCount, 1578 const sp<IMemory>& sharedBuffer, 1579 int sessionId, 1580 bool isTimed, 1581 status_t *status) 1582{ 1583 sp<Track> track; 1584 status_t lStatus; 1585 1586 if (mType == DIRECT) { 1587 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1588 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1589 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1590 "for output %p with format %d", 1591 sampleRate, format, channelMask, mOutput, mFormat); 1592 lStatus = BAD_VALUE; 1593 goto Exit; 1594 } 1595 } 1596 } else { 1597 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1598 if (sampleRate > mSampleRate*2) { 1599 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1600 lStatus = BAD_VALUE; 1601 goto Exit; 1602 } 1603 } 1604 1605 lStatus = initCheck(); 1606 if (lStatus != NO_ERROR) { 1607 ALOGE("Audio driver not initialized."); 1608 goto Exit; 1609 } 1610 1611 { // scope for mLock 1612 Mutex::Autolock _l(mLock); 1613 1614 // all tracks in same audio session must share the same routing strategy otherwise 1615 // conflicts will happen when tracks are moved from one output to another by audio policy 1616 // manager 1617 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1618 for (size_t i = 0; i < mTracks.size(); ++i) { 1619 sp<Track> t = mTracks[i]; 1620 if (t != 0) { 1621 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1622 if (sessionId == t->sessionId() && strategy != actual) { 1623 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1624 strategy, actual); 1625 lStatus = BAD_VALUE; 1626 goto Exit; 1627 } 1628 } 1629 } 1630 1631 if (!isTimed) { 1632 track = new Track(this, client, streamType, sampleRate, format, 1633 channelMask, frameCount, sharedBuffer, sessionId); 1634 } else { 1635 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1636 channelMask, frameCount, sharedBuffer, sessionId); 1637 } 1638 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1639 lStatus = NO_MEMORY; 1640 goto Exit; 1641 } 1642 mTracks.add(track); 1643 1644 sp<EffectChain> chain = getEffectChain_l(sessionId); 1645 if (chain != 0) { 1646 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1647 track->setMainBuffer(chain->inBuffer()); 1648 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1649 chain->incTrackCnt(); 1650 } 1651 1652 // invalidate track immediately if the stream type was moved to another thread since 1653 // createTrack() was called by the client process. 1654 if (!mStreamTypes[streamType].valid) { 1655 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1656 this, streamType); 1657 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1658 } 1659 } 1660 lStatus = NO_ERROR; 1661 1662Exit: 1663 if(status) { 1664 *status = lStatus; 1665 } 1666 return track; 1667} 1668 1669uint32_t AudioFlinger::PlaybackThread::latency() const 1670{ 1671 Mutex::Autolock _l(mLock); 1672 if (initCheck() == NO_ERROR) { 1673 return mOutput->stream->get_latency(mOutput->stream); 1674 } else { 1675 return 0; 1676 } 1677} 1678 1679void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1680{ 1681 Mutex::Autolock _l(mLock); 1682 mMasterVolume = value; 1683} 1684 1685void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1686{ 1687 Mutex::Autolock _l(mLock); 1688 setMasterMute_l(muted); 1689} 1690 1691void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1692{ 1693 Mutex::Autolock _l(mLock); 1694 mStreamTypes[stream].volume = value; 1695} 1696 1697void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1698{ 1699 Mutex::Autolock _l(mLock); 1700 mStreamTypes[stream].mute = muted; 1701} 1702 1703float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1704{ 1705 Mutex::Autolock _l(mLock); 1706 return mStreamTypes[stream].volume; 1707} 1708 1709// addTrack_l() must be called with ThreadBase::mLock held 1710status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1711{ 1712 status_t status = ALREADY_EXISTS; 1713 1714 // set retry count for buffer fill 1715 track->mRetryCount = kMaxTrackStartupRetries; 1716 if (mActiveTracks.indexOf(track) < 0) { 1717 // the track is newly added, make sure it fills up all its 1718 // buffers before playing. This is to ensure the client will 1719 // effectively get the latency it requested. 1720 track->mFillingUpStatus = Track::FS_FILLING; 1721 track->mResetDone = false; 1722 mActiveTracks.add(track); 1723 if (track->mainBuffer() != mMixBuffer) { 1724 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1725 if (chain != 0) { 1726 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1727 chain->incActiveTrackCnt(); 1728 } 1729 } 1730 1731 status = NO_ERROR; 1732 } 1733 1734 ALOGV("mWaitWorkCV.broadcast"); 1735 mWaitWorkCV.broadcast(); 1736 1737 return status; 1738} 1739 1740// destroyTrack_l() must be called with ThreadBase::mLock held 1741void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1742{ 1743 track->mState = TrackBase::TERMINATED; 1744 if (mActiveTracks.indexOf(track) < 0) { 1745 removeTrack_l(track); 1746 } 1747} 1748 1749void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1750{ 1751 mTracks.remove(track); 1752 deleteTrackName_l(track->name()); 1753 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1754 if (chain != 0) { 1755 chain->decTrackCnt(); 1756 } 1757} 1758 1759String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1760{ 1761 String8 out_s8 = String8(""); 1762 char *s; 1763 1764 Mutex::Autolock _l(mLock); 1765 if (initCheck() != NO_ERROR) { 1766 return out_s8; 1767 } 1768 1769 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1770 out_s8 = String8(s); 1771 free(s); 1772 return out_s8; 1773} 1774 1775// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1776void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1777 AudioSystem::OutputDescriptor desc; 1778 void *param2 = NULL; 1779 1780 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1781 1782 switch (event) { 1783 case AudioSystem::OUTPUT_OPENED: 1784 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1785 desc.channels = mChannelMask; 1786 desc.samplingRate = mSampleRate; 1787 desc.format = mFormat; 1788 desc.frameCount = mFrameCount; 1789 desc.latency = latency(); 1790 param2 = &desc; 1791 break; 1792 1793 case AudioSystem::STREAM_CONFIG_CHANGED: 1794 param2 = ¶m; 1795 case AudioSystem::OUTPUT_CLOSED: 1796 default: 1797 break; 1798 } 1799 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1800} 1801 1802void AudioFlinger::PlaybackThread::readOutputParameters() 1803{ 1804 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1805 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1806 mChannelCount = (uint16_t)popcount(mChannelMask); 1807 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1808 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1809 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1810 1811 // FIXME - Current mixer implementation only supports stereo output: Always 1812 // Allocate a stereo buffer even if HW output is mono. 1813 delete[] mMixBuffer; 1814 mMixBuffer = new int16_t[mFrameCount * 2]; 1815 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1816 1817 // force reconfiguration of effect chains and engines to take new buffer size and audio 1818 // parameters into account 1819 // Note that mLock is not held when readOutputParameters() is called from the constructor 1820 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1821 // matter. 1822 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1823 Vector< sp<EffectChain> > effectChains = mEffectChains; 1824 for (size_t i = 0; i < effectChains.size(); i ++) { 1825 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1826 } 1827} 1828 1829status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1830{ 1831 if (halFrames == NULL || dspFrames == NULL) { 1832 return BAD_VALUE; 1833 } 1834 Mutex::Autolock _l(mLock); 1835 if (initCheck() != NO_ERROR) { 1836 return INVALID_OPERATION; 1837 } 1838 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1839 1840 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1841} 1842 1843uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1844{ 1845 Mutex::Autolock _l(mLock); 1846 uint32_t result = 0; 1847 if (getEffectChain_l(sessionId) != 0) { 1848 result = EFFECT_SESSION; 1849 } 1850 1851 for (size_t i = 0; i < mTracks.size(); ++i) { 1852 sp<Track> track = mTracks[i]; 1853 if (sessionId == track->sessionId() && 1854 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1855 result |= TRACK_SESSION; 1856 break; 1857 } 1858 } 1859 1860 return result; 1861} 1862 1863uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1864{ 1865 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1866 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1867 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1868 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1869 } 1870 for (size_t i = 0; i < mTracks.size(); i++) { 1871 sp<Track> track = mTracks[i]; 1872 if (sessionId == track->sessionId() && 1873 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1874 return AudioSystem::getStrategyForStream(track->streamType()); 1875 } 1876 } 1877 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1878} 1879 1880 1881AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1882{ 1883 Mutex::Autolock _l(mLock); 1884 return mOutput; 1885} 1886 1887AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1888{ 1889 Mutex::Autolock _l(mLock); 1890 AudioStreamOut *output = mOutput; 1891 mOutput = NULL; 1892 return output; 1893} 1894 1895// this method must always be called either with ThreadBase mLock held or inside the thread loop 1896audio_stream_t* AudioFlinger::PlaybackThread::stream() 1897{ 1898 if (mOutput == NULL) { 1899 return NULL; 1900 } 1901 return &mOutput->stream->common; 1902} 1903 1904uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1905{ 1906 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1907 // decoding and transfer time. So sleeping for half of the latency would likely cause 1908 // underruns 1909 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1910 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1911 } else { 1912 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1913 } 1914} 1915 1916// ---------------------------------------------------------------------------- 1917 1918AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1919 audio_io_handle_t id, uint32_t device, type_t type) 1920 : PlaybackThread(audioFlinger, output, id, device, type), 1921 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1922 mPrevMixerStatus(MIXER_IDLE) 1923{ 1924 // FIXME - Current mixer implementation only supports stereo output 1925 if (mChannelCount == 1) { 1926 ALOGE("Invalid audio hardware channel count"); 1927 } 1928} 1929 1930AudioFlinger::MixerThread::~MixerThread() 1931{ 1932 delete mAudioMixer; 1933} 1934 1935bool AudioFlinger::MixerThread::threadLoop() 1936{ 1937 Vector< sp<Track> > tracksToRemove; 1938 mixer_state mixerStatus = MIXER_IDLE; 1939 nsecs_t standbyTime = systemTime(); 1940 size_t mixBufferSize = mFrameCount * mFrameSize; 1941 // FIXME: Relaxed timing because of a certain device that can't meet latency 1942 // Should be reduced to 2x after the vendor fixes the driver issue 1943 // increase threshold again due to low power audio mode. The way this warning threshold is 1944 // calculated and its usefulness should be reconsidered anyway. 1945 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1946 nsecs_t lastWarning = 0; 1947 bool longStandbyExit = false; 1948 uint32_t activeSleepTime = activeSleepTimeUs(); 1949 uint32_t idleSleepTime = idleSleepTimeUs(); 1950 uint32_t sleepTime = idleSleepTime; 1951 uint32_t sleepTimeShift = 0; 1952 Vector< sp<EffectChain> > effectChains; 1953#ifdef DEBUG_CPU_USAGE 1954 ThreadCpuUsage cpu; 1955 const CentralTendencyStatistics& stats = cpu.statistics(); 1956#endif 1957 1958 acquireWakeLock(); 1959 1960 while (!exitPending()) 1961 { 1962#ifdef DEBUG_CPU_USAGE 1963 cpu.sampleAndEnable(); 1964 unsigned n = stats.n(); 1965 // cpu.elapsed() is expensive, so don't call it every loop 1966 if ((n & 127) == 1) { 1967 long long elapsed = cpu.elapsed(); 1968 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1969 double perLoop = elapsed / (double) n; 1970 double perLoop100 = perLoop * 0.01; 1971 double mean = stats.mean(); 1972 double stddev = stats.stddev(); 1973 double minimum = stats.minimum(); 1974 double maximum = stats.maximum(); 1975 cpu.resetStatistics(); 1976 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", 1977 elapsed * .000000001, n, perLoop * .000001, 1978 mean * .001, 1979 stddev * .001, 1980 minimum * .001, 1981 maximum * .001, 1982 mean / perLoop100, 1983 stddev / perLoop100, 1984 minimum / perLoop100, 1985 maximum / perLoop100); 1986 } 1987 } 1988#endif 1989 processConfigEvents(); 1990 1991 mixerStatus = MIXER_IDLE; 1992 { // scope for mLock 1993 1994 Mutex::Autolock _l(mLock); 1995 1996 if (checkForNewParameters_l()) { 1997 mixBufferSize = mFrameCount * mFrameSize; 1998 // FIXME: Relaxed timing because of a certain device that can't meet latency 1999 // Should be reduced to 2x after the vendor fixes the driver issue 2000 // increase threshold again due to low power audio mode. The way this warning 2001 // threshold is calculated and its usefulness should be reconsidered anyway. 2002 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2003 activeSleepTime = activeSleepTimeUs(); 2004 idleSleepTime = idleSleepTimeUs(); 2005 } 2006 2007 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2008 2009 // put audio hardware into standby after short delay 2010 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2011 mSuspended)) { 2012 if (!mStandby) { 2013 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2014 mOutput->stream->common.standby(&mOutput->stream->common); 2015 mStandby = true; 2016 mBytesWritten = 0; 2017 } 2018 2019 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2020 // we're about to wait, flush the binder command buffer 2021 IPCThreadState::self()->flushCommands(); 2022 2023 if (exitPending()) break; 2024 2025 releaseWakeLock_l(); 2026 // wait until we have something to do... 2027 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 2028 mWaitWorkCV.wait(mLock); 2029 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 2030 acquireWakeLock_l(); 2031 2032 mPrevMixerStatus = MIXER_IDLE; 2033 if (!mMasterMute) { 2034 char value[PROPERTY_VALUE_MAX]; 2035 property_get("ro.audio.silent", value, "0"); 2036 if (atoi(value)) { 2037 ALOGD("Silence is golden"); 2038 setMasterMute_l(true); 2039 } 2040 } 2041 2042 standbyTime = systemTime() + mStandbyTimeInNsecs; 2043 sleepTime = idleSleepTime; 2044 sleepTimeShift = 0; 2045 continue; 2046 } 2047 } 2048 2049 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2050 2051 // prevent any changes in effect chain list and in each effect chain 2052 // during mixing and effect process as the audio buffers could be deleted 2053 // or modified if an effect is created or deleted 2054 lockEffectChains_l(effectChains); 2055 } 2056 2057 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2058 // obtain the presentation timestamp of the next output buffer 2059 int64_t pts; 2060 status_t status = INVALID_OPERATION; 2061 2062 if (NULL != mOutput->stream->get_next_write_timestamp) { 2063 status = mOutput->stream->get_next_write_timestamp( 2064 mOutput->stream, &pts); 2065 } 2066 2067 if (status != NO_ERROR) { 2068 pts = AudioBufferProvider::kInvalidPTS; 2069 } 2070 2071 // mix buffers... 2072 mAudioMixer->process(pts); 2073 // increase sleep time progressively when application underrun condition clears. 2074 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2075 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2076 // such that we would underrun the audio HAL. 2077 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2078 sleepTimeShift--; 2079 } 2080 sleepTime = 0; 2081 standbyTime = systemTime() + mStandbyTimeInNsecs; 2082 //TODO: delay standby when effects have a tail 2083 } else { 2084 // If no tracks are ready, sleep once for the duration of an output 2085 // buffer size, then write 0s to the output 2086 if (sleepTime == 0) { 2087 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2088 sleepTime = activeSleepTime >> sleepTimeShift; 2089 if (sleepTime < kMinThreadSleepTimeUs) { 2090 sleepTime = kMinThreadSleepTimeUs; 2091 } 2092 // reduce sleep time in case of consecutive application underruns to avoid 2093 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2094 // duration we would end up writing less data than needed by the audio HAL if 2095 // the condition persists. 2096 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2097 sleepTimeShift++; 2098 } 2099 } else { 2100 sleepTime = idleSleepTime; 2101 } 2102 } else if (mBytesWritten != 0 || 2103 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2104 memset (mMixBuffer, 0, mixBufferSize); 2105 sleepTime = 0; 2106 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2107 } 2108 // TODO add standby time extension fct of effect tail 2109 } 2110 2111 if (mSuspended) { 2112 sleepTime = suspendSleepTimeUs(); 2113 } 2114 // sleepTime == 0 means we must write to audio hardware 2115 if (sleepTime == 0) { 2116 for (size_t i = 0; i < effectChains.size(); i ++) { 2117 effectChains[i]->process_l(); 2118 } 2119 // enable changes in effect chain 2120 unlockEffectChains(effectChains); 2121 mLastWriteTime = systemTime(); 2122 mInWrite = true; 2123 mBytesWritten += mixBufferSize; 2124 2125 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2126 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2127 mNumWrites++; 2128 mInWrite = false; 2129 nsecs_t now = systemTime(); 2130 nsecs_t delta = now - mLastWriteTime; 2131 if (!mStandby && delta > maxPeriod) { 2132 mNumDelayedWrites++; 2133 if ((now - lastWarning) > kWarningThrottleNs) { 2134 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2135 ns2ms(delta), mNumDelayedWrites, this); 2136 lastWarning = now; 2137 } 2138 if (mStandby) { 2139 longStandbyExit = true; 2140 } 2141 } 2142 mStandby = false; 2143 } else { 2144 // enable changes in effect chain 2145 unlockEffectChains(effectChains); 2146 usleep(sleepTime); 2147 } 2148 2149 // finally let go of all our tracks, without the lock held 2150 // since we can't guarantee the destructors won't acquire that 2151 // same lock. 2152 tracksToRemove.clear(); 2153 2154 // Effect chains will be actually deleted here if they were removed from 2155 // mEffectChains list during mixing or effects processing 2156 effectChains.clear(); 2157 } 2158 2159 if (!mStandby) { 2160 mOutput->stream->common.standby(&mOutput->stream->common); 2161 } 2162 2163 releaseWakeLock(); 2164 2165 ALOGV("MixerThread %p exiting", this); 2166 return false; 2167} 2168 2169// prepareTracks_l() must be called with ThreadBase::mLock held 2170AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2171 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2172{ 2173 2174 mixer_state mixerStatus = MIXER_IDLE; 2175 // find out which tracks need to be processed 2176 size_t count = activeTracks.size(); 2177 size_t mixedTracks = 0; 2178 size_t tracksWithEffect = 0; 2179 2180 float masterVolume = mMasterVolume; 2181 bool masterMute = mMasterMute; 2182 2183 if (masterMute) { 2184 masterVolume = 0; 2185 } 2186 // Delegate master volume control to effect in output mix effect chain if needed 2187 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2188 if (chain != 0) { 2189 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2190 chain->setVolume_l(&v, &v); 2191 masterVolume = (float)((v + (1 << 23)) >> 24); 2192 chain.clear(); 2193 } 2194 2195 for (size_t i=0 ; i<count ; i++) { 2196 sp<Track> t = activeTracks[i].promote(); 2197 if (t == 0) continue; 2198 2199 // this const just means the local variable doesn't change 2200 Track* const track = t.get(); 2201 audio_track_cblk_t* cblk = track->cblk(); 2202 2203 // The first time a track is added we wait 2204 // for all its buffers to be filled before processing it 2205 int name = track->name(); 2206 // make sure that we have enough frames to mix one full buffer. 2207 // enforce this condition only once to enable draining the buffer in case the client 2208 // app does not call stop() and relies on underrun to stop: 2209 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2210 // during last round 2211 uint32_t minFrames = 1; 2212 if (!track->isStopped() && !track->isPausing() && 2213 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2214 if (t->sampleRate() == (int)mSampleRate) { 2215 minFrames = mFrameCount; 2216 } else { 2217 // +1 for rounding and +1 for additional sample needed for interpolation 2218 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2219 // add frames already consumed but not yet released by the resampler 2220 // because cblk->framesReady() will include these frames 2221 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2222 // the minimum track buffer size is normally twice the number of frames necessary 2223 // to fill one buffer and the resampler should not leave more than one buffer worth 2224 // of unreleased frames after each pass, but just in case... 2225 ALOG_ASSERT(minFrames <= cblk->frameCount); 2226 } 2227 } 2228 if ((track->framesReady() >= minFrames) && track->isReady() && 2229 !track->isPaused() && !track->isTerminated()) 2230 { 2231 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2232 2233 mixedTracks++; 2234 2235 // track->mainBuffer() != mMixBuffer means there is an effect chain 2236 // connected to the track 2237 chain.clear(); 2238 if (track->mainBuffer() != mMixBuffer) { 2239 chain = getEffectChain_l(track->sessionId()); 2240 // Delegate volume control to effect in track effect chain if needed 2241 if (chain != 0) { 2242 tracksWithEffect++; 2243 } else { 2244 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2245 name, track->sessionId()); 2246 } 2247 } 2248 2249 2250 int param = AudioMixer::VOLUME; 2251 if (track->mFillingUpStatus == Track::FS_FILLED) { 2252 // no ramp for the first volume setting 2253 track->mFillingUpStatus = Track::FS_ACTIVE; 2254 if (track->mState == TrackBase::RESUMING) { 2255 track->mState = TrackBase::ACTIVE; 2256 param = AudioMixer::RAMP_VOLUME; 2257 } 2258 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2259 } else if (cblk->server != 0) { 2260 // If the track is stopped before the first frame was mixed, 2261 // do not apply ramp 2262 param = AudioMixer::RAMP_VOLUME; 2263 } 2264 2265 // compute volume for this track 2266 uint32_t vl, vr, va; 2267 if (track->isMuted() || track->isPausing() || 2268 mStreamTypes[track->streamType()].mute) { 2269 vl = vr = va = 0; 2270 if (track->isPausing()) { 2271 track->setPaused(); 2272 } 2273 } else { 2274 2275 // read original volumes with volume control 2276 float typeVolume = mStreamTypes[track->streamType()].volume; 2277 float v = masterVolume * typeVolume; 2278 uint32_t vlr = cblk->getVolumeLR(); 2279 vl = vlr & 0xFFFF; 2280 vr = vlr >> 16; 2281 // track volumes come from shared memory, so can't be trusted and must be clamped 2282 if (vl > MAX_GAIN_INT) { 2283 ALOGV("Track left volume out of range: %04X", vl); 2284 vl = MAX_GAIN_INT; 2285 } 2286 if (vr > MAX_GAIN_INT) { 2287 ALOGV("Track right volume out of range: %04X", vr); 2288 vr = MAX_GAIN_INT; 2289 } 2290 // now apply the master volume and stream type volume 2291 vl = (uint32_t)(v * vl) << 12; 2292 vr = (uint32_t)(v * vr) << 12; 2293 // assuming master volume and stream type volume each go up to 1.0, 2294 // vl and vr are now in 8.24 format 2295 2296 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2297 // send level comes from shared memory and so may be corrupt 2298 if (sendLevel > MAX_GAIN_INT) { 2299 ALOGV("Track send level out of range: %04X", sendLevel); 2300 sendLevel = MAX_GAIN_INT; 2301 } 2302 va = (uint32_t)(v * sendLevel); 2303 } 2304 // Delegate volume control to effect in track effect chain if needed 2305 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2306 // Do not ramp volume if volume is controlled by effect 2307 param = AudioMixer::VOLUME; 2308 track->mHasVolumeController = true; 2309 } else { 2310 // force no volume ramp when volume controller was just disabled or removed 2311 // from effect chain to avoid volume spike 2312 if (track->mHasVolumeController) { 2313 param = AudioMixer::VOLUME; 2314 } 2315 track->mHasVolumeController = false; 2316 } 2317 2318 // Convert volumes from 8.24 to 4.12 format 2319 // This additional clamping is needed in case chain->setVolume_l() overshot 2320 vl = (vl + (1 << 11)) >> 12; 2321 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2322 vr = (vr + (1 << 11)) >> 12; 2323 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2324 2325 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2326 2327 // XXX: these things DON'T need to be done each time 2328 mAudioMixer->setBufferProvider(name, track); 2329 mAudioMixer->enable(name); 2330 2331 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2332 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2333 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2334 mAudioMixer->setParameter( 2335 name, 2336 AudioMixer::TRACK, 2337 AudioMixer::FORMAT, (void *)track->format()); 2338 mAudioMixer->setParameter( 2339 name, 2340 AudioMixer::TRACK, 2341 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2342 mAudioMixer->setParameter( 2343 name, 2344 AudioMixer::RESAMPLE, 2345 AudioMixer::SAMPLE_RATE, 2346 (void *)(cblk->sampleRate)); 2347 mAudioMixer->setParameter( 2348 name, 2349 AudioMixer::TRACK, 2350 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2351 mAudioMixer->setParameter( 2352 name, 2353 AudioMixer::TRACK, 2354 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2355 2356 // reset retry count 2357 track->mRetryCount = kMaxTrackRetries; 2358 // If one track is ready, set the mixer ready if: 2359 // - the mixer was not ready during previous round OR 2360 // - no other track is not ready 2361 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2362 mixerStatus != MIXER_TRACKS_ENABLED) { 2363 mixerStatus = MIXER_TRACKS_READY; 2364 } 2365 } else { 2366 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2367 if (track->isStopped()) { 2368 track->reset(); 2369 } 2370 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2371 // We have consumed all the buffers of this track. 2372 // Remove it from the list of active tracks. 2373 tracksToRemove->add(track); 2374 } else { 2375 // No buffers for this track. Give it a few chances to 2376 // fill a buffer, then remove it from active list. 2377 if (--(track->mRetryCount) <= 0) { 2378 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2379 tracksToRemove->add(track); 2380 // indicate to client process that the track was disabled because of underrun 2381 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2382 // If one track is not ready, mark the mixer also not ready if: 2383 // - the mixer was ready during previous round OR 2384 // - no other track is ready 2385 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2386 mixerStatus != MIXER_TRACKS_READY) { 2387 mixerStatus = MIXER_TRACKS_ENABLED; 2388 } 2389 } 2390 mAudioMixer->disable(name); 2391 } 2392 } 2393 2394 // remove all the tracks that need to be... 2395 count = tracksToRemove->size(); 2396 if (CC_UNLIKELY(count)) { 2397 for (size_t i=0 ; i<count ; i++) { 2398 const sp<Track>& track = tracksToRemove->itemAt(i); 2399 mActiveTracks.remove(track); 2400 if (track->mainBuffer() != mMixBuffer) { 2401 chain = getEffectChain_l(track->sessionId()); 2402 if (chain != 0) { 2403 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2404 chain->decActiveTrackCnt(); 2405 } 2406 } 2407 if (track->isTerminated()) { 2408 removeTrack_l(track); 2409 } 2410 } 2411 } 2412 2413 // mix buffer must be cleared if all tracks are connected to an 2414 // effect chain as in this case the mixer will not write to 2415 // mix buffer and track effects will accumulate into it 2416 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2417 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2418 } 2419 2420 mPrevMixerStatus = mixerStatus; 2421 return mixerStatus; 2422} 2423 2424void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2425{ 2426 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2427 this, streamType, mTracks.size()); 2428 Mutex::Autolock _l(mLock); 2429 2430 size_t size = mTracks.size(); 2431 for (size_t i = 0; i < size; i++) { 2432 sp<Track> t = mTracks[i]; 2433 if (t->streamType() == streamType) { 2434 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2435 t->mCblk->cv.signal(); 2436 } 2437 } 2438} 2439 2440void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2441{ 2442 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2443 this, streamType, valid); 2444 Mutex::Autolock _l(mLock); 2445 2446 mStreamTypes[streamType].valid = valid; 2447} 2448 2449// getTrackName_l() must be called with ThreadBase::mLock held 2450int AudioFlinger::MixerThread::getTrackName_l() 2451{ 2452 return mAudioMixer->getTrackName(); 2453} 2454 2455// deleteTrackName_l() must be called with ThreadBase::mLock held 2456void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2457{ 2458 ALOGV("remove track (%d) and delete from mixer", name); 2459 mAudioMixer->deleteTrackName(name); 2460} 2461 2462// checkForNewParameters_l() must be called with ThreadBase::mLock held 2463bool AudioFlinger::MixerThread::checkForNewParameters_l() 2464{ 2465 bool reconfig = false; 2466 2467 while (!mNewParameters.isEmpty()) { 2468 status_t status = NO_ERROR; 2469 String8 keyValuePair = mNewParameters[0]; 2470 AudioParameter param = AudioParameter(keyValuePair); 2471 int value; 2472 2473 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2474 reconfig = true; 2475 } 2476 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2477 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2478 status = BAD_VALUE; 2479 } else { 2480 reconfig = true; 2481 } 2482 } 2483 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2484 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2485 status = BAD_VALUE; 2486 } else { 2487 reconfig = true; 2488 } 2489 } 2490 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2491 // do not accept frame count changes if tracks are open as the track buffer 2492 // size depends on frame count and correct behavior would not be guaranteed 2493 // if frame count is changed after track creation 2494 if (!mTracks.isEmpty()) { 2495 status = INVALID_OPERATION; 2496 } else { 2497 reconfig = true; 2498 } 2499 } 2500 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2501 // when changing the audio output device, call addBatteryData to notify 2502 // the change 2503 if ((int)mDevice != value) { 2504 uint32_t params = 0; 2505 // check whether speaker is on 2506 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2507 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2508 } 2509 2510 int deviceWithoutSpeaker 2511 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2512 // check if any other device (except speaker) is on 2513 if (value & deviceWithoutSpeaker ) { 2514 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2515 } 2516 2517 if (params != 0) { 2518 addBatteryData(params); 2519 } 2520 } 2521 2522 // forward device change to effects that have requested to be 2523 // aware of attached audio device. 2524 mDevice = (uint32_t)value; 2525 for (size_t i = 0; i < mEffectChains.size(); i++) { 2526 mEffectChains[i]->setDevice_l(mDevice); 2527 } 2528 } 2529 2530 if (status == NO_ERROR) { 2531 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2532 keyValuePair.string()); 2533 if (!mStandby && status == INVALID_OPERATION) { 2534 mOutput->stream->common.standby(&mOutput->stream->common); 2535 mStandby = true; 2536 mBytesWritten = 0; 2537 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2538 keyValuePair.string()); 2539 } 2540 if (status == NO_ERROR && reconfig) { 2541 delete mAudioMixer; 2542 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2543 mAudioMixer = NULL; 2544 readOutputParameters(); 2545 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2546 for (size_t i = 0; i < mTracks.size() ; i++) { 2547 int name = getTrackName_l(); 2548 if (name < 0) break; 2549 mTracks[i]->mName = name; 2550 // limit track sample rate to 2 x new output sample rate 2551 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2552 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2553 } 2554 } 2555 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2556 } 2557 } 2558 2559 mNewParameters.removeAt(0); 2560 2561 mParamStatus = status; 2562 mParamCond.signal(); 2563 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2564 // already timed out waiting for the status and will never signal the condition. 2565 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2566 } 2567 return reconfig; 2568} 2569 2570status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2571{ 2572 const size_t SIZE = 256; 2573 char buffer[SIZE]; 2574 String8 result; 2575 2576 PlaybackThread::dumpInternals(fd, args); 2577 2578 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2579 result.append(buffer); 2580 write(fd, result.string(), result.size()); 2581 return NO_ERROR; 2582} 2583 2584uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2585{ 2586 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2587} 2588 2589uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2590{ 2591 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2592} 2593 2594// ---------------------------------------------------------------------------- 2595AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2596 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2597 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2598 // mLeftVolFloat, mRightVolFloat 2599 // mLeftVolShort, mRightVolShort 2600{ 2601} 2602 2603AudioFlinger::DirectOutputThread::~DirectOutputThread() 2604{ 2605} 2606 2607void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2608{ 2609 // Do not apply volume on compressed audio 2610 if (!audio_is_linear_pcm(mFormat)) { 2611 return; 2612 } 2613 2614 // convert to signed 16 bit before volume calculation 2615 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2616 size_t count = mFrameCount * mChannelCount; 2617 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2618 int16_t *dst = mMixBuffer + count-1; 2619 while(count--) { 2620 *dst-- = (int16_t)(*src--^0x80) << 8; 2621 } 2622 } 2623 2624 size_t frameCount = mFrameCount; 2625 int16_t *out = mMixBuffer; 2626 if (ramp) { 2627 if (mChannelCount == 1) { 2628 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2629 int32_t vlInc = d / (int32_t)frameCount; 2630 int32_t vl = ((int32_t)mLeftVolShort << 16); 2631 do { 2632 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2633 out++; 2634 vl += vlInc; 2635 } while (--frameCount); 2636 2637 } else { 2638 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2639 int32_t vlInc = d / (int32_t)frameCount; 2640 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2641 int32_t vrInc = d / (int32_t)frameCount; 2642 int32_t vl = ((int32_t)mLeftVolShort << 16); 2643 int32_t vr = ((int32_t)mRightVolShort << 16); 2644 do { 2645 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2646 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2647 out += 2; 2648 vl += vlInc; 2649 vr += vrInc; 2650 } while (--frameCount); 2651 } 2652 } else { 2653 if (mChannelCount == 1) { 2654 do { 2655 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2656 out++; 2657 } while (--frameCount); 2658 } else { 2659 do { 2660 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2661 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2662 out += 2; 2663 } while (--frameCount); 2664 } 2665 } 2666 2667 // convert back to unsigned 8 bit after volume calculation 2668 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2669 size_t count = mFrameCount * mChannelCount; 2670 int16_t *src = mMixBuffer; 2671 uint8_t *dst = (uint8_t *)mMixBuffer; 2672 while(count--) { 2673 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2674 } 2675 } 2676 2677 mLeftVolShort = leftVol; 2678 mRightVolShort = rightVol; 2679} 2680 2681bool AudioFlinger::DirectOutputThread::threadLoop() 2682{ 2683 mixer_state mixerStatus = MIXER_IDLE; 2684 sp<Track> trackToRemove; 2685 sp<Track> activeTrack; 2686 nsecs_t standbyTime = systemTime(); 2687 size_t mixBufferSize = mFrameCount*mFrameSize; 2688 uint32_t activeSleepTime = activeSleepTimeUs(); 2689 uint32_t idleSleepTime = idleSleepTimeUs(); 2690 uint32_t sleepTime = idleSleepTime; 2691 // use shorter standby delay as on normal output to release 2692 // hardware resources as soon as possible 2693 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2694 2695 acquireWakeLock(); 2696 2697 while (!exitPending()) 2698 { 2699 bool rampVolume; 2700 uint16_t leftVol; 2701 uint16_t rightVol; 2702 Vector< sp<EffectChain> > effectChains; 2703 2704 processConfigEvents(); 2705 2706 mixerStatus = MIXER_IDLE; 2707 2708 { // scope for the mLock 2709 2710 Mutex::Autolock _l(mLock); 2711 2712 if (checkForNewParameters_l()) { 2713 mixBufferSize = mFrameCount*mFrameSize; 2714 activeSleepTime = activeSleepTimeUs(); 2715 idleSleepTime = idleSleepTimeUs(); 2716 standbyDelay = microseconds(activeSleepTime*2); 2717 } 2718 2719 // put audio hardware into standby after short delay 2720 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2721 mSuspended)) { 2722 // wait until we have something to do... 2723 if (!mStandby) { 2724 ALOGV("Audio hardware entering standby, mixer %p", this); 2725 mOutput->stream->common.standby(&mOutput->stream->common); 2726 mStandby = true; 2727 mBytesWritten = 0; 2728 } 2729 2730 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2731 // we're about to wait, flush the binder command buffer 2732 IPCThreadState::self()->flushCommands(); 2733 2734 if (exitPending()) break; 2735 2736 releaseWakeLock_l(); 2737 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2738 mWaitWorkCV.wait(mLock); 2739 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2740 acquireWakeLock_l(); 2741 2742 if (!mMasterMute) { 2743 char value[PROPERTY_VALUE_MAX]; 2744 property_get("ro.audio.silent", value, "0"); 2745 if (atoi(value)) { 2746 ALOGD("Silence is golden"); 2747 setMasterMute_l(true); 2748 } 2749 } 2750 2751 standbyTime = systemTime() + standbyDelay; 2752 sleepTime = idleSleepTime; 2753 continue; 2754 } 2755 } 2756 2757 effectChains = mEffectChains; 2758 2759 // find out which tracks need to be processed 2760 if (mActiveTracks.size() != 0) { 2761 sp<Track> t = mActiveTracks[0].promote(); 2762 if (t == 0) continue; 2763 2764 Track* const track = t.get(); 2765 audio_track_cblk_t* cblk = track->cblk(); 2766 2767 // The first time a track is added we wait 2768 // for all its buffers to be filled before processing it 2769 if (cblk->framesReady() && track->isReady() && 2770 !track->isPaused() && !track->isTerminated()) 2771 { 2772 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2773 2774 if (track->mFillingUpStatus == Track::FS_FILLED) { 2775 track->mFillingUpStatus = Track::FS_ACTIVE; 2776 mLeftVolFloat = mRightVolFloat = 0; 2777 mLeftVolShort = mRightVolShort = 0; 2778 if (track->mState == TrackBase::RESUMING) { 2779 track->mState = TrackBase::ACTIVE; 2780 rampVolume = true; 2781 } 2782 } else if (cblk->server != 0) { 2783 // If the track is stopped before the first frame was mixed, 2784 // do not apply ramp 2785 rampVolume = true; 2786 } 2787 // compute volume for this track 2788 float left, right; 2789 if (track->isMuted() || mMasterMute || track->isPausing() || 2790 mStreamTypes[track->streamType()].mute) { 2791 left = right = 0; 2792 if (track->isPausing()) { 2793 track->setPaused(); 2794 } 2795 } else { 2796 float typeVolume = mStreamTypes[track->streamType()].volume; 2797 float v = mMasterVolume * typeVolume; 2798 uint32_t vlr = cblk->getVolumeLR(); 2799 float v_clamped = v * (vlr & 0xFFFF); 2800 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2801 left = v_clamped/MAX_GAIN; 2802 v_clamped = v * (vlr >> 16); 2803 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2804 right = v_clamped/MAX_GAIN; 2805 } 2806 2807 if (left != mLeftVolFloat || right != mRightVolFloat) { 2808 mLeftVolFloat = left; 2809 mRightVolFloat = right; 2810 2811 // If audio HAL implements volume control, 2812 // force software volume to nominal value 2813 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2814 left = 1.0f; 2815 right = 1.0f; 2816 } 2817 2818 // Convert volumes from float to 8.24 2819 uint32_t vl = (uint32_t)(left * (1 << 24)); 2820 uint32_t vr = (uint32_t)(right * (1 << 24)); 2821 2822 // Delegate volume control to effect in track effect chain if needed 2823 // only one effect chain can be present on DirectOutputThread, so if 2824 // there is one, the track is connected to it 2825 if (!effectChains.isEmpty()) { 2826 // Do not ramp volume if volume is controlled by effect 2827 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2828 rampVolume = false; 2829 } 2830 } 2831 2832 // Convert volumes from 8.24 to 4.12 format 2833 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2834 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2835 leftVol = (uint16_t)v_clamped; 2836 v_clamped = (vr + (1 << 11)) >> 12; 2837 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2838 rightVol = (uint16_t)v_clamped; 2839 } else { 2840 leftVol = mLeftVolShort; 2841 rightVol = mRightVolShort; 2842 rampVolume = false; 2843 } 2844 2845 // reset retry count 2846 track->mRetryCount = kMaxTrackRetriesDirect; 2847 activeTrack = t; 2848 mixerStatus = MIXER_TRACKS_READY; 2849 } else { 2850 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2851 if (track->isStopped()) { 2852 track->reset(); 2853 } 2854 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2855 // We have consumed all the buffers of this track. 2856 // Remove it from the list of active tracks. 2857 trackToRemove = track; 2858 } else { 2859 // No buffers for this track. Give it a few chances to 2860 // fill a buffer, then remove it from active list. 2861 if (--(track->mRetryCount) <= 0) { 2862 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2863 trackToRemove = track; 2864 } else { 2865 mixerStatus = MIXER_TRACKS_ENABLED; 2866 } 2867 } 2868 } 2869 } 2870 2871 // remove all the tracks that need to be... 2872 if (CC_UNLIKELY(trackToRemove != 0)) { 2873 mActiveTracks.remove(trackToRemove); 2874 if (!effectChains.isEmpty()) { 2875 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2876 trackToRemove->sessionId()); 2877 effectChains[0]->decActiveTrackCnt(); 2878 } 2879 if (trackToRemove->isTerminated()) { 2880 removeTrack_l(trackToRemove); 2881 } 2882 } 2883 2884 lockEffectChains_l(effectChains); 2885 } 2886 2887 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2888 AudioBufferProvider::Buffer buffer; 2889 size_t frameCount = mFrameCount; 2890 int8_t *curBuf = (int8_t *)mMixBuffer; 2891 // output audio to hardware 2892 while (frameCount) { 2893 buffer.frameCount = frameCount; 2894 activeTrack->getNextBuffer(&buffer); 2895 if (CC_UNLIKELY(buffer.raw == NULL)) { 2896 memset(curBuf, 0, frameCount * mFrameSize); 2897 break; 2898 } 2899 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2900 frameCount -= buffer.frameCount; 2901 curBuf += buffer.frameCount * mFrameSize; 2902 activeTrack->releaseBuffer(&buffer); 2903 } 2904 sleepTime = 0; 2905 standbyTime = systemTime() + standbyDelay; 2906 } else { 2907 if (sleepTime == 0) { 2908 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2909 sleepTime = activeSleepTime; 2910 } else { 2911 sleepTime = idleSleepTime; 2912 } 2913 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2914 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2915 sleepTime = 0; 2916 } 2917 } 2918 2919 if (mSuspended) { 2920 sleepTime = suspendSleepTimeUs(); 2921 } 2922 // sleepTime == 0 means we must write to audio hardware 2923 if (sleepTime == 0) { 2924 if (mixerStatus == MIXER_TRACKS_READY) { 2925 applyVolume(leftVol, rightVol, rampVolume); 2926 } 2927 for (size_t i = 0; i < effectChains.size(); i ++) { 2928 effectChains[i]->process_l(); 2929 } 2930 unlockEffectChains(effectChains); 2931 2932 mLastWriteTime = systemTime(); 2933 mInWrite = true; 2934 mBytesWritten += mixBufferSize; 2935 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2936 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2937 mNumWrites++; 2938 mInWrite = false; 2939 mStandby = false; 2940 } else { 2941 unlockEffectChains(effectChains); 2942 usleep(sleepTime); 2943 } 2944 2945 // finally let go of removed track, without the lock held 2946 // since we can't guarantee the destructors won't acquire that 2947 // same lock. 2948 trackToRemove.clear(); 2949 activeTrack.clear(); 2950 2951 // Effect chains will be actually deleted here if they were removed from 2952 // mEffectChains list during mixing or effects processing 2953 effectChains.clear(); 2954 } 2955 2956 if (!mStandby) { 2957 mOutput->stream->common.standby(&mOutput->stream->common); 2958 } 2959 2960 releaseWakeLock(); 2961 2962 ALOGV("DirectOutputThread %p exiting", this); 2963 return false; 2964} 2965 2966// getTrackName_l() must be called with ThreadBase::mLock held 2967int AudioFlinger::DirectOutputThread::getTrackName_l() 2968{ 2969 return 0; 2970} 2971 2972// deleteTrackName_l() must be called with ThreadBase::mLock held 2973void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2974{ 2975} 2976 2977// checkForNewParameters_l() must be called with ThreadBase::mLock held 2978bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2979{ 2980 bool reconfig = false; 2981 2982 while (!mNewParameters.isEmpty()) { 2983 status_t status = NO_ERROR; 2984 String8 keyValuePair = mNewParameters[0]; 2985 AudioParameter param = AudioParameter(keyValuePair); 2986 int value; 2987 2988 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2989 // do not accept frame count changes if tracks are open as the track buffer 2990 // size depends on frame count and correct behavior would not be garantied 2991 // if frame count is changed after track creation 2992 if (!mTracks.isEmpty()) { 2993 status = INVALID_OPERATION; 2994 } else { 2995 reconfig = true; 2996 } 2997 } 2998 if (status == NO_ERROR) { 2999 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3000 keyValuePair.string()); 3001 if (!mStandby && status == INVALID_OPERATION) { 3002 mOutput->stream->common.standby(&mOutput->stream->common); 3003 mStandby = true; 3004 mBytesWritten = 0; 3005 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3006 keyValuePair.string()); 3007 } 3008 if (status == NO_ERROR && reconfig) { 3009 readOutputParameters(); 3010 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3011 } 3012 } 3013 3014 mNewParameters.removeAt(0); 3015 3016 mParamStatus = status; 3017 mParamCond.signal(); 3018 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3019 // already timed out waiting for the status and will never signal the condition. 3020 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3021 } 3022 return reconfig; 3023} 3024 3025uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3026{ 3027 uint32_t time; 3028 if (audio_is_linear_pcm(mFormat)) { 3029 time = PlaybackThread::activeSleepTimeUs(); 3030 } else { 3031 time = 10000; 3032 } 3033 return time; 3034} 3035 3036uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3037{ 3038 uint32_t time; 3039 if (audio_is_linear_pcm(mFormat)) { 3040 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3041 } else { 3042 time = 10000; 3043 } 3044 return time; 3045} 3046 3047uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3048{ 3049 uint32_t time; 3050 if (audio_is_linear_pcm(mFormat)) { 3051 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3052 } else { 3053 time = 10000; 3054 } 3055 return time; 3056} 3057 3058 3059// ---------------------------------------------------------------------------- 3060 3061AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3062 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3063 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3064 mWaitTimeMs(UINT_MAX) 3065{ 3066 addOutputTrack(mainThread); 3067} 3068 3069AudioFlinger::DuplicatingThread::~DuplicatingThread() 3070{ 3071 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3072 mOutputTracks[i]->destroy(); 3073 } 3074} 3075 3076bool AudioFlinger::DuplicatingThread::threadLoop() 3077{ 3078 Vector< sp<Track> > tracksToRemove; 3079 mixer_state mixerStatus = MIXER_IDLE; 3080 nsecs_t standbyTime = systemTime(); 3081 size_t mixBufferSize = mFrameCount*mFrameSize; 3082 SortedVector< sp<OutputTrack> > outputTracks; 3083 uint32_t writeFrames = 0; 3084 uint32_t activeSleepTime = activeSleepTimeUs(); 3085 uint32_t idleSleepTime = idleSleepTimeUs(); 3086 uint32_t sleepTime = idleSleepTime; 3087 Vector< sp<EffectChain> > effectChains; 3088 3089 acquireWakeLock(); 3090 3091 while (!exitPending()) 3092 { 3093 processConfigEvents(); 3094 3095 mixerStatus = MIXER_IDLE; 3096 { // scope for the mLock 3097 3098 Mutex::Autolock _l(mLock); 3099 3100 if (checkForNewParameters_l()) { 3101 mixBufferSize = mFrameCount*mFrameSize; 3102 updateWaitTime(); 3103 activeSleepTime = activeSleepTimeUs(); 3104 idleSleepTime = idleSleepTimeUs(); 3105 } 3106 3107 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3108 3109 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3110 outputTracks.add(mOutputTracks[i]); 3111 } 3112 3113 // put audio hardware into standby after short delay 3114 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3115 mSuspended)) { 3116 if (!mStandby) { 3117 for (size_t i = 0; i < outputTracks.size(); i++) { 3118 outputTracks[i]->stop(); 3119 } 3120 mStandby = true; 3121 mBytesWritten = 0; 3122 } 3123 3124 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3125 // we're about to wait, flush the binder command buffer 3126 IPCThreadState::self()->flushCommands(); 3127 outputTracks.clear(); 3128 3129 if (exitPending()) break; 3130 3131 releaseWakeLock_l(); 3132 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3133 mWaitWorkCV.wait(mLock); 3134 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3135 acquireWakeLock_l(); 3136 3137 mPrevMixerStatus = MIXER_IDLE; 3138 if (!mMasterMute) { 3139 char value[PROPERTY_VALUE_MAX]; 3140 property_get("ro.audio.silent", value, "0"); 3141 if (atoi(value)) { 3142 ALOGD("Silence is golden"); 3143 setMasterMute_l(true); 3144 } 3145 } 3146 3147 standbyTime = systemTime() + mStandbyTimeInNsecs; 3148 sleepTime = idleSleepTime; 3149 continue; 3150 } 3151 } 3152 3153 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3154 3155 // prevent any changes in effect chain list and in each effect chain 3156 // during mixing and effect process as the audio buffers could be deleted 3157 // or modified if an effect is created or deleted 3158 lockEffectChains_l(effectChains); 3159 } 3160 3161 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3162 // mix buffers... 3163 if (outputsReady(outputTracks)) { 3164 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3165 } else { 3166 memset(mMixBuffer, 0, mixBufferSize); 3167 } 3168 sleepTime = 0; 3169 writeFrames = mFrameCount; 3170 } else { 3171 if (sleepTime == 0) { 3172 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3173 sleepTime = activeSleepTime; 3174 } else { 3175 sleepTime = idleSleepTime; 3176 } 3177 } else if (mBytesWritten != 0) { 3178 // flush remaining overflow buffers in output tracks 3179 for (size_t i = 0; i < outputTracks.size(); i++) { 3180 if (outputTracks[i]->isActive()) { 3181 sleepTime = 0; 3182 writeFrames = 0; 3183 memset(mMixBuffer, 0, mixBufferSize); 3184 break; 3185 } 3186 } 3187 } 3188 } 3189 3190 if (mSuspended) { 3191 sleepTime = suspendSleepTimeUs(); 3192 } 3193 // sleepTime == 0 means we must write to audio hardware 3194 if (sleepTime == 0) { 3195 for (size_t i = 0; i < effectChains.size(); i ++) { 3196 effectChains[i]->process_l(); 3197 } 3198 // enable changes in effect chain 3199 unlockEffectChains(effectChains); 3200 3201 standbyTime = systemTime() + mStandbyTimeInNsecs; 3202 for (size_t i = 0; i < outputTracks.size(); i++) { 3203 outputTracks[i]->write(mMixBuffer, writeFrames); 3204 } 3205 mStandby = false; 3206 mBytesWritten += mixBufferSize; 3207 } else { 3208 // enable changes in effect chain 3209 unlockEffectChains(effectChains); 3210 usleep(sleepTime); 3211 } 3212 3213 // finally let go of all our tracks, without the lock held 3214 // since we can't guarantee the destructors won't acquire that 3215 // same lock. 3216 tracksToRemove.clear(); 3217 outputTracks.clear(); 3218 3219 // Effect chains will be actually deleted here if they were removed from 3220 // mEffectChains list during mixing or effects processing 3221 effectChains.clear(); 3222 } 3223 3224 releaseWakeLock(); 3225 3226 return false; 3227} 3228 3229void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3230{ 3231 // FIXME explain this formula 3232 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3233 OutputTrack *outputTrack = new OutputTrack(thread, 3234 this, 3235 mSampleRate, 3236 mFormat, 3237 mChannelMask, 3238 frameCount); 3239 if (outputTrack->cblk() != NULL) { 3240 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3241 mOutputTracks.add(outputTrack); 3242 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3243 updateWaitTime(); 3244 } 3245} 3246 3247void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3248{ 3249 Mutex::Autolock _l(mLock); 3250 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3251 if (mOutputTracks[i]->thread() == thread) { 3252 mOutputTracks[i]->destroy(); 3253 mOutputTracks.removeAt(i); 3254 updateWaitTime(); 3255 return; 3256 } 3257 } 3258 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3259} 3260 3261void AudioFlinger::DuplicatingThread::updateWaitTime() 3262{ 3263 mWaitTimeMs = UINT_MAX; 3264 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3265 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3266 if (strong != 0) { 3267 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3268 if (waitTimeMs < mWaitTimeMs) { 3269 mWaitTimeMs = waitTimeMs; 3270 } 3271 } 3272 } 3273} 3274 3275 3276bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3277{ 3278 for (size_t i = 0; i < outputTracks.size(); i++) { 3279 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3280 if (thread == 0) { 3281 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3282 return false; 3283 } 3284 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3285 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3286 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3287 return false; 3288 } 3289 } 3290 return true; 3291} 3292 3293uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3294{ 3295 return (mWaitTimeMs * 1000) / 2; 3296} 3297 3298// ---------------------------------------------------------------------------- 3299 3300// TrackBase constructor must be called with AudioFlinger::mLock held 3301AudioFlinger::ThreadBase::TrackBase::TrackBase( 3302 ThreadBase *thread, 3303 const sp<Client>& client, 3304 uint32_t sampleRate, 3305 audio_format_t format, 3306 uint32_t channelMask, 3307 int frameCount, 3308 uint32_t flags, 3309 const sp<IMemory>& sharedBuffer, 3310 int sessionId) 3311 : RefBase(), 3312 mThread(thread), 3313 mClient(client), 3314 mCblk(NULL), 3315 // mBuffer 3316 // mBufferEnd 3317 mFrameCount(0), 3318 mState(IDLE), 3319 mFormat(format), 3320 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3321 mSessionId(sessionId) 3322 // mChannelCount 3323 // mChannelMask 3324{ 3325 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3326 3327 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3328 size_t size = sizeof(audio_track_cblk_t); 3329 uint8_t channelCount = popcount(channelMask); 3330 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3331 if (sharedBuffer == 0) { 3332 size += bufferSize; 3333 } 3334 3335 if (client != NULL) { 3336 mCblkMemory = client->heap()->allocate(size); 3337 if (mCblkMemory != 0) { 3338 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3339 if (mCblk != NULL) { // construct the shared structure in-place. 3340 new(mCblk) audio_track_cblk_t(); 3341 // clear all buffers 3342 mCblk->frameCount = frameCount; 3343 mCblk->sampleRate = sampleRate; 3344 mChannelCount = channelCount; 3345 mChannelMask = channelMask; 3346 if (sharedBuffer == 0) { 3347 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3348 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3349 // Force underrun condition to avoid false underrun callback until first data is 3350 // written to buffer (other flags are cleared) 3351 mCblk->flags = CBLK_UNDERRUN_ON; 3352 } else { 3353 mBuffer = sharedBuffer->pointer(); 3354 } 3355 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3356 } 3357 } else { 3358 ALOGE("not enough memory for AudioTrack size=%u", size); 3359 client->heap()->dump("AudioTrack"); 3360 return; 3361 } 3362 } else { 3363 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3364 // construct the shared structure in-place. 3365 new(mCblk) audio_track_cblk_t(); 3366 // clear all buffers 3367 mCblk->frameCount = frameCount; 3368 mCblk->sampleRate = sampleRate; 3369 mChannelCount = channelCount; 3370 mChannelMask = channelMask; 3371 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3372 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3373 // Force underrun condition to avoid false underrun callback until first data is 3374 // written to buffer (other flags are cleared) 3375 mCblk->flags = CBLK_UNDERRUN_ON; 3376 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3377 } 3378} 3379 3380AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3381{ 3382 if (mCblk != NULL) { 3383 if (mClient == 0) { 3384 delete mCblk; 3385 } else { 3386 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3387 } 3388 } 3389 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3390 if (mClient != 0) { 3391 // Client destructor must run with AudioFlinger mutex locked 3392 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3393 // If the client's reference count drops to zero, the associated destructor 3394 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3395 // relying on the automatic clear() at end of scope. 3396 mClient.clear(); 3397 } 3398} 3399 3400// AudioBufferProvider interface 3401// getNextBuffer() = 0; 3402// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3403void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3404{ 3405 buffer->raw = NULL; 3406 mFrameCount = buffer->frameCount; 3407 (void) step(); // ignore return value of step() 3408 buffer->frameCount = 0; 3409} 3410 3411bool AudioFlinger::ThreadBase::TrackBase::step() { 3412 bool result; 3413 audio_track_cblk_t* cblk = this->cblk(); 3414 3415 result = cblk->stepServer(mFrameCount); 3416 if (!result) { 3417 ALOGV("stepServer failed acquiring cblk mutex"); 3418 mFlags |= STEPSERVER_FAILED; 3419 } 3420 return result; 3421} 3422 3423void AudioFlinger::ThreadBase::TrackBase::reset() { 3424 audio_track_cblk_t* cblk = this->cblk(); 3425 3426 cblk->user = 0; 3427 cblk->server = 0; 3428 cblk->userBase = 0; 3429 cblk->serverBase = 0; 3430 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3431 ALOGV("TrackBase::reset"); 3432} 3433 3434int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3435 return (int)mCblk->sampleRate; 3436} 3437 3438void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3439 audio_track_cblk_t* cblk = this->cblk(); 3440 size_t frameSize = cblk->frameSize; 3441 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3442 int8_t *bufferEnd = bufferStart + frames * frameSize; 3443 3444 // Check validity of returned pointer in case the track control block would have been corrupted. 3445 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3446 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3447 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3448 server %d, serverBase %d, user %d, userBase %d", 3449 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3450 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3451 return NULL; 3452 } 3453 3454 return bufferStart; 3455} 3456 3457// ---------------------------------------------------------------------------- 3458 3459// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3460AudioFlinger::PlaybackThread::Track::Track( 3461 PlaybackThread *thread, 3462 const sp<Client>& client, 3463 audio_stream_type_t streamType, 3464 uint32_t sampleRate, 3465 audio_format_t format, 3466 uint32_t channelMask, 3467 int frameCount, 3468 const sp<IMemory>& sharedBuffer, 3469 int sessionId) 3470 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3471 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3472 mAuxEffectId(0), mHasVolumeController(false) 3473{ 3474 if (mCblk != NULL) { 3475 if (thread != NULL) { 3476 mName = thread->getTrackName_l(); 3477 mMainBuffer = thread->mixBuffer(); 3478 } 3479 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3480 if (mName < 0) { 3481 ALOGE("no more track names available"); 3482 } 3483 mStreamType = streamType; 3484 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3485 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3486 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3487 } 3488} 3489 3490AudioFlinger::PlaybackThread::Track::~Track() 3491{ 3492 ALOGV("PlaybackThread::Track destructor"); 3493 sp<ThreadBase> thread = mThread.promote(); 3494 if (thread != 0) { 3495 Mutex::Autolock _l(thread->mLock); 3496 mState = TERMINATED; 3497 } 3498} 3499 3500void AudioFlinger::PlaybackThread::Track::destroy() 3501{ 3502 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3503 // by removing it from mTracks vector, so there is a risk that this Tracks's 3504 // destructor is called. As the destructor needs to lock mLock, 3505 // we must acquire a strong reference on this Track before locking mLock 3506 // here so that the destructor is called only when exiting this function. 3507 // On the other hand, as long as Track::destroy() is only called by 3508 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3509 // this Track with its member mTrack. 3510 sp<Track> keep(this); 3511 { // scope for mLock 3512 sp<ThreadBase> thread = mThread.promote(); 3513 if (thread != 0) { 3514 if (!isOutputTrack()) { 3515 if (mState == ACTIVE || mState == RESUMING) { 3516 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3517 3518 // to track the speaker usage 3519 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3520 } 3521 AudioSystem::releaseOutput(thread->id()); 3522 } 3523 Mutex::Autolock _l(thread->mLock); 3524 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3525 playbackThread->destroyTrack_l(this); 3526 } 3527 } 3528} 3529 3530void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3531{ 3532 uint32_t vlr = mCblk->getVolumeLR(); 3533 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", 3534 mName - AudioMixer::TRACK0, 3535 (mClient == 0) ? getpid_cached : mClient->pid(), 3536 mStreamType, 3537 mFormat, 3538 mChannelMask, 3539 mSessionId, 3540 mFrameCount, 3541 mState, 3542 mMute, 3543 mFillingUpStatus, 3544 mCblk->sampleRate, 3545 vlr & 0xFFFF, 3546 vlr >> 16, 3547 mCblk->server, 3548 mCblk->user, 3549 (int)mMainBuffer, 3550 (int)mAuxBuffer); 3551} 3552 3553// AudioBufferProvider interface 3554status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3555 AudioBufferProvider::Buffer* buffer, int64_t pts) 3556{ 3557 audio_track_cblk_t* cblk = this->cblk(); 3558 uint32_t framesReady; 3559 uint32_t framesReq = buffer->frameCount; 3560 3561 // Check if last stepServer failed, try to step now 3562 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3563 if (!step()) goto getNextBuffer_exit; 3564 ALOGV("stepServer recovered"); 3565 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3566 } 3567 3568 framesReady = cblk->framesReady(); 3569 3570 if (CC_LIKELY(framesReady)) { 3571 uint32_t s = cblk->server; 3572 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3573 3574 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3575 if (framesReq > framesReady) { 3576 framesReq = framesReady; 3577 } 3578 if (s + framesReq > bufferEnd) { 3579 framesReq = bufferEnd - s; 3580 } 3581 3582 buffer->raw = getBuffer(s, framesReq); 3583 if (buffer->raw == NULL) goto getNextBuffer_exit; 3584 3585 buffer->frameCount = framesReq; 3586 return NO_ERROR; 3587 } 3588 3589getNextBuffer_exit: 3590 buffer->raw = NULL; 3591 buffer->frameCount = 0; 3592 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3593 return NOT_ENOUGH_DATA; 3594} 3595 3596uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3597 return mCblk->framesReady(); 3598} 3599 3600bool AudioFlinger::PlaybackThread::Track::isReady() const { 3601 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3602 3603 if (framesReady() >= mCblk->frameCount || 3604 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3605 mFillingUpStatus = FS_FILLED; 3606 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3607 return true; 3608 } 3609 return false; 3610} 3611 3612status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3613{ 3614 status_t status = NO_ERROR; 3615 ALOGV("start(%d), calling pid %d session %d tid %d", 3616 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3617 sp<ThreadBase> thread = mThread.promote(); 3618 if (thread != 0) { 3619 Mutex::Autolock _l(thread->mLock); 3620 track_state state = mState; 3621 // here the track could be either new, or restarted 3622 // in both cases "unstop" the track 3623 if (mState == PAUSED) { 3624 mState = TrackBase::RESUMING; 3625 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3626 } else { 3627 mState = TrackBase::ACTIVE; 3628 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3629 } 3630 3631 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3632 thread->mLock.unlock(); 3633 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3634 thread->mLock.lock(); 3635 3636 // to track the speaker usage 3637 if (status == NO_ERROR) { 3638 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3639 } 3640 } 3641 if (status == NO_ERROR) { 3642 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3643 playbackThread->addTrack_l(this); 3644 } else { 3645 mState = state; 3646 } 3647 } else { 3648 status = BAD_VALUE; 3649 } 3650 return status; 3651} 3652 3653void AudioFlinger::PlaybackThread::Track::stop() 3654{ 3655 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3656 sp<ThreadBase> thread = mThread.promote(); 3657 if (thread != 0) { 3658 Mutex::Autolock _l(thread->mLock); 3659 track_state state = mState; 3660 if (mState > STOPPED) { 3661 mState = STOPPED; 3662 // If the track is not active (PAUSED and buffers full), flush buffers 3663 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3664 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3665 reset(); 3666 } 3667 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3668 } 3669 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3670 thread->mLock.unlock(); 3671 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3672 thread->mLock.lock(); 3673 3674 // to track the speaker usage 3675 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3676 } 3677 } 3678} 3679 3680void AudioFlinger::PlaybackThread::Track::pause() 3681{ 3682 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3683 sp<ThreadBase> thread = mThread.promote(); 3684 if (thread != 0) { 3685 Mutex::Autolock _l(thread->mLock); 3686 if (mState == ACTIVE || mState == RESUMING) { 3687 mState = PAUSING; 3688 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3689 if (!isOutputTrack()) { 3690 thread->mLock.unlock(); 3691 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3692 thread->mLock.lock(); 3693 3694 // to track the speaker usage 3695 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3696 } 3697 } 3698 } 3699} 3700 3701void AudioFlinger::PlaybackThread::Track::flush() 3702{ 3703 ALOGV("flush(%d)", mName); 3704 sp<ThreadBase> thread = mThread.promote(); 3705 if (thread != 0) { 3706 Mutex::Autolock _l(thread->mLock); 3707 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3708 return; 3709 } 3710 // No point remaining in PAUSED state after a flush => go to 3711 // STOPPED state 3712 mState = STOPPED; 3713 3714 // do not reset the track if it is still in the process of being stopped or paused. 3715 // this will be done by prepareTracks_l() when the track is stopped. 3716 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3717 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3718 reset(); 3719 } 3720 } 3721} 3722 3723void AudioFlinger::PlaybackThread::Track::reset() 3724{ 3725 // Do not reset twice to avoid discarding data written just after a flush and before 3726 // the audioflinger thread detects the track is stopped. 3727 if (!mResetDone) { 3728 TrackBase::reset(); 3729 // Force underrun condition to avoid false underrun callback until first data is 3730 // written to buffer 3731 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3732 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3733 mFillingUpStatus = FS_FILLING; 3734 mResetDone = true; 3735 } 3736} 3737 3738void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3739{ 3740 mMute = muted; 3741} 3742 3743status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3744{ 3745 status_t status = DEAD_OBJECT; 3746 sp<ThreadBase> thread = mThread.promote(); 3747 if (thread != 0) { 3748 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3749 status = playbackThread->attachAuxEffect(this, EffectId); 3750 } 3751 return status; 3752} 3753 3754void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3755{ 3756 mAuxEffectId = EffectId; 3757 mAuxBuffer = buffer; 3758} 3759 3760// timed audio tracks 3761 3762sp<AudioFlinger::PlaybackThread::TimedTrack> 3763AudioFlinger::PlaybackThread::TimedTrack::create( 3764 PlaybackThread *thread, 3765 const sp<Client>& client, 3766 audio_stream_type_t streamType, 3767 uint32_t sampleRate, 3768 audio_format_t format, 3769 uint32_t channelMask, 3770 int frameCount, 3771 const sp<IMemory>& sharedBuffer, 3772 int sessionId) { 3773 if (!client->reserveTimedTrack()) 3774 return NULL; 3775 3776 sp<TimedTrack> track = new TimedTrack( 3777 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3778 sharedBuffer, sessionId); 3779 3780 if (track == NULL) { 3781 client->releaseTimedTrack(); 3782 return NULL; 3783 } 3784 3785 return track; 3786} 3787 3788AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3789 PlaybackThread *thread, 3790 const sp<Client>& client, 3791 audio_stream_type_t streamType, 3792 uint32_t sampleRate, 3793 audio_format_t format, 3794 uint32_t channelMask, 3795 int frameCount, 3796 const sp<IMemory>& sharedBuffer, 3797 int sessionId) 3798 : Track(thread, client, streamType, sampleRate, format, channelMask, 3799 frameCount, sharedBuffer, sessionId), 3800 mTimedSilenceBuffer(NULL), 3801 mTimedSilenceBufferSize(0), 3802 mTimedAudioOutputOnTime(false), 3803 mMediaTimeTransformValid(false) 3804{ 3805 LocalClock lc; 3806 mLocalTimeFreq = lc.getLocalFreq(); 3807 3808 mLocalTimeToSampleTransform.a_zero = 0; 3809 mLocalTimeToSampleTransform.b_zero = 0; 3810 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3811 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3812 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3813 &mLocalTimeToSampleTransform.a_to_b_denom); 3814} 3815 3816AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3817 mClient->releaseTimedTrack(); 3818 delete [] mTimedSilenceBuffer; 3819} 3820 3821status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3822 size_t size, sp<IMemory>* buffer) { 3823 3824 Mutex::Autolock _l(mTimedBufferQueueLock); 3825 3826 trimTimedBufferQueue_l(); 3827 3828 // lazily initialize the shared memory heap for timed buffers 3829 if (mTimedMemoryDealer == NULL) { 3830 const int kTimedBufferHeapSize = 512 << 10; 3831 3832 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3833 "AudioFlingerTimed"); 3834 if (mTimedMemoryDealer == NULL) 3835 return NO_MEMORY; 3836 } 3837 3838 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3839 if (newBuffer == NULL) { 3840 newBuffer = mTimedMemoryDealer->allocate(size); 3841 if (newBuffer == NULL) 3842 return NO_MEMORY; 3843 } 3844 3845 *buffer = newBuffer; 3846 return NO_ERROR; 3847} 3848 3849// caller must hold mTimedBufferQueueLock 3850void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3851 int64_t mediaTimeNow; 3852 { 3853 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3854 if (!mMediaTimeTransformValid) 3855 return; 3856 3857 int64_t targetTimeNow; 3858 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3859 ? mCCHelper.getCommonTime(&targetTimeNow) 3860 : mCCHelper.getLocalTime(&targetTimeNow); 3861 3862 if (OK != res) 3863 return; 3864 3865 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3866 &mediaTimeNow)) { 3867 return; 3868 } 3869 } 3870 3871 size_t trimIndex; 3872 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3873 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3874 break; 3875 } 3876 3877 if (trimIndex) { 3878 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3879 } 3880} 3881 3882status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3883 const sp<IMemory>& buffer, int64_t pts) { 3884 3885 { 3886 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3887 if (!mMediaTimeTransformValid) 3888 return INVALID_OPERATION; 3889 } 3890 3891 Mutex::Autolock _l(mTimedBufferQueueLock); 3892 3893 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3894 3895 return NO_ERROR; 3896} 3897 3898status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3899 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3900 3901 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3902 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3903 target); 3904 3905 if (!(target == TimedAudioTrack::LOCAL_TIME || 3906 target == TimedAudioTrack::COMMON_TIME)) { 3907 return BAD_VALUE; 3908 } 3909 3910 Mutex::Autolock lock(mMediaTimeTransformLock); 3911 mMediaTimeTransform = xform; 3912 mMediaTimeTransformTarget = target; 3913 mMediaTimeTransformValid = true; 3914 3915 return NO_ERROR; 3916} 3917 3918#define min(a, b) ((a) < (b) ? (a) : (b)) 3919 3920// implementation of getNextBuffer for tracks whose buffers have timestamps 3921status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3922 AudioBufferProvider::Buffer* buffer, int64_t pts) 3923{ 3924 if (pts == AudioBufferProvider::kInvalidPTS) { 3925 buffer->raw = 0; 3926 buffer->frameCount = 0; 3927 return INVALID_OPERATION; 3928 } 3929 3930 Mutex::Autolock _l(mTimedBufferQueueLock); 3931 3932 while (true) { 3933 3934 // if we have no timed buffers, then fail 3935 if (mTimedBufferQueue.isEmpty()) { 3936 buffer->raw = 0; 3937 buffer->frameCount = 0; 3938 return NOT_ENOUGH_DATA; 3939 } 3940 3941 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3942 3943 // calculate the PTS of the head of the timed buffer queue expressed in 3944 // local time 3945 int64_t headLocalPTS; 3946 { 3947 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3948 3949 assert(mMediaTimeTransformValid); 3950 3951 if (mMediaTimeTransform.a_to_b_denom == 0) { 3952 // the transform represents a pause, so yield silence 3953 timedYieldSilence(buffer->frameCount, buffer); 3954 return NO_ERROR; 3955 } 3956 3957 int64_t transformedPTS; 3958 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3959 &transformedPTS)) { 3960 // the transform failed. this shouldn't happen, but if it does 3961 // then just drop this buffer 3962 ALOGW("timedGetNextBuffer transform failed"); 3963 buffer->raw = 0; 3964 buffer->frameCount = 0; 3965 mTimedBufferQueue.removeAt(0); 3966 return NO_ERROR; 3967 } 3968 3969 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3970 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3971 &headLocalPTS)) { 3972 buffer->raw = 0; 3973 buffer->frameCount = 0; 3974 return INVALID_OPERATION; 3975 } 3976 } else { 3977 headLocalPTS = transformedPTS; 3978 } 3979 } 3980 3981 // adjust the head buffer's PTS to reflect the portion of the head buffer 3982 // that has already been consumed 3983 int64_t effectivePTS = headLocalPTS + 3984 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3985 3986 // Calculate the delta in samples between the head of the input buffer 3987 // queue and the start of the next output buffer that will be written. 3988 // If the transformation fails because of over or underflow, it means 3989 // that the sample's position in the output stream is so far out of 3990 // whack that it should just be dropped. 3991 int64_t sampleDelta; 3992 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3993 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3994 mTimedBufferQueue.removeAt(0); 3995 continue; 3996 } 3997 if (!mLocalTimeToSampleTransform.doForwardTransform( 3998 (effectivePTS - pts) << 32, &sampleDelta)) { 3999 ALOGV("*** too late during sample rate transform: dropped buffer"); 4000 mTimedBufferQueue.removeAt(0); 4001 continue; 4002 } 4003 4004 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4005 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4006 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4007 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4008 4009 // if the delta between the ideal placement for the next input sample and 4010 // the current output position is within this threshold, then we will 4011 // concatenate the next input samples to the previous output 4012 const int64_t kSampleContinuityThreshold = 4013 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4014 4015 // if this is the first buffer of audio that we're emitting from this track 4016 // then it should be almost exactly on time. 4017 const int64_t kSampleStartupThreshold = 1LL << 32; 4018 4019 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4020 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4021 // the next input is close enough to being on time, so concatenate it 4022 // with the last output 4023 timedYieldSamples(buffer); 4024 4025 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4026 return NO_ERROR; 4027 } else if (sampleDelta > 0) { 4028 // the gap between the current output position and the proper start of 4029 // the next input sample is too big, so fill it with silence 4030 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4031 4032 timedYieldSilence(framesUntilNextInput, buffer); 4033 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4034 return NO_ERROR; 4035 } else { 4036 // the next input sample is late 4037 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4038 size_t onTimeSamplePosition = 4039 head.position() + lateFrames * mCblk->frameSize; 4040 4041 if (onTimeSamplePosition > head.buffer()->size()) { 4042 // all the remaining samples in the head are too late, so 4043 // drop it and move on 4044 ALOGV("*** too late: dropped buffer"); 4045 mTimedBufferQueue.removeAt(0); 4046 continue; 4047 } else { 4048 // skip over the late samples 4049 head.setPosition(onTimeSamplePosition); 4050 4051 // yield the available samples 4052 timedYieldSamples(buffer); 4053 4054 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4055 return NO_ERROR; 4056 } 4057 } 4058 } 4059} 4060 4061// Yield samples from the timed buffer queue head up to the given output 4062// buffer's capacity. 4063// 4064// Caller must hold mTimedBufferQueueLock 4065void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4066 AudioBufferProvider::Buffer* buffer) { 4067 4068 const TimedBuffer& head = mTimedBufferQueue[0]; 4069 4070 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4071 head.position()); 4072 4073 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4074 mCblk->frameSize); 4075 size_t framesRequested = buffer->frameCount; 4076 buffer->frameCount = min(framesLeftInHead, framesRequested); 4077 4078 mTimedAudioOutputOnTime = true; 4079} 4080 4081// Yield samples of silence up to the given output buffer's capacity 4082// 4083// Caller must hold mTimedBufferQueueLock 4084void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4085 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4086 4087 // lazily allocate a buffer filled with silence 4088 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4089 delete [] mTimedSilenceBuffer; 4090 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4091 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4092 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4093 } 4094 4095 buffer->raw = mTimedSilenceBuffer; 4096 size_t framesRequested = buffer->frameCount; 4097 buffer->frameCount = min(numFrames, framesRequested); 4098 4099 mTimedAudioOutputOnTime = false; 4100} 4101 4102// AudioBufferProvider interface 4103void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4104 AudioBufferProvider::Buffer* buffer) { 4105 4106 Mutex::Autolock _l(mTimedBufferQueueLock); 4107 4108 // If the buffer which was just released is part of the buffer at the head 4109 // of the queue, be sure to update the amt of the buffer which has been 4110 // consumed. If the buffer being returned is not part of the head of the 4111 // queue, its either because the buffer is part of the silence buffer, or 4112 // because the head of the timed queue was trimmed after the mixer called 4113 // getNextBuffer but before the mixer called releaseBuffer. 4114 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4115 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4116 4117 void* start = head.buffer()->pointer(); 4118 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4119 4120 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4121 head.setPosition(head.position() + 4122 (buffer->frameCount * mCblk->frameSize)); 4123 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4124 mTimedBufferQueue.removeAt(0); 4125 } 4126 } 4127 } 4128 4129 buffer->raw = 0; 4130 buffer->frameCount = 0; 4131} 4132 4133uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4134 Mutex::Autolock _l(mTimedBufferQueueLock); 4135 4136 uint32_t frames = 0; 4137 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4138 const TimedBuffer& tb = mTimedBufferQueue[i]; 4139 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4140 } 4141 4142 return frames; 4143} 4144 4145AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4146 : mPTS(0), mPosition(0) {} 4147 4148AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4149 const sp<IMemory>& buffer, int64_t pts) 4150 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4151 4152// ---------------------------------------------------------------------------- 4153 4154// RecordTrack constructor must be called with AudioFlinger::mLock held 4155AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4156 RecordThread *thread, 4157 const sp<Client>& client, 4158 uint32_t sampleRate, 4159 audio_format_t format, 4160 uint32_t channelMask, 4161 int frameCount, 4162 uint32_t flags, 4163 int sessionId) 4164 : TrackBase(thread, client, sampleRate, format, 4165 channelMask, frameCount, flags, 0, sessionId), 4166 mOverflow(false) 4167{ 4168 if (mCblk != NULL) { 4169 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4170 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4171 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4172 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4173 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4174 } else { 4175 mCblk->frameSize = sizeof(int8_t); 4176 } 4177 } 4178} 4179 4180AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4181{ 4182 sp<ThreadBase> thread = mThread.promote(); 4183 if (thread != 0) { 4184 AudioSystem::releaseInput(thread->id()); 4185 } 4186} 4187 4188// AudioBufferProvider interface 4189status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4190{ 4191 audio_track_cblk_t* cblk = this->cblk(); 4192 uint32_t framesAvail; 4193 uint32_t framesReq = buffer->frameCount; 4194 4195 // Check if last stepServer failed, try to step now 4196 if (mFlags & TrackBase::STEPSERVER_FAILED) { 4197 if (!step()) goto getNextBuffer_exit; 4198 ALOGV("stepServer recovered"); 4199 mFlags &= ~TrackBase::STEPSERVER_FAILED; 4200 } 4201 4202 framesAvail = cblk->framesAvailable_l(); 4203 4204 if (CC_LIKELY(framesAvail)) { 4205 uint32_t s = cblk->server; 4206 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4207 4208 if (framesReq > framesAvail) { 4209 framesReq = framesAvail; 4210 } 4211 if (s + framesReq > bufferEnd) { 4212 framesReq = bufferEnd - s; 4213 } 4214 4215 buffer->raw = getBuffer(s, framesReq); 4216 if (buffer->raw == NULL) goto getNextBuffer_exit; 4217 4218 buffer->frameCount = framesReq; 4219 return NO_ERROR; 4220 } 4221 4222getNextBuffer_exit: 4223 buffer->raw = NULL; 4224 buffer->frameCount = 0; 4225 return NOT_ENOUGH_DATA; 4226} 4227 4228status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4229{ 4230 sp<ThreadBase> thread = mThread.promote(); 4231 if (thread != 0) { 4232 RecordThread *recordThread = (RecordThread *)thread.get(); 4233 return recordThread->start(this, tid); 4234 } else { 4235 return BAD_VALUE; 4236 } 4237} 4238 4239void AudioFlinger::RecordThread::RecordTrack::stop() 4240{ 4241 sp<ThreadBase> thread = mThread.promote(); 4242 if (thread != 0) { 4243 RecordThread *recordThread = (RecordThread *)thread.get(); 4244 recordThread->stop(this); 4245 TrackBase::reset(); 4246 // Force overerrun condition to avoid false overrun callback until first data is 4247 // read from buffer 4248 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4249 } 4250} 4251 4252void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4253{ 4254 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4255 (mClient == 0) ? getpid_cached : mClient->pid(), 4256 mFormat, 4257 mChannelMask, 4258 mSessionId, 4259 mFrameCount, 4260 mState, 4261 mCblk->sampleRate, 4262 mCblk->server, 4263 mCblk->user); 4264} 4265 4266 4267// ---------------------------------------------------------------------------- 4268 4269AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4270 PlaybackThread *playbackThread, 4271 DuplicatingThread *sourceThread, 4272 uint32_t sampleRate, 4273 audio_format_t format, 4274 uint32_t channelMask, 4275 int frameCount) 4276 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4277 mActive(false), mSourceThread(sourceThread) 4278{ 4279 4280 if (mCblk != NULL) { 4281 mCblk->flags |= CBLK_DIRECTION_OUT; 4282 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4283 mOutBuffer.frameCount = 0; 4284 playbackThread->mTracks.add(this); 4285 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4286 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4287 mCblk, mBuffer, mCblk->buffers, 4288 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4289 } else { 4290 ALOGW("Error creating output track on thread %p", playbackThread); 4291 } 4292} 4293 4294AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4295{ 4296 clearBufferQueue(); 4297} 4298 4299status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4300{ 4301 status_t status = Track::start(tid); 4302 if (status != NO_ERROR) { 4303 return status; 4304 } 4305 4306 mActive = true; 4307 mRetryCount = 127; 4308 return status; 4309} 4310 4311void AudioFlinger::PlaybackThread::OutputTrack::stop() 4312{ 4313 Track::stop(); 4314 clearBufferQueue(); 4315 mOutBuffer.frameCount = 0; 4316 mActive = false; 4317} 4318 4319bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4320{ 4321 Buffer *pInBuffer; 4322 Buffer inBuffer; 4323 uint32_t channelCount = mChannelCount; 4324 bool outputBufferFull = false; 4325 inBuffer.frameCount = frames; 4326 inBuffer.i16 = data; 4327 4328 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4329 4330 if (!mActive && frames != 0) { 4331 start(0); 4332 sp<ThreadBase> thread = mThread.promote(); 4333 if (thread != 0) { 4334 MixerThread *mixerThread = (MixerThread *)thread.get(); 4335 if (mCblk->frameCount > frames){ 4336 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4337 uint32_t startFrames = (mCblk->frameCount - frames); 4338 pInBuffer = new Buffer; 4339 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4340 pInBuffer->frameCount = startFrames; 4341 pInBuffer->i16 = pInBuffer->mBuffer; 4342 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4343 mBufferQueue.add(pInBuffer); 4344 } else { 4345 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4346 } 4347 } 4348 } 4349 } 4350 4351 while (waitTimeLeftMs) { 4352 // First write pending buffers, then new data 4353 if (mBufferQueue.size()) { 4354 pInBuffer = mBufferQueue.itemAt(0); 4355 } else { 4356 pInBuffer = &inBuffer; 4357 } 4358 4359 if (pInBuffer->frameCount == 0) { 4360 break; 4361 } 4362 4363 if (mOutBuffer.frameCount == 0) { 4364 mOutBuffer.frameCount = pInBuffer->frameCount; 4365 nsecs_t startTime = systemTime(); 4366 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4367 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4368 outputBufferFull = true; 4369 break; 4370 } 4371 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4372 if (waitTimeLeftMs >= waitTimeMs) { 4373 waitTimeLeftMs -= waitTimeMs; 4374 } else { 4375 waitTimeLeftMs = 0; 4376 } 4377 } 4378 4379 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4380 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4381 mCblk->stepUser(outFrames); 4382 pInBuffer->frameCount -= outFrames; 4383 pInBuffer->i16 += outFrames * channelCount; 4384 mOutBuffer.frameCount -= outFrames; 4385 mOutBuffer.i16 += outFrames * channelCount; 4386 4387 if (pInBuffer->frameCount == 0) { 4388 if (mBufferQueue.size()) { 4389 mBufferQueue.removeAt(0); 4390 delete [] pInBuffer->mBuffer; 4391 delete pInBuffer; 4392 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4393 } else { 4394 break; 4395 } 4396 } 4397 } 4398 4399 // If we could not write all frames, allocate a buffer and queue it for next time. 4400 if (inBuffer.frameCount) { 4401 sp<ThreadBase> thread = mThread.promote(); 4402 if (thread != 0 && !thread->standby()) { 4403 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4404 pInBuffer = new Buffer; 4405 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4406 pInBuffer->frameCount = inBuffer.frameCount; 4407 pInBuffer->i16 = pInBuffer->mBuffer; 4408 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4409 mBufferQueue.add(pInBuffer); 4410 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4411 } else { 4412 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4413 } 4414 } 4415 } 4416 4417 // Calling write() with a 0 length buffer, means that no more data will be written: 4418 // If no more buffers are pending, fill output track buffer to make sure it is started 4419 // by output mixer. 4420 if (frames == 0 && mBufferQueue.size() == 0) { 4421 if (mCblk->user < mCblk->frameCount) { 4422 frames = mCblk->frameCount - mCblk->user; 4423 pInBuffer = new Buffer; 4424 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4425 pInBuffer->frameCount = frames; 4426 pInBuffer->i16 = pInBuffer->mBuffer; 4427 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4428 mBufferQueue.add(pInBuffer); 4429 } else if (mActive) { 4430 stop(); 4431 } 4432 } 4433 4434 return outputBufferFull; 4435} 4436 4437status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4438{ 4439 int active; 4440 status_t result; 4441 audio_track_cblk_t* cblk = mCblk; 4442 uint32_t framesReq = buffer->frameCount; 4443 4444// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4445 buffer->frameCount = 0; 4446 4447 uint32_t framesAvail = cblk->framesAvailable(); 4448 4449 4450 if (framesAvail == 0) { 4451 Mutex::Autolock _l(cblk->lock); 4452 goto start_loop_here; 4453 while (framesAvail == 0) { 4454 active = mActive; 4455 if (CC_UNLIKELY(!active)) { 4456 ALOGV("Not active and NO_MORE_BUFFERS"); 4457 return NO_MORE_BUFFERS; 4458 } 4459 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4460 if (result != NO_ERROR) { 4461 return NO_MORE_BUFFERS; 4462 } 4463 // read the server count again 4464 start_loop_here: 4465 framesAvail = cblk->framesAvailable_l(); 4466 } 4467 } 4468 4469// if (framesAvail < framesReq) { 4470// return NO_MORE_BUFFERS; 4471// } 4472 4473 if (framesReq > framesAvail) { 4474 framesReq = framesAvail; 4475 } 4476 4477 uint32_t u = cblk->user; 4478 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4479 4480 if (u + framesReq > bufferEnd) { 4481 framesReq = bufferEnd - u; 4482 } 4483 4484 buffer->frameCount = framesReq; 4485 buffer->raw = (void *)cblk->buffer(u); 4486 return NO_ERROR; 4487} 4488 4489 4490void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4491{ 4492 size_t size = mBufferQueue.size(); 4493 4494 for (size_t i = 0; i < size; i++) { 4495 Buffer *pBuffer = mBufferQueue.itemAt(i); 4496 delete [] pBuffer->mBuffer; 4497 delete pBuffer; 4498 } 4499 mBufferQueue.clear(); 4500} 4501 4502// ---------------------------------------------------------------------------- 4503 4504AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4505 : RefBase(), 4506 mAudioFlinger(audioFlinger), 4507 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4508 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4509 mPid(pid), 4510 mTimedTrackCount(0) 4511{ 4512 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4513} 4514 4515// Client destructor must be called with AudioFlinger::mLock held 4516AudioFlinger::Client::~Client() 4517{ 4518 mAudioFlinger->removeClient_l(mPid); 4519} 4520 4521sp<MemoryDealer> AudioFlinger::Client::heap() const 4522{ 4523 return mMemoryDealer; 4524} 4525 4526// Reserve one of the limited slots for a timed audio track associated 4527// with this client 4528bool AudioFlinger::Client::reserveTimedTrack() 4529{ 4530 const int kMaxTimedTracksPerClient = 4; 4531 4532 Mutex::Autolock _l(mTimedTrackLock); 4533 4534 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4535 ALOGW("can not create timed track - pid %d has exceeded the limit", 4536 mPid); 4537 return false; 4538 } 4539 4540 mTimedTrackCount++; 4541 return true; 4542} 4543 4544// Release a slot for a timed audio track 4545void AudioFlinger::Client::releaseTimedTrack() 4546{ 4547 Mutex::Autolock _l(mTimedTrackLock); 4548 mTimedTrackCount--; 4549} 4550 4551// ---------------------------------------------------------------------------- 4552 4553AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4554 const sp<IAudioFlingerClient>& client, 4555 pid_t pid) 4556 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4557{ 4558} 4559 4560AudioFlinger::NotificationClient::~NotificationClient() 4561{ 4562} 4563 4564void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4565{ 4566 sp<NotificationClient> keep(this); 4567 mAudioFlinger->removeNotificationClient(mPid); 4568} 4569 4570// ---------------------------------------------------------------------------- 4571 4572AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4573 : BnAudioTrack(), 4574 mTrack(track) 4575{ 4576} 4577 4578AudioFlinger::TrackHandle::~TrackHandle() { 4579 // just stop the track on deletion, associated resources 4580 // will be freed from the main thread once all pending buffers have 4581 // been played. Unless it's not in the active track list, in which 4582 // case we free everything now... 4583 mTrack->destroy(); 4584} 4585 4586sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4587 return mTrack->getCblk(); 4588} 4589 4590status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4591 return mTrack->start(tid); 4592} 4593 4594void AudioFlinger::TrackHandle::stop() { 4595 mTrack->stop(); 4596} 4597 4598void AudioFlinger::TrackHandle::flush() { 4599 mTrack->flush(); 4600} 4601 4602void AudioFlinger::TrackHandle::mute(bool e) { 4603 mTrack->mute(e); 4604} 4605 4606void AudioFlinger::TrackHandle::pause() { 4607 mTrack->pause(); 4608} 4609 4610status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4611{ 4612 return mTrack->attachAuxEffect(EffectId); 4613} 4614 4615status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4616 sp<IMemory>* buffer) { 4617 if (!mTrack->isTimedTrack()) 4618 return INVALID_OPERATION; 4619 4620 PlaybackThread::TimedTrack* tt = 4621 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4622 return tt->allocateTimedBuffer(size, buffer); 4623} 4624 4625status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4626 int64_t pts) { 4627 if (!mTrack->isTimedTrack()) 4628 return INVALID_OPERATION; 4629 4630 PlaybackThread::TimedTrack* tt = 4631 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4632 return tt->queueTimedBuffer(buffer, pts); 4633} 4634 4635status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4636 const LinearTransform& xform, int target) { 4637 4638 if (!mTrack->isTimedTrack()) 4639 return INVALID_OPERATION; 4640 4641 PlaybackThread::TimedTrack* tt = 4642 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4643 return tt->setMediaTimeTransform( 4644 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4645} 4646 4647status_t AudioFlinger::TrackHandle::onTransact( 4648 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4649{ 4650 return BnAudioTrack::onTransact(code, data, reply, flags); 4651} 4652 4653// ---------------------------------------------------------------------------- 4654 4655sp<IAudioRecord> AudioFlinger::openRecord( 4656 pid_t pid, 4657 audio_io_handle_t input, 4658 uint32_t sampleRate, 4659 audio_format_t format, 4660 uint32_t channelMask, 4661 int frameCount, 4662 uint32_t flags, 4663 int *sessionId, 4664 status_t *status) 4665{ 4666 sp<RecordThread::RecordTrack> recordTrack; 4667 sp<RecordHandle> recordHandle; 4668 sp<Client> client; 4669 status_t lStatus; 4670 RecordThread *thread; 4671 size_t inFrameCount; 4672 int lSessionId; 4673 4674 // check calling permissions 4675 if (!recordingAllowed()) { 4676 lStatus = PERMISSION_DENIED; 4677 goto Exit; 4678 } 4679 4680 // add client to list 4681 { // scope for mLock 4682 Mutex::Autolock _l(mLock); 4683 thread = checkRecordThread_l(input); 4684 if (thread == NULL) { 4685 lStatus = BAD_VALUE; 4686 goto Exit; 4687 } 4688 4689 client = registerPid_l(pid); 4690 4691 // If no audio session id is provided, create one here 4692 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4693 lSessionId = *sessionId; 4694 } else { 4695 lSessionId = nextUniqueId(); 4696 if (sessionId != NULL) { 4697 *sessionId = lSessionId; 4698 } 4699 } 4700 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4701 recordTrack = thread->createRecordTrack_l(client, 4702 sampleRate, 4703 format, 4704 channelMask, 4705 frameCount, 4706 flags, 4707 lSessionId, 4708 &lStatus); 4709 } 4710 if (lStatus != NO_ERROR) { 4711 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4712 // destructor is called by the TrackBase destructor with mLock held 4713 client.clear(); 4714 recordTrack.clear(); 4715 goto Exit; 4716 } 4717 4718 // return to handle to client 4719 recordHandle = new RecordHandle(recordTrack); 4720 lStatus = NO_ERROR; 4721 4722Exit: 4723 if (status) { 4724 *status = lStatus; 4725 } 4726 return recordHandle; 4727} 4728 4729// ---------------------------------------------------------------------------- 4730 4731AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4732 : BnAudioRecord(), 4733 mRecordTrack(recordTrack) 4734{ 4735} 4736 4737AudioFlinger::RecordHandle::~RecordHandle() { 4738 stop(); 4739} 4740 4741sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4742 return mRecordTrack->getCblk(); 4743} 4744 4745status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4746 ALOGV("RecordHandle::start()"); 4747 return mRecordTrack->start(tid); 4748} 4749 4750void AudioFlinger::RecordHandle::stop() { 4751 ALOGV("RecordHandle::stop()"); 4752 mRecordTrack->stop(); 4753} 4754 4755status_t AudioFlinger::RecordHandle::onTransact( 4756 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4757{ 4758 return BnAudioRecord::onTransact(code, data, reply, flags); 4759} 4760 4761// ---------------------------------------------------------------------------- 4762 4763AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4764 AudioStreamIn *input, 4765 uint32_t sampleRate, 4766 uint32_t channels, 4767 audio_io_handle_t id, 4768 uint32_t device) : 4769 ThreadBase(audioFlinger, id, device, RECORD), 4770 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4771 // mRsmpInIndex and mInputBytes set by readInputParameters() 4772 mReqChannelCount(popcount(channels)), 4773 mReqSampleRate(sampleRate) 4774 // mBytesRead is only meaningful while active, and so is cleared in start() 4775 // (but might be better to also clear here for dump?) 4776{ 4777 snprintf(mName, kNameLength, "AudioIn_%d", id); 4778 4779 readInputParameters(); 4780} 4781 4782 4783AudioFlinger::RecordThread::~RecordThread() 4784{ 4785 delete[] mRsmpInBuffer; 4786 delete mResampler; 4787 delete[] mRsmpOutBuffer; 4788} 4789 4790void AudioFlinger::RecordThread::onFirstRef() 4791{ 4792 run(mName, PRIORITY_URGENT_AUDIO); 4793} 4794 4795status_t AudioFlinger::RecordThread::readyToRun() 4796{ 4797 status_t status = initCheck(); 4798 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4799 return status; 4800} 4801 4802bool AudioFlinger::RecordThread::threadLoop() 4803{ 4804 AudioBufferProvider::Buffer buffer; 4805 sp<RecordTrack> activeTrack; 4806 Vector< sp<EffectChain> > effectChains; 4807 4808 nsecs_t lastWarning = 0; 4809 4810 acquireWakeLock(); 4811 4812 // start recording 4813 while (!exitPending()) { 4814 4815 processConfigEvents(); 4816 4817 { // scope for mLock 4818 Mutex::Autolock _l(mLock); 4819 checkForNewParameters_l(); 4820 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4821 if (!mStandby) { 4822 mInput->stream->common.standby(&mInput->stream->common); 4823 mStandby = true; 4824 } 4825 4826 if (exitPending()) break; 4827 4828 releaseWakeLock_l(); 4829 ALOGV("RecordThread: loop stopping"); 4830 // go to sleep 4831 mWaitWorkCV.wait(mLock); 4832 ALOGV("RecordThread: loop starting"); 4833 acquireWakeLock_l(); 4834 continue; 4835 } 4836 if (mActiveTrack != 0) { 4837 if (mActiveTrack->mState == TrackBase::PAUSING) { 4838 if (!mStandby) { 4839 mInput->stream->common.standby(&mInput->stream->common); 4840 mStandby = true; 4841 } 4842 mActiveTrack.clear(); 4843 mStartStopCond.broadcast(); 4844 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4845 if (mReqChannelCount != mActiveTrack->channelCount()) { 4846 mActiveTrack.clear(); 4847 mStartStopCond.broadcast(); 4848 } else if (mBytesRead != 0) { 4849 // record start succeeds only if first read from audio input 4850 // succeeds 4851 if (mBytesRead > 0) { 4852 mActiveTrack->mState = TrackBase::ACTIVE; 4853 } else { 4854 mActiveTrack.clear(); 4855 } 4856 mStartStopCond.broadcast(); 4857 } 4858 mStandby = false; 4859 } 4860 } 4861 lockEffectChains_l(effectChains); 4862 } 4863 4864 if (mActiveTrack != 0) { 4865 if (mActiveTrack->mState != TrackBase::ACTIVE && 4866 mActiveTrack->mState != TrackBase::RESUMING) { 4867 unlockEffectChains(effectChains); 4868 usleep(kRecordThreadSleepUs); 4869 continue; 4870 } 4871 for (size_t i = 0; i < effectChains.size(); i ++) { 4872 effectChains[i]->process_l(); 4873 } 4874 4875 buffer.frameCount = mFrameCount; 4876 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4877 size_t framesOut = buffer.frameCount; 4878 if (mResampler == NULL) { 4879 // no resampling 4880 while (framesOut) { 4881 size_t framesIn = mFrameCount - mRsmpInIndex; 4882 if (framesIn) { 4883 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4884 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4885 if (framesIn > framesOut) 4886 framesIn = framesOut; 4887 mRsmpInIndex += framesIn; 4888 framesOut -= framesIn; 4889 if ((int)mChannelCount == mReqChannelCount || 4890 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4891 memcpy(dst, src, framesIn * mFrameSize); 4892 } else { 4893 int16_t *src16 = (int16_t *)src; 4894 int16_t *dst16 = (int16_t *)dst; 4895 if (mChannelCount == 1) { 4896 while (framesIn--) { 4897 *dst16++ = *src16; 4898 *dst16++ = *src16++; 4899 } 4900 } else { 4901 while (framesIn--) { 4902 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4903 src16 += 2; 4904 } 4905 } 4906 } 4907 } 4908 if (framesOut && mFrameCount == mRsmpInIndex) { 4909 if (framesOut == mFrameCount && 4910 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4911 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4912 framesOut = 0; 4913 } else { 4914 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4915 mRsmpInIndex = 0; 4916 } 4917 if (mBytesRead < 0) { 4918 ALOGE("Error reading audio input"); 4919 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4920 // Force input into standby so that it tries to 4921 // recover at next read attempt 4922 mInput->stream->common.standby(&mInput->stream->common); 4923 usleep(kRecordThreadSleepUs); 4924 } 4925 mRsmpInIndex = mFrameCount; 4926 framesOut = 0; 4927 buffer.frameCount = 0; 4928 } 4929 } 4930 } 4931 } else { 4932 // resampling 4933 4934 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4935 // alter output frame count as if we were expecting stereo samples 4936 if (mChannelCount == 1 && mReqChannelCount == 1) { 4937 framesOut >>= 1; 4938 } 4939 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4940 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4941 // are 32 bit aligned which should be always true. 4942 if (mChannelCount == 2 && mReqChannelCount == 1) { 4943 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4944 // the resampler always outputs stereo samples: do post stereo to mono conversion 4945 int16_t *src = (int16_t *)mRsmpOutBuffer; 4946 int16_t *dst = buffer.i16; 4947 while (framesOut--) { 4948 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4949 src += 2; 4950 } 4951 } else { 4952 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4953 } 4954 4955 } 4956 mActiveTrack->releaseBuffer(&buffer); 4957 mActiveTrack->overflow(); 4958 } 4959 // client isn't retrieving buffers fast enough 4960 else { 4961 if (!mActiveTrack->setOverflow()) { 4962 nsecs_t now = systemTime(); 4963 if ((now - lastWarning) > kWarningThrottleNs) { 4964 ALOGW("RecordThread: buffer overflow"); 4965 lastWarning = now; 4966 } 4967 } 4968 // Release the processor for a while before asking for a new buffer. 4969 // This will give the application more chance to read from the buffer and 4970 // clear the overflow. 4971 usleep(kRecordThreadSleepUs); 4972 } 4973 } 4974 // enable changes in effect chain 4975 unlockEffectChains(effectChains); 4976 effectChains.clear(); 4977 } 4978 4979 if (!mStandby) { 4980 mInput->stream->common.standby(&mInput->stream->common); 4981 } 4982 mActiveTrack.clear(); 4983 4984 mStartStopCond.broadcast(); 4985 4986 releaseWakeLock(); 4987 4988 ALOGV("RecordThread %p exiting", this); 4989 return false; 4990} 4991 4992 4993sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4994 const sp<AudioFlinger::Client>& client, 4995 uint32_t sampleRate, 4996 audio_format_t format, 4997 int channelMask, 4998 int frameCount, 4999 uint32_t flags, 5000 int sessionId, 5001 status_t *status) 5002{ 5003 sp<RecordTrack> track; 5004 status_t lStatus; 5005 5006 lStatus = initCheck(); 5007 if (lStatus != NO_ERROR) { 5008 ALOGE("Audio driver not initialized."); 5009 goto Exit; 5010 } 5011 5012 { // scope for mLock 5013 Mutex::Autolock _l(mLock); 5014 5015 track = new RecordTrack(this, client, sampleRate, 5016 format, channelMask, frameCount, flags, sessionId); 5017 5018 if (track->getCblk() == 0) { 5019 lStatus = NO_MEMORY; 5020 goto Exit; 5021 } 5022 5023 mTrack = track.get(); 5024 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5025 bool suspend = audio_is_bluetooth_sco_device( 5026 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5027 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5028 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5029 } 5030 lStatus = NO_ERROR; 5031 5032Exit: 5033 if (status) { 5034 *status = lStatus; 5035 } 5036 return track; 5037} 5038 5039status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5040{ 5041 ALOGV("RecordThread::start tid=%d", tid); 5042 sp <ThreadBase> strongMe = this; 5043 status_t status = NO_ERROR; 5044 { 5045 AutoMutex lock(mLock); 5046 if (mActiveTrack != 0) { 5047 if (recordTrack != mActiveTrack.get()) { 5048 status = -EBUSY; 5049 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5050 mActiveTrack->mState = TrackBase::ACTIVE; 5051 } 5052 return status; 5053 } 5054 5055 recordTrack->mState = TrackBase::IDLE; 5056 mActiveTrack = recordTrack; 5057 mLock.unlock(); 5058 status_t status = AudioSystem::startInput(mId); 5059 mLock.lock(); 5060 if (status != NO_ERROR) { 5061 mActiveTrack.clear(); 5062 return status; 5063 } 5064 mRsmpInIndex = mFrameCount; 5065 mBytesRead = 0; 5066 if (mResampler != NULL) { 5067 mResampler->reset(); 5068 } 5069 mActiveTrack->mState = TrackBase::RESUMING; 5070 // signal thread to start 5071 ALOGV("Signal record thread"); 5072 mWaitWorkCV.signal(); 5073 // do not wait for mStartStopCond if exiting 5074 if (exitPending()) { 5075 mActiveTrack.clear(); 5076 status = INVALID_OPERATION; 5077 goto startError; 5078 } 5079 mStartStopCond.wait(mLock); 5080 if (mActiveTrack == 0) { 5081 ALOGV("Record failed to start"); 5082 status = BAD_VALUE; 5083 goto startError; 5084 } 5085 ALOGV("Record started OK"); 5086 return status; 5087 } 5088startError: 5089 AudioSystem::stopInput(mId); 5090 return status; 5091} 5092 5093void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5094 ALOGV("RecordThread::stop"); 5095 sp <ThreadBase> strongMe = this; 5096 { 5097 AutoMutex lock(mLock); 5098 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5099 mActiveTrack->mState = TrackBase::PAUSING; 5100 // do not wait for mStartStopCond if exiting 5101 if (exitPending()) { 5102 return; 5103 } 5104 mStartStopCond.wait(mLock); 5105 // if we have been restarted, recordTrack == mActiveTrack.get() here 5106 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5107 mLock.unlock(); 5108 AudioSystem::stopInput(mId); 5109 mLock.lock(); 5110 ALOGV("Record stopped OK"); 5111 } 5112 } 5113 } 5114} 5115 5116status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5117{ 5118 const size_t SIZE = 256; 5119 char buffer[SIZE]; 5120 String8 result; 5121 5122 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5123 result.append(buffer); 5124 5125 if (mActiveTrack != 0) { 5126 result.append("Active Track:\n"); 5127 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5128 mActiveTrack->dump(buffer, SIZE); 5129 result.append(buffer); 5130 5131 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5132 result.append(buffer); 5133 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5134 result.append(buffer); 5135 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5136 result.append(buffer); 5137 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5138 result.append(buffer); 5139 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5140 result.append(buffer); 5141 5142 5143 } else { 5144 result.append("No record client\n"); 5145 } 5146 write(fd, result.string(), result.size()); 5147 5148 dumpBase(fd, args); 5149 dumpEffectChains(fd, args); 5150 5151 return NO_ERROR; 5152} 5153 5154// AudioBufferProvider interface 5155status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5156{ 5157 size_t framesReq = buffer->frameCount; 5158 size_t framesReady = mFrameCount - mRsmpInIndex; 5159 int channelCount; 5160 5161 if (framesReady == 0) { 5162 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5163 if (mBytesRead < 0) { 5164 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5165 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5166 // Force input into standby so that it tries to 5167 // recover at next read attempt 5168 mInput->stream->common.standby(&mInput->stream->common); 5169 usleep(kRecordThreadSleepUs); 5170 } 5171 buffer->raw = NULL; 5172 buffer->frameCount = 0; 5173 return NOT_ENOUGH_DATA; 5174 } 5175 mRsmpInIndex = 0; 5176 framesReady = mFrameCount; 5177 } 5178 5179 if (framesReq > framesReady) { 5180 framesReq = framesReady; 5181 } 5182 5183 if (mChannelCount == 1 && mReqChannelCount == 2) { 5184 channelCount = 1; 5185 } else { 5186 channelCount = 2; 5187 } 5188 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5189 buffer->frameCount = framesReq; 5190 return NO_ERROR; 5191} 5192 5193// AudioBufferProvider interface 5194void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5195{ 5196 mRsmpInIndex += buffer->frameCount; 5197 buffer->frameCount = 0; 5198} 5199 5200bool AudioFlinger::RecordThread::checkForNewParameters_l() 5201{ 5202 bool reconfig = false; 5203 5204 while (!mNewParameters.isEmpty()) { 5205 status_t status = NO_ERROR; 5206 String8 keyValuePair = mNewParameters[0]; 5207 AudioParameter param = AudioParameter(keyValuePair); 5208 int value; 5209 audio_format_t reqFormat = mFormat; 5210 int reqSamplingRate = mReqSampleRate; 5211 int reqChannelCount = mReqChannelCount; 5212 5213 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5214 reqSamplingRate = value; 5215 reconfig = true; 5216 } 5217 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5218 reqFormat = (audio_format_t) value; 5219 reconfig = true; 5220 } 5221 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5222 reqChannelCount = popcount(value); 5223 reconfig = true; 5224 } 5225 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5226 // do not accept frame count changes if tracks are open as the track buffer 5227 // size depends on frame count and correct behavior would not be guaranteed 5228 // if frame count is changed after track creation 5229 if (mActiveTrack != 0) { 5230 status = INVALID_OPERATION; 5231 } else { 5232 reconfig = true; 5233 } 5234 } 5235 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5236 // forward device change to effects that have requested to be 5237 // aware of attached audio device. 5238 for (size_t i = 0; i < mEffectChains.size(); i++) { 5239 mEffectChains[i]->setDevice_l(value); 5240 } 5241 // store input device and output device but do not forward output device to audio HAL. 5242 // Note that status is ignored by the caller for output device 5243 // (see AudioFlinger::setParameters() 5244 if (value & AUDIO_DEVICE_OUT_ALL) { 5245 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5246 status = BAD_VALUE; 5247 } else { 5248 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5249 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5250 if (mTrack != NULL) { 5251 bool suspend = audio_is_bluetooth_sco_device( 5252 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5253 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5254 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5255 } 5256 } 5257 mDevice |= (uint32_t)value; 5258 } 5259 if (status == NO_ERROR) { 5260 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5261 if (status == INVALID_OPERATION) { 5262 mInput->stream->common.standby(&mInput->stream->common); 5263 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5264 } 5265 if (reconfig) { 5266 if (status == BAD_VALUE && 5267 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5268 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5269 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5270 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5271 (reqChannelCount < 3)) { 5272 status = NO_ERROR; 5273 } 5274 if (status == NO_ERROR) { 5275 readInputParameters(); 5276 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5277 } 5278 } 5279 } 5280 5281 mNewParameters.removeAt(0); 5282 5283 mParamStatus = status; 5284 mParamCond.signal(); 5285 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5286 // already timed out waiting for the status and will never signal the condition. 5287 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5288 } 5289 return reconfig; 5290} 5291 5292String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5293{ 5294 char *s; 5295 String8 out_s8 = String8(); 5296 5297 Mutex::Autolock _l(mLock); 5298 if (initCheck() != NO_ERROR) { 5299 return out_s8; 5300 } 5301 5302 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5303 out_s8 = String8(s); 5304 free(s); 5305 return out_s8; 5306} 5307 5308void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5309 AudioSystem::OutputDescriptor desc; 5310 void *param2 = NULL; 5311 5312 switch (event) { 5313 case AudioSystem::INPUT_OPENED: 5314 case AudioSystem::INPUT_CONFIG_CHANGED: 5315 desc.channels = mChannelMask; 5316 desc.samplingRate = mSampleRate; 5317 desc.format = mFormat; 5318 desc.frameCount = mFrameCount; 5319 desc.latency = 0; 5320 param2 = &desc; 5321 break; 5322 5323 case AudioSystem::INPUT_CLOSED: 5324 default: 5325 break; 5326 } 5327 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5328} 5329 5330void AudioFlinger::RecordThread::readInputParameters() 5331{ 5332 delete mRsmpInBuffer; 5333 // mRsmpInBuffer is always assigned a new[] below 5334 delete mRsmpOutBuffer; 5335 mRsmpOutBuffer = NULL; 5336 delete mResampler; 5337 mResampler = NULL; 5338 5339 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5340 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5341 mChannelCount = (uint16_t)popcount(mChannelMask); 5342 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5343 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5344 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5345 mFrameCount = mInputBytes / mFrameSize; 5346 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5347 5348 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5349 { 5350 int channelCount; 5351 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5352 // stereo to mono post process as the resampler always outputs stereo. 5353 if (mChannelCount == 1 && mReqChannelCount == 2) { 5354 channelCount = 1; 5355 } else { 5356 channelCount = 2; 5357 } 5358 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5359 mResampler->setSampleRate(mSampleRate); 5360 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5361 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5362 5363 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5364 if (mChannelCount == 1 && mReqChannelCount == 1) { 5365 mFrameCount >>= 1; 5366 } 5367 5368 } 5369 mRsmpInIndex = mFrameCount; 5370} 5371 5372unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5373{ 5374 Mutex::Autolock _l(mLock); 5375 if (initCheck() != NO_ERROR) { 5376 return 0; 5377 } 5378 5379 return mInput->stream->get_input_frames_lost(mInput->stream); 5380} 5381 5382uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5383{ 5384 Mutex::Autolock _l(mLock); 5385 uint32_t result = 0; 5386 if (getEffectChain_l(sessionId) != 0) { 5387 result = EFFECT_SESSION; 5388 } 5389 5390 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5391 result |= TRACK_SESSION; 5392 } 5393 5394 return result; 5395} 5396 5397AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5398{ 5399 Mutex::Autolock _l(mLock); 5400 return mTrack; 5401} 5402 5403AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5404{ 5405 Mutex::Autolock _l(mLock); 5406 return mInput; 5407} 5408 5409AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5410{ 5411 Mutex::Autolock _l(mLock); 5412 AudioStreamIn *input = mInput; 5413 mInput = NULL; 5414 return input; 5415} 5416 5417// this method must always be called either with ThreadBase mLock held or inside the thread loop 5418audio_stream_t* AudioFlinger::RecordThread::stream() 5419{ 5420 if (mInput == NULL) { 5421 return NULL; 5422 } 5423 return &mInput->stream->common; 5424} 5425 5426 5427// ---------------------------------------------------------------------------- 5428 5429audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5430 uint32_t *pSamplingRate, 5431 audio_format_t *pFormat, 5432 uint32_t *pChannels, 5433 uint32_t *pLatencyMs, 5434 uint32_t flags) 5435{ 5436 status_t status; 5437 PlaybackThread *thread = NULL; 5438 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5439 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5440 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5441 uint32_t channels = pChannels ? *pChannels : 0; 5442 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5443 audio_stream_out_t *outStream; 5444 audio_hw_device_t *outHwDev; 5445 5446 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5447 pDevices ? *pDevices : 0, 5448 samplingRate, 5449 format, 5450 channels, 5451 flags); 5452 5453 if (pDevices == NULL || *pDevices == 0) { 5454 return 0; 5455 } 5456 5457 Mutex::Autolock _l(mLock); 5458 5459 outHwDev = findSuitableHwDev_l(*pDevices); 5460 if (outHwDev == NULL) 5461 return 0; 5462 5463 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5464 &channels, &samplingRate, &outStream); 5465 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5466 outStream, 5467 samplingRate, 5468 format, 5469 channels, 5470 status); 5471 5472 mHardwareStatus = AUDIO_HW_IDLE; 5473 if (outStream != NULL) { 5474 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5475 audio_io_handle_t id = nextUniqueId(); 5476 5477 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5478 (format != AUDIO_FORMAT_PCM_16_BIT) || 5479 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5480 thread = new DirectOutputThread(this, output, id, *pDevices); 5481 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5482 } else { 5483 thread = new MixerThread(this, output, id, *pDevices); 5484 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5485 } 5486 mPlaybackThreads.add(id, thread); 5487 5488 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5489 if (pFormat != NULL) *pFormat = format; 5490 if (pChannels != NULL) *pChannels = channels; 5491 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5492 5493 // notify client processes of the new output creation 5494 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5495 return id; 5496 } 5497 5498 return 0; 5499} 5500 5501audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5502 audio_io_handle_t output2) 5503{ 5504 Mutex::Autolock _l(mLock); 5505 MixerThread *thread1 = checkMixerThread_l(output1); 5506 MixerThread *thread2 = checkMixerThread_l(output2); 5507 5508 if (thread1 == NULL || thread2 == NULL) { 5509 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5510 return 0; 5511 } 5512 5513 audio_io_handle_t id = nextUniqueId(); 5514 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5515 thread->addOutputTrack(thread2); 5516 mPlaybackThreads.add(id, thread); 5517 // notify client processes of the new output creation 5518 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5519 return id; 5520} 5521 5522status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5523{ 5524 // keep strong reference on the playback thread so that 5525 // it is not destroyed while exit() is executed 5526 sp <PlaybackThread> thread; 5527 { 5528 Mutex::Autolock _l(mLock); 5529 thread = checkPlaybackThread_l(output); 5530 if (thread == NULL) { 5531 return BAD_VALUE; 5532 } 5533 5534 ALOGV("closeOutput() %d", output); 5535 5536 if (thread->type() == ThreadBase::MIXER) { 5537 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5538 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5539 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5540 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5541 } 5542 } 5543 } 5544 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5545 mPlaybackThreads.removeItem(output); 5546 } 5547 thread->exit(); 5548 // The thread entity (active unit of execution) is no longer running here, 5549 // but the ThreadBase container still exists. 5550 5551 if (thread->type() != ThreadBase::DUPLICATING) { 5552 AudioStreamOut *out = thread->clearOutput(); 5553 assert(out != NULL); 5554 // from now on thread->mOutput is NULL 5555 out->hwDev->close_output_stream(out->hwDev, out->stream); 5556 delete out; 5557 } 5558 return NO_ERROR; 5559} 5560 5561status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5562{ 5563 Mutex::Autolock _l(mLock); 5564 PlaybackThread *thread = checkPlaybackThread_l(output); 5565 5566 if (thread == NULL) { 5567 return BAD_VALUE; 5568 } 5569 5570 ALOGV("suspendOutput() %d", output); 5571 thread->suspend(); 5572 5573 return NO_ERROR; 5574} 5575 5576status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5577{ 5578 Mutex::Autolock _l(mLock); 5579 PlaybackThread *thread = checkPlaybackThread_l(output); 5580 5581 if (thread == NULL) { 5582 return BAD_VALUE; 5583 } 5584 5585 ALOGV("restoreOutput() %d", output); 5586 5587 thread->restore(); 5588 5589 return NO_ERROR; 5590} 5591 5592audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5593 uint32_t *pSamplingRate, 5594 audio_format_t *pFormat, 5595 uint32_t *pChannels, 5596 audio_in_acoustics_t acoustics) 5597{ 5598 status_t status; 5599 RecordThread *thread = NULL; 5600 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5601 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5602 uint32_t channels = pChannels ? *pChannels : 0; 5603 uint32_t reqSamplingRate = samplingRate; 5604 audio_format_t reqFormat = format; 5605 uint32_t reqChannels = channels; 5606 audio_stream_in_t *inStream; 5607 audio_hw_device_t *inHwDev; 5608 5609 if (pDevices == NULL || *pDevices == 0) { 5610 return 0; 5611 } 5612 5613 Mutex::Autolock _l(mLock); 5614 5615 inHwDev = findSuitableHwDev_l(*pDevices); 5616 if (inHwDev == NULL) 5617 return 0; 5618 5619 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5620 &channels, &samplingRate, 5621 acoustics, 5622 &inStream); 5623 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5624 inStream, 5625 samplingRate, 5626 format, 5627 channels, 5628 acoustics, 5629 status); 5630 5631 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5632 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5633 // or stereo to mono conversions on 16 bit PCM inputs. 5634 if (inStream == NULL && status == BAD_VALUE && 5635 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5636 (samplingRate <= 2 * reqSamplingRate) && 5637 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5638 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5639 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5640 &channels, &samplingRate, 5641 acoustics, 5642 &inStream); 5643 } 5644 5645 if (inStream != NULL) { 5646 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5647 5648 audio_io_handle_t id = nextUniqueId(); 5649 // Start record thread 5650 // RecorThread require both input and output device indication to forward to audio 5651 // pre processing modules 5652 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5653 thread = new RecordThread(this, 5654 input, 5655 reqSamplingRate, 5656 reqChannels, 5657 id, 5658 device); 5659 mRecordThreads.add(id, thread); 5660 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5661 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5662 if (pFormat != NULL) *pFormat = format; 5663 if (pChannels != NULL) *pChannels = reqChannels; 5664 5665 input->stream->common.standby(&input->stream->common); 5666 5667 // notify client processes of the new input creation 5668 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5669 return id; 5670 } 5671 5672 return 0; 5673} 5674 5675status_t AudioFlinger::closeInput(audio_io_handle_t input) 5676{ 5677 // keep strong reference on the record thread so that 5678 // it is not destroyed while exit() is executed 5679 sp <RecordThread> thread; 5680 { 5681 Mutex::Autolock _l(mLock); 5682 thread = checkRecordThread_l(input); 5683 if (thread == NULL) { 5684 return BAD_VALUE; 5685 } 5686 5687 ALOGV("closeInput() %d", input); 5688 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5689 mRecordThreads.removeItem(input); 5690 } 5691 thread->exit(); 5692 // The thread entity (active unit of execution) is no longer running here, 5693 // but the ThreadBase container still exists. 5694 5695 AudioStreamIn *in = thread->clearInput(); 5696 assert(in != NULL); 5697 // from now on thread->mInput is NULL 5698 in->hwDev->close_input_stream(in->hwDev, in->stream); 5699 delete in; 5700 5701 return NO_ERROR; 5702} 5703 5704status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5705{ 5706 Mutex::Autolock _l(mLock); 5707 MixerThread *dstThread = checkMixerThread_l(output); 5708 if (dstThread == NULL) { 5709 ALOGW("setStreamOutput() bad output id %d", output); 5710 return BAD_VALUE; 5711 } 5712 5713 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5714 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5715 5716 dstThread->setStreamValid(stream, true); 5717 5718 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5719 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5720 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5721 MixerThread *srcThread = (MixerThread *)thread; 5722 srcThread->setStreamValid(stream, false); 5723 srcThread->invalidateTracks(stream); 5724 } 5725 } 5726 5727 return NO_ERROR; 5728} 5729 5730 5731int AudioFlinger::newAudioSessionId() 5732{ 5733 return nextUniqueId(); 5734} 5735 5736void AudioFlinger::acquireAudioSessionId(int audioSession) 5737{ 5738 Mutex::Autolock _l(mLock); 5739 pid_t caller = IPCThreadState::self()->getCallingPid(); 5740 ALOGV("acquiring %d from %d", audioSession, caller); 5741 size_t num = mAudioSessionRefs.size(); 5742 for (size_t i = 0; i< num; i++) { 5743 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5744 if (ref->sessionid == audioSession && ref->pid == caller) { 5745 ref->cnt++; 5746 ALOGV(" incremented refcount to %d", ref->cnt); 5747 return; 5748 } 5749 } 5750 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5751 ALOGV(" added new entry for %d", audioSession); 5752} 5753 5754void AudioFlinger::releaseAudioSessionId(int audioSession) 5755{ 5756 Mutex::Autolock _l(mLock); 5757 pid_t caller = IPCThreadState::self()->getCallingPid(); 5758 ALOGV("releasing %d from %d", audioSession, caller); 5759 size_t num = mAudioSessionRefs.size(); 5760 for (size_t i = 0; i< num; i++) { 5761 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5762 if (ref->sessionid == audioSession && ref->pid == caller) { 5763 ref->cnt--; 5764 ALOGV(" decremented refcount to %d", ref->cnt); 5765 if (ref->cnt == 0) { 5766 mAudioSessionRefs.removeAt(i); 5767 delete ref; 5768 purgeStaleEffects_l(); 5769 } 5770 return; 5771 } 5772 } 5773 ALOGW("session id %d not found for pid %d", audioSession, caller); 5774} 5775 5776void AudioFlinger::purgeStaleEffects_l() { 5777 5778 ALOGV("purging stale effects"); 5779 5780 Vector< sp<EffectChain> > chains; 5781 5782 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5783 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5784 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5785 sp<EffectChain> ec = t->mEffectChains[j]; 5786 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5787 chains.push(ec); 5788 } 5789 } 5790 } 5791 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5792 sp<RecordThread> t = mRecordThreads.valueAt(i); 5793 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5794 sp<EffectChain> ec = t->mEffectChains[j]; 5795 chains.push(ec); 5796 } 5797 } 5798 5799 for (size_t i = 0; i < chains.size(); i++) { 5800 sp<EffectChain> ec = chains[i]; 5801 int sessionid = ec->sessionId(); 5802 sp<ThreadBase> t = ec->mThread.promote(); 5803 if (t == 0) { 5804 continue; 5805 } 5806 size_t numsessionrefs = mAudioSessionRefs.size(); 5807 bool found = false; 5808 for (size_t k = 0; k < numsessionrefs; k++) { 5809 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5810 if (ref->sessionid == sessionid) { 5811 ALOGV(" session %d still exists for %d with %d refs", 5812 sessionid, ref->pid, ref->cnt); 5813 found = true; 5814 break; 5815 } 5816 } 5817 if (!found) { 5818 // remove all effects from the chain 5819 while (ec->mEffects.size()) { 5820 sp<EffectModule> effect = ec->mEffects[0]; 5821 effect->unPin(); 5822 Mutex::Autolock _l (t->mLock); 5823 t->removeEffect_l(effect); 5824 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5825 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5826 if (handle != 0) { 5827 handle->mEffect.clear(); 5828 if (handle->mHasControl && handle->mEnabled) { 5829 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5830 } 5831 } 5832 } 5833 AudioSystem::unregisterEffect(effect->id()); 5834 } 5835 } 5836 } 5837 return; 5838} 5839 5840// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5841AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5842{ 5843 return mPlaybackThreads.valueFor(output).get(); 5844} 5845 5846// checkMixerThread_l() must be called with AudioFlinger::mLock held 5847AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5848{ 5849 PlaybackThread *thread = checkPlaybackThread_l(output); 5850 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5851} 5852 5853// checkRecordThread_l() must be called with AudioFlinger::mLock held 5854AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5855{ 5856 return mRecordThreads.valueFor(input).get(); 5857} 5858 5859uint32_t AudioFlinger::nextUniqueId() 5860{ 5861 return android_atomic_inc(&mNextUniqueId); 5862} 5863 5864AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5865{ 5866 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5867 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5868 AudioStreamOut *output = thread->getOutput(); 5869 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5870 return thread; 5871 } 5872 } 5873 return NULL; 5874} 5875 5876uint32_t AudioFlinger::primaryOutputDevice_l() 5877{ 5878 PlaybackThread *thread = primaryPlaybackThread_l(); 5879 5880 if (thread == NULL) { 5881 return 0; 5882 } 5883 5884 return thread->device(); 5885} 5886 5887 5888// ---------------------------------------------------------------------------- 5889// Effect management 5890// ---------------------------------------------------------------------------- 5891 5892 5893status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5894{ 5895 Mutex::Autolock _l(mLock); 5896 return EffectQueryNumberEffects(numEffects); 5897} 5898 5899status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5900{ 5901 Mutex::Autolock _l(mLock); 5902 return EffectQueryEffect(index, descriptor); 5903} 5904 5905status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5906 effect_descriptor_t *descriptor) const 5907{ 5908 Mutex::Autolock _l(mLock); 5909 return EffectGetDescriptor(pUuid, descriptor); 5910} 5911 5912 5913sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5914 effect_descriptor_t *pDesc, 5915 const sp<IEffectClient>& effectClient, 5916 int32_t priority, 5917 audio_io_handle_t io, 5918 int sessionId, 5919 status_t *status, 5920 int *id, 5921 int *enabled) 5922{ 5923 status_t lStatus = NO_ERROR; 5924 sp<EffectHandle> handle; 5925 effect_descriptor_t desc; 5926 5927 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5928 pid, effectClient.get(), priority, sessionId, io); 5929 5930 if (pDesc == NULL) { 5931 lStatus = BAD_VALUE; 5932 goto Exit; 5933 } 5934 5935 // check audio settings permission for global effects 5936 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5937 lStatus = PERMISSION_DENIED; 5938 goto Exit; 5939 } 5940 5941 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5942 // that can only be created by audio policy manager (running in same process) 5943 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5944 lStatus = PERMISSION_DENIED; 5945 goto Exit; 5946 } 5947 5948 if (io == 0) { 5949 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5950 // output must be specified by AudioPolicyManager when using session 5951 // AUDIO_SESSION_OUTPUT_STAGE 5952 lStatus = BAD_VALUE; 5953 goto Exit; 5954 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5955 // if the output returned by getOutputForEffect() is removed before we lock the 5956 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5957 // and we will exit safely 5958 io = AudioSystem::getOutputForEffect(&desc); 5959 } 5960 } 5961 5962 { 5963 Mutex::Autolock _l(mLock); 5964 5965 5966 if (!EffectIsNullUuid(&pDesc->uuid)) { 5967 // if uuid is specified, request effect descriptor 5968 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5969 if (lStatus < 0) { 5970 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5971 goto Exit; 5972 } 5973 } else { 5974 // if uuid is not specified, look for an available implementation 5975 // of the required type in effect factory 5976 if (EffectIsNullUuid(&pDesc->type)) { 5977 ALOGW("createEffect() no effect type"); 5978 lStatus = BAD_VALUE; 5979 goto Exit; 5980 } 5981 uint32_t numEffects = 0; 5982 effect_descriptor_t d; 5983 d.flags = 0; // prevent compiler warning 5984 bool found = false; 5985 5986 lStatus = EffectQueryNumberEffects(&numEffects); 5987 if (lStatus < 0) { 5988 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5989 goto Exit; 5990 } 5991 for (uint32_t i = 0; i < numEffects; i++) { 5992 lStatus = EffectQueryEffect(i, &desc); 5993 if (lStatus < 0) { 5994 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5995 continue; 5996 } 5997 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5998 // If matching type found save effect descriptor. If the session is 5999 // 0 and the effect is not auxiliary, continue enumeration in case 6000 // an auxiliary version of this effect type is available 6001 found = true; 6002 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6003 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6004 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6005 break; 6006 } 6007 } 6008 } 6009 if (!found) { 6010 lStatus = BAD_VALUE; 6011 ALOGW("createEffect() effect not found"); 6012 goto Exit; 6013 } 6014 // For same effect type, chose auxiliary version over insert version if 6015 // connect to output mix (Compliance to OpenSL ES) 6016 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6017 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6018 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6019 } 6020 } 6021 6022 // Do not allow auxiliary effects on a session different from 0 (output mix) 6023 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6024 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6025 lStatus = INVALID_OPERATION; 6026 goto Exit; 6027 } 6028 6029 // check recording permission for visualizer 6030 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6031 !recordingAllowed()) { 6032 lStatus = PERMISSION_DENIED; 6033 goto Exit; 6034 } 6035 6036 // return effect descriptor 6037 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6038 6039 // If output is not specified try to find a matching audio session ID in one of the 6040 // output threads. 6041 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6042 // because of code checking output when entering the function. 6043 // Note: io is never 0 when creating an effect on an input 6044 if (io == 0) { 6045 // look for the thread where the specified audio session is present 6046 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6047 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6048 io = mPlaybackThreads.keyAt(i); 6049 break; 6050 } 6051 } 6052 if (io == 0) { 6053 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6054 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6055 io = mRecordThreads.keyAt(i); 6056 break; 6057 } 6058 } 6059 } 6060 // If no output thread contains the requested session ID, default to 6061 // first output. The effect chain will be moved to the correct output 6062 // thread when a track with the same session ID is created 6063 if (io == 0 && mPlaybackThreads.size()) { 6064 io = mPlaybackThreads.keyAt(0); 6065 } 6066 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6067 } 6068 ThreadBase *thread = checkRecordThread_l(io); 6069 if (thread == NULL) { 6070 thread = checkPlaybackThread_l(io); 6071 if (thread == NULL) { 6072 ALOGE("createEffect() unknown output thread"); 6073 lStatus = BAD_VALUE; 6074 goto Exit; 6075 } 6076 } 6077 6078 sp<Client> client = registerPid_l(pid); 6079 6080 // create effect on selected output thread 6081 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6082 &desc, enabled, &lStatus); 6083 if (handle != 0 && id != NULL) { 6084 *id = handle->id(); 6085 } 6086 } 6087 6088Exit: 6089 if(status) { 6090 *status = lStatus; 6091 } 6092 return handle; 6093} 6094 6095status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6096 audio_io_handle_t dstOutput) 6097{ 6098 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6099 sessionId, srcOutput, dstOutput); 6100 Mutex::Autolock _l(mLock); 6101 if (srcOutput == dstOutput) { 6102 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6103 return NO_ERROR; 6104 } 6105 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6106 if (srcThread == NULL) { 6107 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6108 return BAD_VALUE; 6109 } 6110 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6111 if (dstThread == NULL) { 6112 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6113 return BAD_VALUE; 6114 } 6115 6116 Mutex::Autolock _dl(dstThread->mLock); 6117 Mutex::Autolock _sl(srcThread->mLock); 6118 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6119 6120 return NO_ERROR; 6121} 6122 6123// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6124status_t AudioFlinger::moveEffectChain_l(int sessionId, 6125 AudioFlinger::PlaybackThread *srcThread, 6126 AudioFlinger::PlaybackThread *dstThread, 6127 bool reRegister) 6128{ 6129 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6130 sessionId, srcThread, dstThread); 6131 6132 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6133 if (chain == 0) { 6134 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6135 sessionId, srcThread); 6136 return INVALID_OPERATION; 6137 } 6138 6139 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6140 // so that a new chain is created with correct parameters when first effect is added. This is 6141 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6142 // removed. 6143 srcThread->removeEffectChain_l(chain); 6144 6145 // transfer all effects one by one so that new effect chain is created on new thread with 6146 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6147 audio_io_handle_t dstOutput = dstThread->id(); 6148 sp<EffectChain> dstChain; 6149 uint32_t strategy = 0; // prevent compiler warning 6150 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6151 while (effect != 0) { 6152 srcThread->removeEffect_l(effect); 6153 dstThread->addEffect_l(effect); 6154 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6155 if (effect->state() == EffectModule::ACTIVE || 6156 effect->state() == EffectModule::STOPPING) { 6157 effect->start(); 6158 } 6159 // if the move request is not received from audio policy manager, the effect must be 6160 // re-registered with the new strategy and output 6161 if (dstChain == 0) { 6162 dstChain = effect->chain().promote(); 6163 if (dstChain == 0) { 6164 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6165 srcThread->addEffect_l(effect); 6166 return NO_INIT; 6167 } 6168 strategy = dstChain->strategy(); 6169 } 6170 if (reRegister) { 6171 AudioSystem::unregisterEffect(effect->id()); 6172 AudioSystem::registerEffect(&effect->desc(), 6173 dstOutput, 6174 strategy, 6175 sessionId, 6176 effect->id()); 6177 } 6178 effect = chain->getEffectFromId_l(0); 6179 } 6180 6181 return NO_ERROR; 6182} 6183 6184 6185// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6186sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6187 const sp<AudioFlinger::Client>& client, 6188 const sp<IEffectClient>& effectClient, 6189 int32_t priority, 6190 int sessionId, 6191 effect_descriptor_t *desc, 6192 int *enabled, 6193 status_t *status 6194 ) 6195{ 6196 sp<EffectModule> effect; 6197 sp<EffectHandle> handle; 6198 status_t lStatus; 6199 sp<EffectChain> chain; 6200 bool chainCreated = false; 6201 bool effectCreated = false; 6202 bool effectRegistered = false; 6203 6204 lStatus = initCheck(); 6205 if (lStatus != NO_ERROR) { 6206 ALOGW("createEffect_l() Audio driver not initialized."); 6207 goto Exit; 6208 } 6209 6210 // Do not allow effects with session ID 0 on direct output or duplicating threads 6211 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6212 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6213 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6214 desc->name, sessionId); 6215 lStatus = BAD_VALUE; 6216 goto Exit; 6217 } 6218 // Only Pre processor effects are allowed on input threads and only on input threads 6219 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6220 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6221 desc->name, desc->flags, mType); 6222 lStatus = BAD_VALUE; 6223 goto Exit; 6224 } 6225 6226 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6227 6228 { // scope for mLock 6229 Mutex::Autolock _l(mLock); 6230 6231 // check for existing effect chain with the requested audio session 6232 chain = getEffectChain_l(sessionId); 6233 if (chain == 0) { 6234 // create a new chain for this session 6235 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6236 chain = new EffectChain(this, sessionId); 6237 addEffectChain_l(chain); 6238 chain->setStrategy(getStrategyForSession_l(sessionId)); 6239 chainCreated = true; 6240 } else { 6241 effect = chain->getEffectFromDesc_l(desc); 6242 } 6243 6244 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6245 6246 if (effect == 0) { 6247 int id = mAudioFlinger->nextUniqueId(); 6248 // Check CPU and memory usage 6249 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6250 if (lStatus != NO_ERROR) { 6251 goto Exit; 6252 } 6253 effectRegistered = true; 6254 // create a new effect module if none present in the chain 6255 effect = new EffectModule(this, chain, desc, id, sessionId); 6256 lStatus = effect->status(); 6257 if (lStatus != NO_ERROR) { 6258 goto Exit; 6259 } 6260 lStatus = chain->addEffect_l(effect); 6261 if (lStatus != NO_ERROR) { 6262 goto Exit; 6263 } 6264 effectCreated = true; 6265 6266 effect->setDevice(mDevice); 6267 effect->setMode(mAudioFlinger->getMode()); 6268 } 6269 // create effect handle and connect it to effect module 6270 handle = new EffectHandle(effect, client, effectClient, priority); 6271 lStatus = effect->addHandle(handle); 6272 if (enabled != NULL) { 6273 *enabled = (int)effect->isEnabled(); 6274 } 6275 } 6276 6277Exit: 6278 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6279 Mutex::Autolock _l(mLock); 6280 if (effectCreated) { 6281 chain->removeEffect_l(effect); 6282 } 6283 if (effectRegistered) { 6284 AudioSystem::unregisterEffect(effect->id()); 6285 } 6286 if (chainCreated) { 6287 removeEffectChain_l(chain); 6288 } 6289 handle.clear(); 6290 } 6291 6292 if(status) { 6293 *status = lStatus; 6294 } 6295 return handle; 6296} 6297 6298sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6299{ 6300 sp<EffectChain> chain = getEffectChain_l(sessionId); 6301 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6302} 6303 6304// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6305// PlaybackThread::mLock held 6306status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6307{ 6308 // check for existing effect chain with the requested audio session 6309 int sessionId = effect->sessionId(); 6310 sp<EffectChain> chain = getEffectChain_l(sessionId); 6311 bool chainCreated = false; 6312 6313 if (chain == 0) { 6314 // create a new chain for this session 6315 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6316 chain = new EffectChain(this, sessionId); 6317 addEffectChain_l(chain); 6318 chain->setStrategy(getStrategyForSession_l(sessionId)); 6319 chainCreated = true; 6320 } 6321 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6322 6323 if (chain->getEffectFromId_l(effect->id()) != 0) { 6324 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6325 this, effect->desc().name, chain.get()); 6326 return BAD_VALUE; 6327 } 6328 6329 status_t status = chain->addEffect_l(effect); 6330 if (status != NO_ERROR) { 6331 if (chainCreated) { 6332 removeEffectChain_l(chain); 6333 } 6334 return status; 6335 } 6336 6337 effect->setDevice(mDevice); 6338 effect->setMode(mAudioFlinger->getMode()); 6339 return NO_ERROR; 6340} 6341 6342void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6343 6344 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6345 effect_descriptor_t desc = effect->desc(); 6346 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6347 detachAuxEffect_l(effect->id()); 6348 } 6349 6350 sp<EffectChain> chain = effect->chain().promote(); 6351 if (chain != 0) { 6352 // remove effect chain if removing last effect 6353 if (chain->removeEffect_l(effect) == 0) { 6354 removeEffectChain_l(chain); 6355 } 6356 } else { 6357 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6358 } 6359} 6360 6361void AudioFlinger::ThreadBase::lockEffectChains_l( 6362 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6363{ 6364 effectChains = mEffectChains; 6365 for (size_t i = 0; i < mEffectChains.size(); i++) { 6366 mEffectChains[i]->lock(); 6367 } 6368} 6369 6370void AudioFlinger::ThreadBase::unlockEffectChains( 6371 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6372{ 6373 for (size_t i = 0; i < effectChains.size(); i++) { 6374 effectChains[i]->unlock(); 6375 } 6376} 6377 6378sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6379{ 6380 Mutex::Autolock _l(mLock); 6381 return getEffectChain_l(sessionId); 6382} 6383 6384sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6385{ 6386 size_t size = mEffectChains.size(); 6387 for (size_t i = 0; i < size; i++) { 6388 if (mEffectChains[i]->sessionId() == sessionId) { 6389 return mEffectChains[i]; 6390 } 6391 } 6392 return 0; 6393} 6394 6395void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6396{ 6397 Mutex::Autolock _l(mLock); 6398 size_t size = mEffectChains.size(); 6399 for (size_t i = 0; i < size; i++) { 6400 mEffectChains[i]->setMode_l(mode); 6401 } 6402} 6403 6404void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6405 const wp<EffectHandle>& handle, 6406 bool unpinIfLast) { 6407 6408 Mutex::Autolock _l(mLock); 6409 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6410 // delete the effect module if removing last handle on it 6411 if (effect->removeHandle(handle) == 0) { 6412 if (!effect->isPinned() || unpinIfLast) { 6413 removeEffect_l(effect); 6414 AudioSystem::unregisterEffect(effect->id()); 6415 } 6416 } 6417} 6418 6419status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6420{ 6421 int session = chain->sessionId(); 6422 int16_t *buffer = mMixBuffer; 6423 bool ownsBuffer = false; 6424 6425 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6426 if (session > 0) { 6427 // Only one effect chain can be present in direct output thread and it uses 6428 // the mix buffer as input 6429 if (mType != DIRECT) { 6430 size_t numSamples = mFrameCount * mChannelCount; 6431 buffer = new int16_t[numSamples]; 6432 memset(buffer, 0, numSamples * sizeof(int16_t)); 6433 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6434 ownsBuffer = true; 6435 } 6436 6437 // Attach all tracks with same session ID to this chain. 6438 for (size_t i = 0; i < mTracks.size(); ++i) { 6439 sp<Track> track = mTracks[i]; 6440 if (session == track->sessionId()) { 6441 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6442 track->setMainBuffer(buffer); 6443 chain->incTrackCnt(); 6444 } 6445 } 6446 6447 // indicate all active tracks in the chain 6448 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6449 sp<Track> track = mActiveTracks[i].promote(); 6450 if (track == 0) continue; 6451 if (session == track->sessionId()) { 6452 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6453 chain->incActiveTrackCnt(); 6454 } 6455 } 6456 } 6457 6458 chain->setInBuffer(buffer, ownsBuffer); 6459 chain->setOutBuffer(mMixBuffer); 6460 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6461 // chains list in order to be processed last as it contains output stage effects 6462 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6463 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6464 // after track specific effects and before output stage 6465 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6466 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6467 // Effect chain for other sessions are inserted at beginning of effect 6468 // chains list to be processed before output mix effects. Relative order between other 6469 // sessions is not important 6470 size_t size = mEffectChains.size(); 6471 size_t i = 0; 6472 for (i = 0; i < size; i++) { 6473 if (mEffectChains[i]->sessionId() < session) break; 6474 } 6475 mEffectChains.insertAt(chain, i); 6476 checkSuspendOnAddEffectChain_l(chain); 6477 6478 return NO_ERROR; 6479} 6480 6481size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6482{ 6483 int session = chain->sessionId(); 6484 6485 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6486 6487 for (size_t i = 0; i < mEffectChains.size(); i++) { 6488 if (chain == mEffectChains[i]) { 6489 mEffectChains.removeAt(i); 6490 // detach all active tracks from the chain 6491 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6492 sp<Track> track = mActiveTracks[i].promote(); 6493 if (track == 0) continue; 6494 if (session == track->sessionId()) { 6495 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6496 chain.get(), session); 6497 chain->decActiveTrackCnt(); 6498 } 6499 } 6500 6501 // detach all tracks with same session ID from this chain 6502 for (size_t i = 0; i < mTracks.size(); ++i) { 6503 sp<Track> track = mTracks[i]; 6504 if (session == track->sessionId()) { 6505 track->setMainBuffer(mMixBuffer); 6506 chain->decTrackCnt(); 6507 } 6508 } 6509 break; 6510 } 6511 } 6512 return mEffectChains.size(); 6513} 6514 6515status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6516 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6517{ 6518 Mutex::Autolock _l(mLock); 6519 return attachAuxEffect_l(track, EffectId); 6520} 6521 6522status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6523 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6524{ 6525 status_t status = NO_ERROR; 6526 6527 if (EffectId == 0) { 6528 track->setAuxBuffer(0, NULL); 6529 } else { 6530 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6531 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6532 if (effect != 0) { 6533 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6534 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6535 } else { 6536 status = INVALID_OPERATION; 6537 } 6538 } else { 6539 status = BAD_VALUE; 6540 } 6541 } 6542 return status; 6543} 6544 6545void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6546{ 6547 for (size_t i = 0; i < mTracks.size(); ++i) { 6548 sp<Track> track = mTracks[i]; 6549 if (track->auxEffectId() == effectId) { 6550 attachAuxEffect_l(track, 0); 6551 } 6552 } 6553} 6554 6555status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6556{ 6557 // only one chain per input thread 6558 if (mEffectChains.size() != 0) { 6559 return INVALID_OPERATION; 6560 } 6561 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6562 6563 chain->setInBuffer(NULL); 6564 chain->setOutBuffer(NULL); 6565 6566 checkSuspendOnAddEffectChain_l(chain); 6567 6568 mEffectChains.add(chain); 6569 6570 return NO_ERROR; 6571} 6572 6573size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6574{ 6575 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6576 ALOGW_IF(mEffectChains.size() != 1, 6577 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6578 chain.get(), mEffectChains.size(), this); 6579 if (mEffectChains.size() == 1) { 6580 mEffectChains.removeAt(0); 6581 } 6582 return 0; 6583} 6584 6585// ---------------------------------------------------------------------------- 6586// EffectModule implementation 6587// ---------------------------------------------------------------------------- 6588 6589#undef LOG_TAG 6590#define LOG_TAG "AudioFlinger::EffectModule" 6591 6592AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6593 const wp<AudioFlinger::EffectChain>& chain, 6594 effect_descriptor_t *desc, 6595 int id, 6596 int sessionId) 6597 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6598 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6599{ 6600 ALOGV("Constructor %p", this); 6601 int lStatus; 6602 if (thread == NULL) { 6603 return; 6604 } 6605 6606 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6607 6608 // create effect engine from effect factory 6609 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6610 6611 if (mStatus != NO_ERROR) { 6612 return; 6613 } 6614 lStatus = init(); 6615 if (lStatus < 0) { 6616 mStatus = lStatus; 6617 goto Error; 6618 } 6619 6620 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6621 mPinned = true; 6622 } 6623 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6624 return; 6625Error: 6626 EffectRelease(mEffectInterface); 6627 mEffectInterface = NULL; 6628 ALOGV("Constructor Error %d", mStatus); 6629} 6630 6631AudioFlinger::EffectModule::~EffectModule() 6632{ 6633 ALOGV("Destructor %p", this); 6634 if (mEffectInterface != NULL) { 6635 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6636 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6637 sp<ThreadBase> thread = mThread.promote(); 6638 if (thread != 0) { 6639 audio_stream_t *stream = thread->stream(); 6640 if (stream != NULL) { 6641 stream->remove_audio_effect(stream, mEffectInterface); 6642 } 6643 } 6644 } 6645 // release effect engine 6646 EffectRelease(mEffectInterface); 6647 } 6648} 6649 6650status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6651{ 6652 status_t status; 6653 6654 Mutex::Autolock _l(mLock); 6655 int priority = handle->priority(); 6656 size_t size = mHandles.size(); 6657 sp<EffectHandle> h; 6658 size_t i; 6659 for (i = 0; i < size; i++) { 6660 h = mHandles[i].promote(); 6661 if (h == 0) continue; 6662 if (h->priority() <= priority) break; 6663 } 6664 // if inserted in first place, move effect control from previous owner to this handle 6665 if (i == 0) { 6666 bool enabled = false; 6667 if (h != 0) { 6668 enabled = h->enabled(); 6669 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6670 } 6671 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6672 status = NO_ERROR; 6673 } else { 6674 status = ALREADY_EXISTS; 6675 } 6676 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6677 mHandles.insertAt(handle, i); 6678 return status; 6679} 6680 6681size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6682{ 6683 Mutex::Autolock _l(mLock); 6684 size_t size = mHandles.size(); 6685 size_t i; 6686 for (i = 0; i < size; i++) { 6687 if (mHandles[i] == handle) break; 6688 } 6689 if (i == size) { 6690 return size; 6691 } 6692 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6693 6694 bool enabled = false; 6695 EffectHandle *hdl = handle.unsafe_get(); 6696 if (hdl != NULL) { 6697 ALOGV("removeHandle() unsafe_get OK"); 6698 enabled = hdl->enabled(); 6699 } 6700 mHandles.removeAt(i); 6701 size = mHandles.size(); 6702 // if removed from first place, move effect control from this handle to next in line 6703 if (i == 0 && size != 0) { 6704 sp<EffectHandle> h = mHandles[0].promote(); 6705 if (h != 0) { 6706 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6707 } 6708 } 6709 6710 // Prevent calls to process() and other functions on effect interface from now on. 6711 // The effect engine will be released by the destructor when the last strong reference on 6712 // this object is released which can happen after next process is called. 6713 if (size == 0 && !mPinned) { 6714 mState = DESTROYED; 6715 } 6716 6717 return size; 6718} 6719 6720sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6721{ 6722 Mutex::Autolock _l(mLock); 6723 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6724} 6725 6726void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6727{ 6728 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6729 // keep a strong reference on this EffectModule to avoid calling the 6730 // destructor before we exit 6731 sp<EffectModule> keep(this); 6732 { 6733 sp<ThreadBase> thread = mThread.promote(); 6734 if (thread != 0) { 6735 thread->disconnectEffect(keep, handle, unpinIfLast); 6736 } 6737 } 6738} 6739 6740void AudioFlinger::EffectModule::updateState() { 6741 Mutex::Autolock _l(mLock); 6742 6743 switch (mState) { 6744 case RESTART: 6745 reset_l(); 6746 // FALL THROUGH 6747 6748 case STARTING: 6749 // clear auxiliary effect input buffer for next accumulation 6750 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6751 memset(mConfig.inputCfg.buffer.raw, 6752 0, 6753 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6754 } 6755 start_l(); 6756 mState = ACTIVE; 6757 break; 6758 case STOPPING: 6759 stop_l(); 6760 mDisableWaitCnt = mMaxDisableWaitCnt; 6761 mState = STOPPED; 6762 break; 6763 case STOPPED: 6764 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6765 // turn off sequence. 6766 if (--mDisableWaitCnt == 0) { 6767 reset_l(); 6768 mState = IDLE; 6769 } 6770 break; 6771 default: //IDLE , ACTIVE, DESTROYED 6772 break; 6773 } 6774} 6775 6776void AudioFlinger::EffectModule::process() 6777{ 6778 Mutex::Autolock _l(mLock); 6779 6780 if (mState == DESTROYED || mEffectInterface == NULL || 6781 mConfig.inputCfg.buffer.raw == NULL || 6782 mConfig.outputCfg.buffer.raw == NULL) { 6783 return; 6784 } 6785 6786 if (isProcessEnabled()) { 6787 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6788 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6789 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6790 mConfig.inputCfg.buffer.s32, 6791 mConfig.inputCfg.buffer.frameCount/2); 6792 } 6793 6794 // do the actual processing in the effect engine 6795 int ret = (*mEffectInterface)->process(mEffectInterface, 6796 &mConfig.inputCfg.buffer, 6797 &mConfig.outputCfg.buffer); 6798 6799 // force transition to IDLE state when engine is ready 6800 if (mState == STOPPED && ret == -ENODATA) { 6801 mDisableWaitCnt = 1; 6802 } 6803 6804 // clear auxiliary effect input buffer for next accumulation 6805 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6806 memset(mConfig.inputCfg.buffer.raw, 0, 6807 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6808 } 6809 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6810 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6811 // If an insert effect is idle and input buffer is different from output buffer, 6812 // accumulate input onto output 6813 sp<EffectChain> chain = mChain.promote(); 6814 if (chain != 0 && chain->activeTrackCnt() != 0) { 6815 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6816 int16_t *in = mConfig.inputCfg.buffer.s16; 6817 int16_t *out = mConfig.outputCfg.buffer.s16; 6818 for (size_t i = 0; i < frameCnt; i++) { 6819 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6820 } 6821 } 6822 } 6823} 6824 6825void AudioFlinger::EffectModule::reset_l() 6826{ 6827 if (mEffectInterface == NULL) { 6828 return; 6829 } 6830 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6831} 6832 6833status_t AudioFlinger::EffectModule::configure() 6834{ 6835 uint32_t channels; 6836 if (mEffectInterface == NULL) { 6837 return NO_INIT; 6838 } 6839 6840 sp<ThreadBase> thread = mThread.promote(); 6841 if (thread == 0) { 6842 return DEAD_OBJECT; 6843 } 6844 6845 // TODO: handle configuration of effects replacing track process 6846 if (thread->channelCount() == 1) { 6847 channels = AUDIO_CHANNEL_OUT_MONO; 6848 } else { 6849 channels = AUDIO_CHANNEL_OUT_STEREO; 6850 } 6851 6852 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6853 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6854 } else { 6855 mConfig.inputCfg.channels = channels; 6856 } 6857 mConfig.outputCfg.channels = channels; 6858 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6859 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6860 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6861 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6862 mConfig.inputCfg.bufferProvider.cookie = NULL; 6863 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6864 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6865 mConfig.outputCfg.bufferProvider.cookie = NULL; 6866 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6867 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6868 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6869 // Insert effect: 6870 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6871 // always overwrites output buffer: input buffer == output buffer 6872 // - in other sessions: 6873 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6874 // other effect: overwrites output buffer: input buffer == output buffer 6875 // Auxiliary effect: 6876 // accumulates in output buffer: input buffer != output buffer 6877 // Therefore: accumulate <=> input buffer != output buffer 6878 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6879 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6880 } else { 6881 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6882 } 6883 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6884 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6885 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6886 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6887 6888 ALOGV("configure() %p thread %p buffer %p framecount %d", 6889 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6890 6891 status_t cmdStatus; 6892 uint32_t size = sizeof(int); 6893 status_t status = (*mEffectInterface)->command(mEffectInterface, 6894 EFFECT_CMD_SET_CONFIG, 6895 sizeof(effect_config_t), 6896 &mConfig, 6897 &size, 6898 &cmdStatus); 6899 if (status == 0) { 6900 status = cmdStatus; 6901 } 6902 6903 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6904 (1000 * mConfig.outputCfg.buffer.frameCount); 6905 6906 return status; 6907} 6908 6909status_t AudioFlinger::EffectModule::init() 6910{ 6911 Mutex::Autolock _l(mLock); 6912 if (mEffectInterface == NULL) { 6913 return NO_INIT; 6914 } 6915 status_t cmdStatus; 6916 uint32_t size = sizeof(status_t); 6917 status_t status = (*mEffectInterface)->command(mEffectInterface, 6918 EFFECT_CMD_INIT, 6919 0, 6920 NULL, 6921 &size, 6922 &cmdStatus); 6923 if (status == 0) { 6924 status = cmdStatus; 6925 } 6926 return status; 6927} 6928 6929status_t AudioFlinger::EffectModule::start() 6930{ 6931 Mutex::Autolock _l(mLock); 6932 return start_l(); 6933} 6934 6935status_t AudioFlinger::EffectModule::start_l() 6936{ 6937 if (mEffectInterface == NULL) { 6938 return NO_INIT; 6939 } 6940 status_t cmdStatus; 6941 uint32_t size = sizeof(status_t); 6942 status_t status = (*mEffectInterface)->command(mEffectInterface, 6943 EFFECT_CMD_ENABLE, 6944 0, 6945 NULL, 6946 &size, 6947 &cmdStatus); 6948 if (status == 0) { 6949 status = cmdStatus; 6950 } 6951 if (status == 0 && 6952 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6953 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6954 sp<ThreadBase> thread = mThread.promote(); 6955 if (thread != 0) { 6956 audio_stream_t *stream = thread->stream(); 6957 if (stream != NULL) { 6958 stream->add_audio_effect(stream, mEffectInterface); 6959 } 6960 } 6961 } 6962 return status; 6963} 6964 6965status_t AudioFlinger::EffectModule::stop() 6966{ 6967 Mutex::Autolock _l(mLock); 6968 return stop_l(); 6969} 6970 6971status_t AudioFlinger::EffectModule::stop_l() 6972{ 6973 if (mEffectInterface == NULL) { 6974 return NO_INIT; 6975 } 6976 status_t cmdStatus; 6977 uint32_t size = sizeof(status_t); 6978 status_t status = (*mEffectInterface)->command(mEffectInterface, 6979 EFFECT_CMD_DISABLE, 6980 0, 6981 NULL, 6982 &size, 6983 &cmdStatus); 6984 if (status == 0) { 6985 status = cmdStatus; 6986 } 6987 if (status == 0 && 6988 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6989 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6990 sp<ThreadBase> thread = mThread.promote(); 6991 if (thread != 0) { 6992 audio_stream_t *stream = thread->stream(); 6993 if (stream != NULL) { 6994 stream->remove_audio_effect(stream, mEffectInterface); 6995 } 6996 } 6997 } 6998 return status; 6999} 7000 7001status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7002 uint32_t cmdSize, 7003 void *pCmdData, 7004 uint32_t *replySize, 7005 void *pReplyData) 7006{ 7007 Mutex::Autolock _l(mLock); 7008// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7009 7010 if (mState == DESTROYED || mEffectInterface == NULL) { 7011 return NO_INIT; 7012 } 7013 status_t status = (*mEffectInterface)->command(mEffectInterface, 7014 cmdCode, 7015 cmdSize, 7016 pCmdData, 7017 replySize, 7018 pReplyData); 7019 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7020 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7021 for (size_t i = 1; i < mHandles.size(); i++) { 7022 sp<EffectHandle> h = mHandles[i].promote(); 7023 if (h != 0) { 7024 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7025 } 7026 } 7027 } 7028 return status; 7029} 7030 7031status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7032{ 7033 7034 Mutex::Autolock _l(mLock); 7035 ALOGV("setEnabled %p enabled %d", this, enabled); 7036 7037 if (enabled != isEnabled()) { 7038 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7039 if (enabled && status != NO_ERROR) { 7040 return status; 7041 } 7042 7043 switch (mState) { 7044 // going from disabled to enabled 7045 case IDLE: 7046 mState = STARTING; 7047 break; 7048 case STOPPED: 7049 mState = RESTART; 7050 break; 7051 case STOPPING: 7052 mState = ACTIVE; 7053 break; 7054 7055 // going from enabled to disabled 7056 case RESTART: 7057 mState = STOPPED; 7058 break; 7059 case STARTING: 7060 mState = IDLE; 7061 break; 7062 case ACTIVE: 7063 mState = STOPPING; 7064 break; 7065 case DESTROYED: 7066 return NO_ERROR; // simply ignore as we are being destroyed 7067 } 7068 for (size_t i = 1; i < mHandles.size(); i++) { 7069 sp<EffectHandle> h = mHandles[i].promote(); 7070 if (h != 0) { 7071 h->setEnabled(enabled); 7072 } 7073 } 7074 } 7075 return NO_ERROR; 7076} 7077 7078bool AudioFlinger::EffectModule::isEnabled() const 7079{ 7080 switch (mState) { 7081 case RESTART: 7082 case STARTING: 7083 case ACTIVE: 7084 return true; 7085 case IDLE: 7086 case STOPPING: 7087 case STOPPED: 7088 case DESTROYED: 7089 default: 7090 return false; 7091 } 7092} 7093 7094bool AudioFlinger::EffectModule::isProcessEnabled() const 7095{ 7096 switch (mState) { 7097 case RESTART: 7098 case ACTIVE: 7099 case STOPPING: 7100 case STOPPED: 7101 return true; 7102 case IDLE: 7103 case STARTING: 7104 case DESTROYED: 7105 default: 7106 return false; 7107 } 7108} 7109 7110status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7111{ 7112 Mutex::Autolock _l(mLock); 7113 status_t status = NO_ERROR; 7114 7115 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7116 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7117 if (isProcessEnabled() && 7118 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7119 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7120 status_t cmdStatus; 7121 uint32_t volume[2]; 7122 uint32_t *pVolume = NULL; 7123 uint32_t size = sizeof(volume); 7124 volume[0] = *left; 7125 volume[1] = *right; 7126 if (controller) { 7127 pVolume = volume; 7128 } 7129 status = (*mEffectInterface)->command(mEffectInterface, 7130 EFFECT_CMD_SET_VOLUME, 7131 size, 7132 volume, 7133 &size, 7134 pVolume); 7135 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7136 *left = volume[0]; 7137 *right = volume[1]; 7138 } 7139 } 7140 return status; 7141} 7142 7143status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7144{ 7145 Mutex::Autolock _l(mLock); 7146 status_t status = NO_ERROR; 7147 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7148 // audio pre processing modules on RecordThread can receive both output and 7149 // input device indication in the same call 7150 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7151 if (dev) { 7152 status_t cmdStatus; 7153 uint32_t size = sizeof(status_t); 7154 7155 status = (*mEffectInterface)->command(mEffectInterface, 7156 EFFECT_CMD_SET_DEVICE, 7157 sizeof(uint32_t), 7158 &dev, 7159 &size, 7160 &cmdStatus); 7161 if (status == NO_ERROR) { 7162 status = cmdStatus; 7163 } 7164 } 7165 dev = device & AUDIO_DEVICE_IN_ALL; 7166 if (dev) { 7167 status_t cmdStatus; 7168 uint32_t size = sizeof(status_t); 7169 7170 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7171 EFFECT_CMD_SET_INPUT_DEVICE, 7172 sizeof(uint32_t), 7173 &dev, 7174 &size, 7175 &cmdStatus); 7176 if (status2 == NO_ERROR) { 7177 status2 = cmdStatus; 7178 } 7179 if (status == NO_ERROR) { 7180 status = status2; 7181 } 7182 } 7183 } 7184 return status; 7185} 7186 7187status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7188{ 7189 Mutex::Autolock _l(mLock); 7190 status_t status = NO_ERROR; 7191 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7192 status_t cmdStatus; 7193 uint32_t size = sizeof(status_t); 7194 status = (*mEffectInterface)->command(mEffectInterface, 7195 EFFECT_CMD_SET_AUDIO_MODE, 7196 sizeof(audio_mode_t), 7197 &mode, 7198 &size, 7199 &cmdStatus); 7200 if (status == NO_ERROR) { 7201 status = cmdStatus; 7202 } 7203 } 7204 return status; 7205} 7206 7207void AudioFlinger::EffectModule::setSuspended(bool suspended) 7208{ 7209 Mutex::Autolock _l(mLock); 7210 mSuspended = suspended; 7211} 7212 7213bool AudioFlinger::EffectModule::suspended() const 7214{ 7215 Mutex::Autolock _l(mLock); 7216 return mSuspended; 7217} 7218 7219status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7220{ 7221 const size_t SIZE = 256; 7222 char buffer[SIZE]; 7223 String8 result; 7224 7225 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7226 result.append(buffer); 7227 7228 bool locked = tryLock(mLock); 7229 // failed to lock - AudioFlinger is probably deadlocked 7230 if (!locked) { 7231 result.append("\t\tCould not lock Fx mutex:\n"); 7232 } 7233 7234 result.append("\t\tSession Status State Engine:\n"); 7235 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7236 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7237 result.append(buffer); 7238 7239 result.append("\t\tDescriptor:\n"); 7240 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7241 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7242 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7243 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7244 result.append(buffer); 7245 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7246 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7247 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7248 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7249 result.append(buffer); 7250 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7251 mDescriptor.apiVersion, 7252 mDescriptor.flags); 7253 result.append(buffer); 7254 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7255 mDescriptor.name); 7256 result.append(buffer); 7257 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7258 mDescriptor.implementor); 7259 result.append(buffer); 7260 7261 result.append("\t\t- Input configuration:\n"); 7262 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7263 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7264 (uint32_t)mConfig.inputCfg.buffer.raw, 7265 mConfig.inputCfg.buffer.frameCount, 7266 mConfig.inputCfg.samplingRate, 7267 mConfig.inputCfg.channels, 7268 mConfig.inputCfg.format); 7269 result.append(buffer); 7270 7271 result.append("\t\t- Output configuration:\n"); 7272 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7273 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7274 (uint32_t)mConfig.outputCfg.buffer.raw, 7275 mConfig.outputCfg.buffer.frameCount, 7276 mConfig.outputCfg.samplingRate, 7277 mConfig.outputCfg.channels, 7278 mConfig.outputCfg.format); 7279 result.append(buffer); 7280 7281 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7282 result.append(buffer); 7283 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7284 for (size_t i = 0; i < mHandles.size(); ++i) { 7285 sp<EffectHandle> handle = mHandles[i].promote(); 7286 if (handle != 0) { 7287 handle->dump(buffer, SIZE); 7288 result.append(buffer); 7289 } 7290 } 7291 7292 result.append("\n"); 7293 7294 write(fd, result.string(), result.length()); 7295 7296 if (locked) { 7297 mLock.unlock(); 7298 } 7299 7300 return NO_ERROR; 7301} 7302 7303// ---------------------------------------------------------------------------- 7304// EffectHandle implementation 7305// ---------------------------------------------------------------------------- 7306 7307#undef LOG_TAG 7308#define LOG_TAG "AudioFlinger::EffectHandle" 7309 7310AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7311 const sp<AudioFlinger::Client>& client, 7312 const sp<IEffectClient>& effectClient, 7313 int32_t priority) 7314 : BnEffect(), 7315 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7316 mPriority(priority), mHasControl(false), mEnabled(false) 7317{ 7318 ALOGV("constructor %p", this); 7319 7320 if (client == 0) { 7321 return; 7322 } 7323 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7324 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7325 if (mCblkMemory != 0) { 7326 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7327 7328 if (mCblk != NULL) { 7329 new(mCblk) effect_param_cblk_t(); 7330 mBuffer = (uint8_t *)mCblk + bufOffset; 7331 } 7332 } else { 7333 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7334 return; 7335 } 7336} 7337 7338AudioFlinger::EffectHandle::~EffectHandle() 7339{ 7340 ALOGV("Destructor %p", this); 7341 disconnect(false); 7342 ALOGV("Destructor DONE %p", this); 7343} 7344 7345status_t AudioFlinger::EffectHandle::enable() 7346{ 7347 ALOGV("enable %p", this); 7348 if (!mHasControl) return INVALID_OPERATION; 7349 if (mEffect == 0) return DEAD_OBJECT; 7350 7351 if (mEnabled) { 7352 return NO_ERROR; 7353 } 7354 7355 mEnabled = true; 7356 7357 sp<ThreadBase> thread = mEffect->thread().promote(); 7358 if (thread != 0) { 7359 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7360 } 7361 7362 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7363 if (mEffect->suspended()) { 7364 return NO_ERROR; 7365 } 7366 7367 status_t status = mEffect->setEnabled(true); 7368 if (status != NO_ERROR) { 7369 if (thread != 0) { 7370 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7371 } 7372 mEnabled = false; 7373 } 7374 return status; 7375} 7376 7377status_t AudioFlinger::EffectHandle::disable() 7378{ 7379 ALOGV("disable %p", this); 7380 if (!mHasControl) return INVALID_OPERATION; 7381 if (mEffect == 0) return DEAD_OBJECT; 7382 7383 if (!mEnabled) { 7384 return NO_ERROR; 7385 } 7386 mEnabled = false; 7387 7388 if (mEffect->suspended()) { 7389 return NO_ERROR; 7390 } 7391 7392 status_t status = mEffect->setEnabled(false); 7393 7394 sp<ThreadBase> thread = mEffect->thread().promote(); 7395 if (thread != 0) { 7396 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7397 } 7398 7399 return status; 7400} 7401 7402void AudioFlinger::EffectHandle::disconnect() 7403{ 7404 disconnect(true); 7405} 7406 7407void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7408{ 7409 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7410 if (mEffect == 0) { 7411 return; 7412 } 7413 mEffect->disconnect(this, unpinIfLast); 7414 7415 if (mHasControl && mEnabled) { 7416 sp<ThreadBase> thread = mEffect->thread().promote(); 7417 if (thread != 0) { 7418 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7419 } 7420 } 7421 7422 // release sp on module => module destructor can be called now 7423 mEffect.clear(); 7424 if (mClient != 0) { 7425 if (mCblk != NULL) { 7426 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7427 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7428 } 7429 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7430 // Client destructor must run with AudioFlinger mutex locked 7431 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7432 mClient.clear(); 7433 } 7434} 7435 7436status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7437 uint32_t cmdSize, 7438 void *pCmdData, 7439 uint32_t *replySize, 7440 void *pReplyData) 7441{ 7442// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7443// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7444 7445 // only get parameter command is permitted for applications not controlling the effect 7446 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7447 return INVALID_OPERATION; 7448 } 7449 if (mEffect == 0) return DEAD_OBJECT; 7450 if (mClient == 0) return INVALID_OPERATION; 7451 7452 // handle commands that are not forwarded transparently to effect engine 7453 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7454 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7455 // no risk to block the whole media server process or mixer threads is we are stuck here 7456 Mutex::Autolock _l(mCblk->lock); 7457 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7458 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7459 mCblk->serverIndex = 0; 7460 mCblk->clientIndex = 0; 7461 return BAD_VALUE; 7462 } 7463 status_t status = NO_ERROR; 7464 while (mCblk->serverIndex < mCblk->clientIndex) { 7465 int reply; 7466 uint32_t rsize = sizeof(int); 7467 int *p = (int *)(mBuffer + mCblk->serverIndex); 7468 int size = *p++; 7469 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7470 ALOGW("command(): invalid parameter block size"); 7471 break; 7472 } 7473 effect_param_t *param = (effect_param_t *)p; 7474 if (param->psize == 0 || param->vsize == 0) { 7475 ALOGW("command(): null parameter or value size"); 7476 mCblk->serverIndex += size; 7477 continue; 7478 } 7479 uint32_t psize = sizeof(effect_param_t) + 7480 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7481 param->vsize; 7482 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7483 psize, 7484 p, 7485 &rsize, 7486 &reply); 7487 // stop at first error encountered 7488 if (ret != NO_ERROR) { 7489 status = ret; 7490 *(int *)pReplyData = reply; 7491 break; 7492 } else if (reply != NO_ERROR) { 7493 *(int *)pReplyData = reply; 7494 break; 7495 } 7496 mCblk->serverIndex += size; 7497 } 7498 mCblk->serverIndex = 0; 7499 mCblk->clientIndex = 0; 7500 return status; 7501 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7502 *(int *)pReplyData = NO_ERROR; 7503 return enable(); 7504 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7505 *(int *)pReplyData = NO_ERROR; 7506 return disable(); 7507 } 7508 7509 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7510} 7511 7512void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7513{ 7514 ALOGV("setControl %p control %d", this, hasControl); 7515 7516 mHasControl = hasControl; 7517 mEnabled = enabled; 7518 7519 if (signal && mEffectClient != 0) { 7520 mEffectClient->controlStatusChanged(hasControl); 7521 } 7522} 7523 7524void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7525 uint32_t cmdSize, 7526 void *pCmdData, 7527 uint32_t replySize, 7528 void *pReplyData) 7529{ 7530 if (mEffectClient != 0) { 7531 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7532 } 7533} 7534 7535 7536 7537void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7538{ 7539 if (mEffectClient != 0) { 7540 mEffectClient->enableStatusChanged(enabled); 7541 } 7542} 7543 7544status_t AudioFlinger::EffectHandle::onTransact( 7545 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7546{ 7547 return BnEffect::onTransact(code, data, reply, flags); 7548} 7549 7550 7551void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7552{ 7553 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7554 7555 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7556 (mClient == 0) ? getpid_cached : mClient->pid(), 7557 mPriority, 7558 mHasControl, 7559 !locked, 7560 mCblk ? mCblk->clientIndex : 0, 7561 mCblk ? mCblk->serverIndex : 0 7562 ); 7563 7564 if (locked) { 7565 mCblk->lock.unlock(); 7566 } 7567} 7568 7569#undef LOG_TAG 7570#define LOG_TAG "AudioFlinger::EffectChain" 7571 7572AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7573 int sessionId) 7574 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7575 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7576 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7577{ 7578 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7579 if (thread == NULL) { 7580 return; 7581 } 7582 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7583 thread->frameCount(); 7584} 7585 7586AudioFlinger::EffectChain::~EffectChain() 7587{ 7588 if (mOwnInBuffer) { 7589 delete mInBuffer; 7590 } 7591 7592} 7593 7594// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7595sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7596{ 7597 size_t size = mEffects.size(); 7598 7599 for (size_t i = 0; i < size; i++) { 7600 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7601 return mEffects[i]; 7602 } 7603 } 7604 return 0; 7605} 7606 7607// getEffectFromId_l() must be called with ThreadBase::mLock held 7608sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7609{ 7610 size_t size = mEffects.size(); 7611 7612 for (size_t i = 0; i < size; i++) { 7613 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7614 if (id == 0 || mEffects[i]->id() == id) { 7615 return mEffects[i]; 7616 } 7617 } 7618 return 0; 7619} 7620 7621// getEffectFromType_l() must be called with ThreadBase::mLock held 7622sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7623 const effect_uuid_t *type) 7624{ 7625 size_t size = mEffects.size(); 7626 7627 for (size_t i = 0; i < size; i++) { 7628 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7629 return mEffects[i]; 7630 } 7631 } 7632 return 0; 7633} 7634 7635// Must be called with EffectChain::mLock locked 7636void AudioFlinger::EffectChain::process_l() 7637{ 7638 sp<ThreadBase> thread = mThread.promote(); 7639 if (thread == 0) { 7640 ALOGW("process_l(): cannot promote mixer thread"); 7641 return; 7642 } 7643 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7644 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7645 // always process effects unless no more tracks are on the session and the effect tail 7646 // has been rendered 7647 bool doProcess = true; 7648 if (!isGlobalSession) { 7649 bool tracksOnSession = (trackCnt() != 0); 7650 7651 if (!tracksOnSession && mTailBufferCount == 0) { 7652 doProcess = false; 7653 } 7654 7655 if (activeTrackCnt() == 0) { 7656 // if no track is active and the effect tail has not been rendered, 7657 // the input buffer must be cleared here as the mixer process will not do it 7658 if (tracksOnSession || mTailBufferCount > 0) { 7659 size_t numSamples = thread->frameCount() * thread->channelCount(); 7660 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7661 if (mTailBufferCount > 0) { 7662 mTailBufferCount--; 7663 } 7664 } 7665 } 7666 } 7667 7668 size_t size = mEffects.size(); 7669 if (doProcess) { 7670 for (size_t i = 0; i < size; i++) { 7671 mEffects[i]->process(); 7672 } 7673 } 7674 for (size_t i = 0; i < size; i++) { 7675 mEffects[i]->updateState(); 7676 } 7677} 7678 7679// addEffect_l() must be called with PlaybackThread::mLock held 7680status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7681{ 7682 effect_descriptor_t desc = effect->desc(); 7683 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7684 7685 Mutex::Autolock _l(mLock); 7686 effect->setChain(this); 7687 sp<ThreadBase> thread = mThread.promote(); 7688 if (thread == 0) { 7689 return NO_INIT; 7690 } 7691 effect->setThread(thread); 7692 7693 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7694 // Auxiliary effects are inserted at the beginning of mEffects vector as 7695 // they are processed first and accumulated in chain input buffer 7696 mEffects.insertAt(effect, 0); 7697 7698 // the input buffer for auxiliary effect contains mono samples in 7699 // 32 bit format. This is to avoid saturation in AudoMixer 7700 // accumulation stage. Saturation is done in EffectModule::process() before 7701 // calling the process in effect engine 7702 size_t numSamples = thread->frameCount(); 7703 int32_t *buffer = new int32_t[numSamples]; 7704 memset(buffer, 0, numSamples * sizeof(int32_t)); 7705 effect->setInBuffer((int16_t *)buffer); 7706 // auxiliary effects output samples to chain input buffer for further processing 7707 // by insert effects 7708 effect->setOutBuffer(mInBuffer); 7709 } else { 7710 // Insert effects are inserted at the end of mEffects vector as they are processed 7711 // after track and auxiliary effects. 7712 // Insert effect order as a function of indicated preference: 7713 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7714 // another effect is present 7715 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7716 // last effect claiming first position 7717 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7718 // first effect claiming last position 7719 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7720 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7721 // already present 7722 7723 size_t size = mEffects.size(); 7724 size_t idx_insert = size; 7725 ssize_t idx_insert_first = -1; 7726 ssize_t idx_insert_last = -1; 7727 7728 for (size_t i = 0; i < size; i++) { 7729 effect_descriptor_t d = mEffects[i]->desc(); 7730 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7731 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7732 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7733 // check invalid effect chaining combinations 7734 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7735 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7736 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7737 return INVALID_OPERATION; 7738 } 7739 // remember position of first insert effect and by default 7740 // select this as insert position for new effect 7741 if (idx_insert == size) { 7742 idx_insert = i; 7743 } 7744 // remember position of last insert effect claiming 7745 // first position 7746 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7747 idx_insert_first = i; 7748 } 7749 // remember position of first insert effect claiming 7750 // last position 7751 if (iPref == EFFECT_FLAG_INSERT_LAST && 7752 idx_insert_last == -1) { 7753 idx_insert_last = i; 7754 } 7755 } 7756 } 7757 7758 // modify idx_insert from first position if needed 7759 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7760 if (idx_insert_last != -1) { 7761 idx_insert = idx_insert_last; 7762 } else { 7763 idx_insert = size; 7764 } 7765 } else { 7766 if (idx_insert_first != -1) { 7767 idx_insert = idx_insert_first + 1; 7768 } 7769 } 7770 7771 // always read samples from chain input buffer 7772 effect->setInBuffer(mInBuffer); 7773 7774 // if last effect in the chain, output samples to chain 7775 // output buffer, otherwise to chain input buffer 7776 if (idx_insert == size) { 7777 if (idx_insert != 0) { 7778 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7779 mEffects[idx_insert-1]->configure(); 7780 } 7781 effect->setOutBuffer(mOutBuffer); 7782 } else { 7783 effect->setOutBuffer(mInBuffer); 7784 } 7785 mEffects.insertAt(effect, idx_insert); 7786 7787 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7788 } 7789 effect->configure(); 7790 return NO_ERROR; 7791} 7792 7793// removeEffect_l() must be called with PlaybackThread::mLock held 7794size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7795{ 7796 Mutex::Autolock _l(mLock); 7797 size_t size = mEffects.size(); 7798 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7799 7800 for (size_t i = 0; i < size; i++) { 7801 if (effect == mEffects[i]) { 7802 // calling stop here will remove pre-processing effect from the audio HAL. 7803 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7804 // the middle of a read from audio HAL 7805 if (mEffects[i]->state() == EffectModule::ACTIVE || 7806 mEffects[i]->state() == EffectModule::STOPPING) { 7807 mEffects[i]->stop(); 7808 } 7809 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7810 delete[] effect->inBuffer(); 7811 } else { 7812 if (i == size - 1 && i != 0) { 7813 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7814 mEffects[i - 1]->configure(); 7815 } 7816 } 7817 mEffects.removeAt(i); 7818 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7819 break; 7820 } 7821 } 7822 7823 return mEffects.size(); 7824} 7825 7826// setDevice_l() must be called with PlaybackThread::mLock held 7827void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7828{ 7829 size_t size = mEffects.size(); 7830 for (size_t i = 0; i < size; i++) { 7831 mEffects[i]->setDevice(device); 7832 } 7833} 7834 7835// setMode_l() must be called with PlaybackThread::mLock held 7836void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7837{ 7838 size_t size = mEffects.size(); 7839 for (size_t i = 0; i < size; i++) { 7840 mEffects[i]->setMode(mode); 7841 } 7842} 7843 7844// setVolume_l() must be called with PlaybackThread::mLock held 7845bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7846{ 7847 uint32_t newLeft = *left; 7848 uint32_t newRight = *right; 7849 bool hasControl = false; 7850 int ctrlIdx = -1; 7851 size_t size = mEffects.size(); 7852 7853 // first update volume controller 7854 for (size_t i = size; i > 0; i--) { 7855 if (mEffects[i - 1]->isProcessEnabled() && 7856 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7857 ctrlIdx = i - 1; 7858 hasControl = true; 7859 break; 7860 } 7861 } 7862 7863 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7864 if (hasControl) { 7865 *left = mNewLeftVolume; 7866 *right = mNewRightVolume; 7867 } 7868 return hasControl; 7869 } 7870 7871 mVolumeCtrlIdx = ctrlIdx; 7872 mLeftVolume = newLeft; 7873 mRightVolume = newRight; 7874 7875 // second get volume update from volume controller 7876 if (ctrlIdx >= 0) { 7877 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7878 mNewLeftVolume = newLeft; 7879 mNewRightVolume = newRight; 7880 } 7881 // then indicate volume to all other effects in chain. 7882 // Pass altered volume to effects before volume controller 7883 // and requested volume to effects after controller 7884 uint32_t lVol = newLeft; 7885 uint32_t rVol = newRight; 7886 7887 for (size_t i = 0; i < size; i++) { 7888 if ((int)i == ctrlIdx) continue; 7889 // this also works for ctrlIdx == -1 when there is no volume controller 7890 if ((int)i > ctrlIdx) { 7891 lVol = *left; 7892 rVol = *right; 7893 } 7894 mEffects[i]->setVolume(&lVol, &rVol, false); 7895 } 7896 *left = newLeft; 7897 *right = newRight; 7898 7899 return hasControl; 7900} 7901 7902status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7903{ 7904 const size_t SIZE = 256; 7905 char buffer[SIZE]; 7906 String8 result; 7907 7908 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7909 result.append(buffer); 7910 7911 bool locked = tryLock(mLock); 7912 // failed to lock - AudioFlinger is probably deadlocked 7913 if (!locked) { 7914 result.append("\tCould not lock mutex:\n"); 7915 } 7916 7917 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7918 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7919 mEffects.size(), 7920 (uint32_t)mInBuffer, 7921 (uint32_t)mOutBuffer, 7922 mActiveTrackCnt); 7923 result.append(buffer); 7924 write(fd, result.string(), result.size()); 7925 7926 for (size_t i = 0; i < mEffects.size(); ++i) { 7927 sp<EffectModule> effect = mEffects[i]; 7928 if (effect != 0) { 7929 effect->dump(fd, args); 7930 } 7931 } 7932 7933 if (locked) { 7934 mLock.unlock(); 7935 } 7936 7937 return NO_ERROR; 7938} 7939 7940// must be called with ThreadBase::mLock held 7941void AudioFlinger::EffectChain::setEffectSuspended_l( 7942 const effect_uuid_t *type, bool suspend) 7943{ 7944 sp<SuspendedEffectDesc> desc; 7945 // use effect type UUID timelow as key as there is no real risk of identical 7946 // timeLow fields among effect type UUIDs. 7947 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7948 if (suspend) { 7949 if (index >= 0) { 7950 desc = mSuspendedEffects.valueAt(index); 7951 } else { 7952 desc = new SuspendedEffectDesc(); 7953 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7954 mSuspendedEffects.add(type->timeLow, desc); 7955 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7956 } 7957 if (desc->mRefCount++ == 0) { 7958 sp<EffectModule> effect = getEffectIfEnabled(type); 7959 if (effect != 0) { 7960 desc->mEffect = effect; 7961 effect->setSuspended(true); 7962 effect->setEnabled(false); 7963 } 7964 } 7965 } else { 7966 if (index < 0) { 7967 return; 7968 } 7969 desc = mSuspendedEffects.valueAt(index); 7970 if (desc->mRefCount <= 0) { 7971 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7972 desc->mRefCount = 1; 7973 } 7974 if (--desc->mRefCount == 0) { 7975 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7976 if (desc->mEffect != 0) { 7977 sp<EffectModule> effect = desc->mEffect.promote(); 7978 if (effect != 0) { 7979 effect->setSuspended(false); 7980 sp<EffectHandle> handle = effect->controlHandle(); 7981 if (handle != 0) { 7982 effect->setEnabled(handle->enabled()); 7983 } 7984 } 7985 desc->mEffect.clear(); 7986 } 7987 mSuspendedEffects.removeItemsAt(index); 7988 } 7989 } 7990} 7991 7992// must be called with ThreadBase::mLock held 7993void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7994{ 7995 sp<SuspendedEffectDesc> desc; 7996 7997 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7998 if (suspend) { 7999 if (index >= 0) { 8000 desc = mSuspendedEffects.valueAt(index); 8001 } else { 8002 desc = new SuspendedEffectDesc(); 8003 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8004 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8005 } 8006 if (desc->mRefCount++ == 0) { 8007 Vector< sp<EffectModule> > effects; 8008 getSuspendEligibleEffects(effects); 8009 for (size_t i = 0; i < effects.size(); i++) { 8010 setEffectSuspended_l(&effects[i]->desc().type, true); 8011 } 8012 } 8013 } else { 8014 if (index < 0) { 8015 return; 8016 } 8017 desc = mSuspendedEffects.valueAt(index); 8018 if (desc->mRefCount <= 0) { 8019 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8020 desc->mRefCount = 1; 8021 } 8022 if (--desc->mRefCount == 0) { 8023 Vector<const effect_uuid_t *> types; 8024 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8025 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8026 continue; 8027 } 8028 types.add(&mSuspendedEffects.valueAt(i)->mType); 8029 } 8030 for (size_t i = 0; i < types.size(); i++) { 8031 setEffectSuspended_l(types[i], false); 8032 } 8033 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8034 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8035 } 8036 } 8037} 8038 8039 8040// The volume effect is used for automated tests only 8041#ifndef OPENSL_ES_H_ 8042static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8043 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8044const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8045#endif //OPENSL_ES_H_ 8046 8047bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8048{ 8049 // auxiliary effects and visualizer are never suspended on output mix 8050 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8051 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8052 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8053 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8054 return false; 8055 } 8056 return true; 8057} 8058 8059void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8060{ 8061 effects.clear(); 8062 for (size_t i = 0; i < mEffects.size(); i++) { 8063 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8064 effects.add(mEffects[i]); 8065 } 8066 } 8067} 8068 8069sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8070 const effect_uuid_t *type) 8071{ 8072 sp<EffectModule> effect = getEffectFromType_l(type); 8073 return effect != 0 && effect->isEnabled() ? effect : 0; 8074} 8075 8076void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8077 bool enabled) 8078{ 8079 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8080 if (enabled) { 8081 if (index < 0) { 8082 // if the effect is not suspend check if all effects are suspended 8083 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8084 if (index < 0) { 8085 return; 8086 } 8087 if (!isEffectEligibleForSuspend(effect->desc())) { 8088 return; 8089 } 8090 setEffectSuspended_l(&effect->desc().type, enabled); 8091 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8092 if (index < 0) { 8093 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8094 return; 8095 } 8096 } 8097 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8098 effect->desc().type.timeLow); 8099 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8100 // if effect is requested to suspended but was not yet enabled, supend it now. 8101 if (desc->mEffect == 0) { 8102 desc->mEffect = effect; 8103 effect->setEnabled(false); 8104 effect->setSuspended(true); 8105 } 8106 } else { 8107 if (index < 0) { 8108 return; 8109 } 8110 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8111 effect->desc().type.timeLow); 8112 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8113 desc->mEffect.clear(); 8114 effect->setSuspended(false); 8115 } 8116} 8117 8118#undef LOG_TAG 8119#define LOG_TAG "AudioFlinger" 8120 8121// ---------------------------------------------------------------------------- 8122 8123status_t AudioFlinger::onTransact( 8124 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8125{ 8126 return BnAudioFlinger::onTransact(code, data, reply, flags); 8127} 8128 8129}; // namespace android 8130