AudioFlinger.cpp revision 8abf44d2f2bcd20a2835570efe89d89c19db426a
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_IDLE; 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_IDLE; 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 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 826 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 827 // check calling permissions 828 if (!settingsAllowed()) { 829 return PERMISSION_DENIED; 830 } 831 832 // ioHandle == 0 means the parameters are global to the audio hardware interface 833 if (ioHandle == 0) { 834 status_t final_result = NO_ERROR; 835 { 836 AutoMutex lock(mHardwareLock); 837 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 838 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 839 audio_hw_device_t *dev = mAudioHwDevs[i]; 840 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 841 final_result = result ?: final_result; 842 } 843 mHardwareStatus = AUDIO_HW_IDLE; 844 } 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 char *s; 908 { 909 AutoMutex lock(mHardwareLock); 910 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 911 audio_hw_device_t *dev = mAudioHwDevs[i]; 912 s = dev->get_parameters(dev, keys.string()); 913 mHardwareStatus = AUDIO_HW_IDLE; 914 } 915 out_s8 += String8(s ? s : ""); 916 free(s); 917 } 918 return out_s8; 919 } 920 921 Mutex::Autolock _l(mLock); 922 923 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 924 if (playbackThread != NULL) { 925 return playbackThread->getParameters(keys); 926 } 927 RecordThread *recordThread = checkRecordThread_l(ioHandle); 928 if (recordThread != NULL) { 929 return recordThread->getParameters(keys); 930 } 931 return String8(""); 932} 933 934size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 935{ 936 status_t ret = initCheck(); 937 if (ret != NO_ERROR) { 938 return 0; 939 } 940 941 AutoMutex lock(mHardwareLock); 942 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 943 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 944 mHardwareStatus = AUDIO_HW_IDLE; 945 return size; 946} 947 948unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 949{ 950 if (ioHandle == 0) { 951 return 0; 952 } 953 954 Mutex::Autolock _l(mLock); 955 956 RecordThread *recordThread = checkRecordThread_l(ioHandle); 957 if (recordThread != NULL) { 958 return recordThread->getInputFramesLost(); 959 } 960 return 0; 961} 962 963status_t AudioFlinger::setVoiceVolume(float value) 964{ 965 status_t ret = initCheck(); 966 if (ret != NO_ERROR) { 967 return ret; 968 } 969 970 // check calling permissions 971 if (!settingsAllowed()) { 972 return PERMISSION_DENIED; 973 } 974 975 AutoMutex lock(mHardwareLock); 976 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 977 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 978 mHardwareStatus = AUDIO_HW_IDLE; 979 980 return ret; 981} 982 983status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 984 audio_io_handle_t output) const 985{ 986 status_t status; 987 988 Mutex::Autolock _l(mLock); 989 990 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 991 if (playbackThread != NULL) { 992 return playbackThread->getRenderPosition(halFrames, dspFrames); 993 } 994 995 return BAD_VALUE; 996} 997 998void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 999{ 1000 1001 Mutex::Autolock _l(mLock); 1002 1003 pid_t pid = IPCThreadState::self()->getCallingPid(); 1004 if (mNotificationClients.indexOfKey(pid) < 0) { 1005 sp<NotificationClient> notificationClient = new NotificationClient(this, 1006 client, 1007 pid); 1008 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1009 1010 mNotificationClients.add(pid, notificationClient); 1011 1012 sp<IBinder> binder = client->asBinder(); 1013 binder->linkToDeath(notificationClient); 1014 1015 // the config change is always sent from playback or record threads to avoid deadlock 1016 // with AudioSystem::gLock 1017 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1018 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1019 } 1020 1021 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1022 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1023 } 1024 } 1025} 1026 1027void AudioFlinger::removeNotificationClient(pid_t pid) 1028{ 1029 Mutex::Autolock _l(mLock); 1030 1031 ssize_t index = mNotificationClients.indexOfKey(pid); 1032 if (index >= 0) { 1033 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 1034 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 1035 mNotificationClients.removeItem(pid); 1036 } 1037 1038 ALOGV("%d died, releasing its sessions", pid); 1039 size_t num = mAudioSessionRefs.size(); 1040 bool removed = false; 1041 for (size_t i = 0; i< num; ) { 1042 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1043 ALOGV(" pid %d @ %d", ref->pid, i); 1044 if (ref->pid == pid) { 1045 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1046 mAudioSessionRefs.removeAt(i); 1047 delete ref; 1048 removed = true; 1049 num--; 1050 } else { 1051 i++; 1052 } 1053 } 1054 if (removed) { 1055 purgeStaleEffects_l(); 1056 } 1057} 1058 1059// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1060void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1061{ 1062 size_t size = mNotificationClients.size(); 1063 for (size_t i = 0; i < size; i++) { 1064 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1065 param2); 1066 } 1067} 1068 1069// removeClient_l() must be called with AudioFlinger::mLock held 1070void AudioFlinger::removeClient_l(pid_t pid) 1071{ 1072 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1073 mClients.removeItem(pid); 1074} 1075 1076 1077// ---------------------------------------------------------------------------- 1078 1079AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1080 uint32_t device, type_t type) 1081 : Thread(false), 1082 mType(type), 1083 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1084 // mChannelMask 1085 mChannelCount(0), 1086 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1087 mParamStatus(NO_ERROR), 1088 mStandby(false), mId(id), 1089 mDevice(device), 1090 mDeathRecipient(new PMDeathRecipient(this)) 1091{ 1092} 1093 1094AudioFlinger::ThreadBase::~ThreadBase() 1095{ 1096 mParamCond.broadcast(); 1097 // do not lock the mutex in destructor 1098 releaseWakeLock_l(); 1099 if (mPowerManager != 0) { 1100 sp<IBinder> binder = mPowerManager->asBinder(); 1101 binder->unlinkToDeath(mDeathRecipient); 1102 } 1103} 1104 1105void AudioFlinger::ThreadBase::exit() 1106{ 1107 ALOGV("ThreadBase::exit"); 1108 { 1109 // This lock prevents the following race in thread (uniprocessor for illustration): 1110 // if (!exitPending()) { 1111 // // context switch from here to exit() 1112 // // exit() calls requestExit(), what exitPending() observes 1113 // // exit() calls signal(), which is dropped since no waiters 1114 // // context switch back from exit() to here 1115 // mWaitWorkCV.wait(...); 1116 // // now thread is hung 1117 // } 1118 AutoMutex lock(mLock); 1119 requestExit(); 1120 mWaitWorkCV.signal(); 1121 } 1122 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1123 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1124 requestExitAndWait(); 1125} 1126 1127status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1128{ 1129 status_t status; 1130 1131 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1132 Mutex::Autolock _l(mLock); 1133 1134 mNewParameters.add(keyValuePairs); 1135 mWaitWorkCV.signal(); 1136 // wait condition with timeout in case the thread loop has exited 1137 // before the request could be processed 1138 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1139 status = mParamStatus; 1140 mWaitWorkCV.signal(); 1141 } else { 1142 status = TIMED_OUT; 1143 } 1144 return status; 1145} 1146 1147void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1148{ 1149 Mutex::Autolock _l(mLock); 1150 sendConfigEvent_l(event, param); 1151} 1152 1153// sendConfigEvent_l() must be called with ThreadBase::mLock held 1154void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1155{ 1156 ConfigEvent configEvent; 1157 configEvent.mEvent = event; 1158 configEvent.mParam = param; 1159 mConfigEvents.add(configEvent); 1160 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1161 mWaitWorkCV.signal(); 1162} 1163 1164void AudioFlinger::ThreadBase::processConfigEvents() 1165{ 1166 mLock.lock(); 1167 while(!mConfigEvents.isEmpty()) { 1168 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1169 ConfigEvent configEvent = mConfigEvents[0]; 1170 mConfigEvents.removeAt(0); 1171 // release mLock before locking AudioFlinger mLock: lock order is always 1172 // AudioFlinger then ThreadBase to avoid cross deadlock 1173 mLock.unlock(); 1174 mAudioFlinger->mLock.lock(); 1175 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1176 mAudioFlinger->mLock.unlock(); 1177 mLock.lock(); 1178 } 1179 mLock.unlock(); 1180} 1181 1182status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1183{ 1184 const size_t SIZE = 256; 1185 char buffer[SIZE]; 1186 String8 result; 1187 1188 bool locked = tryLock(mLock); 1189 if (!locked) { 1190 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1191 write(fd, buffer, strlen(buffer)); 1192 } 1193 1194 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1197 result.append(buffer); 1198 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1199 result.append(buffer); 1200 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1201 result.append(buffer); 1202 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1203 result.append(buffer); 1204 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1205 result.append(buffer); 1206 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1207 result.append(buffer); 1208 1209 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1210 result.append(buffer); 1211 result.append(" Index Command"); 1212 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1213 snprintf(buffer, SIZE, "\n %02d ", i); 1214 result.append(buffer); 1215 result.append(mNewParameters[i]); 1216 } 1217 1218 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1219 result.append(buffer); 1220 snprintf(buffer, SIZE, " Index event param\n"); 1221 result.append(buffer); 1222 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1223 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1224 result.append(buffer); 1225 } 1226 result.append("\n"); 1227 1228 write(fd, result.string(), result.size()); 1229 1230 if (locked) { 1231 mLock.unlock(); 1232 } 1233 return NO_ERROR; 1234} 1235 1236status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1237{ 1238 const size_t SIZE = 256; 1239 char buffer[SIZE]; 1240 String8 result; 1241 1242 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1243 write(fd, buffer, strlen(buffer)); 1244 1245 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1246 sp<EffectChain> chain = mEffectChains[i]; 1247 if (chain != 0) { 1248 chain->dump(fd, args); 1249 } 1250 } 1251 return NO_ERROR; 1252} 1253 1254void AudioFlinger::ThreadBase::acquireWakeLock() 1255{ 1256 Mutex::Autolock _l(mLock); 1257 acquireWakeLock_l(); 1258} 1259 1260void AudioFlinger::ThreadBase::acquireWakeLock_l() 1261{ 1262 if (mPowerManager == 0) { 1263 // use checkService() to avoid blocking if power service is not up yet 1264 sp<IBinder> binder = 1265 defaultServiceManager()->checkService(String16("power")); 1266 if (binder == 0) { 1267 ALOGW("Thread %s cannot connect to the power manager service", mName); 1268 } else { 1269 mPowerManager = interface_cast<IPowerManager>(binder); 1270 binder->linkToDeath(mDeathRecipient); 1271 } 1272 } 1273 if (mPowerManager != 0) { 1274 sp<IBinder> binder = new BBinder(); 1275 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1276 binder, 1277 String16(mName)); 1278 if (status == NO_ERROR) { 1279 mWakeLockToken = binder; 1280 } 1281 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1282 } 1283} 1284 1285void AudioFlinger::ThreadBase::releaseWakeLock() 1286{ 1287 Mutex::Autolock _l(mLock); 1288 releaseWakeLock_l(); 1289} 1290 1291void AudioFlinger::ThreadBase::releaseWakeLock_l() 1292{ 1293 if (mWakeLockToken != 0) { 1294 ALOGV("releaseWakeLock_l() %s", mName); 1295 if (mPowerManager != 0) { 1296 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1297 } 1298 mWakeLockToken.clear(); 1299 } 1300} 1301 1302void AudioFlinger::ThreadBase::clearPowerManager() 1303{ 1304 Mutex::Autolock _l(mLock); 1305 releaseWakeLock_l(); 1306 mPowerManager.clear(); 1307} 1308 1309void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1310{ 1311 sp<ThreadBase> thread = mThread.promote(); 1312 if (thread != 0) { 1313 thread->clearPowerManager(); 1314 } 1315 ALOGW("power manager service died !!!"); 1316} 1317 1318void AudioFlinger::ThreadBase::setEffectSuspended( 1319 const effect_uuid_t *type, bool suspend, int sessionId) 1320{ 1321 Mutex::Autolock _l(mLock); 1322 setEffectSuspended_l(type, suspend, sessionId); 1323} 1324 1325void AudioFlinger::ThreadBase::setEffectSuspended_l( 1326 const effect_uuid_t *type, bool suspend, int sessionId) 1327{ 1328 sp<EffectChain> chain = getEffectChain_l(sessionId); 1329 if (chain != 0) { 1330 if (type != NULL) { 1331 chain->setEffectSuspended_l(type, suspend); 1332 } else { 1333 chain->setEffectSuspendedAll_l(suspend); 1334 } 1335 } 1336 1337 updateSuspendedSessions_l(type, suspend, sessionId); 1338} 1339 1340void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1341{ 1342 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1343 if (index < 0) { 1344 return; 1345 } 1346 1347 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1348 mSuspendedSessions.editValueAt(index); 1349 1350 for (size_t i = 0; i < sessionEffects.size(); i++) { 1351 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1352 for (int j = 0; j < desc->mRefCount; j++) { 1353 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1354 chain->setEffectSuspendedAll_l(true); 1355 } else { 1356 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1357 desc->mType.timeLow); 1358 chain->setEffectSuspended_l(&desc->mType, true); 1359 } 1360 } 1361 } 1362} 1363 1364void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1365 bool suspend, 1366 int sessionId) 1367{ 1368 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1369 1370 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1371 1372 if (suspend) { 1373 if (index >= 0) { 1374 sessionEffects = mSuspendedSessions.editValueAt(index); 1375 } else { 1376 mSuspendedSessions.add(sessionId, sessionEffects); 1377 } 1378 } else { 1379 if (index < 0) { 1380 return; 1381 } 1382 sessionEffects = mSuspendedSessions.editValueAt(index); 1383 } 1384 1385 1386 int key = EffectChain::kKeyForSuspendAll; 1387 if (type != NULL) { 1388 key = type->timeLow; 1389 } 1390 index = sessionEffects.indexOfKey(key); 1391 1392 sp <SuspendedSessionDesc> desc; 1393 if (suspend) { 1394 if (index >= 0) { 1395 desc = sessionEffects.valueAt(index); 1396 } else { 1397 desc = new SuspendedSessionDesc(); 1398 if (type != NULL) { 1399 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1400 } 1401 sessionEffects.add(key, desc); 1402 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1403 } 1404 desc->mRefCount++; 1405 } else { 1406 if (index < 0) { 1407 return; 1408 } 1409 desc = sessionEffects.valueAt(index); 1410 if (--desc->mRefCount == 0) { 1411 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1412 sessionEffects.removeItemsAt(index); 1413 if (sessionEffects.isEmpty()) { 1414 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1415 sessionId); 1416 mSuspendedSessions.removeItem(sessionId); 1417 } 1418 } 1419 } 1420 if (!sessionEffects.isEmpty()) { 1421 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1422 } 1423} 1424 1425void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1426 bool enabled, 1427 int sessionId) 1428{ 1429 Mutex::Autolock _l(mLock); 1430 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1431} 1432 1433void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1434 bool enabled, 1435 int sessionId) 1436{ 1437 if (mType != RECORD) { 1438 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1439 // another session. This gives the priority to well behaved effect control panels 1440 // and applications not using global effects. 1441 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1442 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1443 } 1444 } 1445 1446 sp<EffectChain> chain = getEffectChain_l(sessionId); 1447 if (chain != 0) { 1448 chain->checkSuspendOnEffectEnabled(effect, enabled); 1449 } 1450} 1451 1452// ---------------------------------------------------------------------------- 1453 1454AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1455 AudioStreamOut* output, 1456 audio_io_handle_t id, 1457 uint32_t device, 1458 type_t type) 1459 : ThreadBase(audioFlinger, id, device, type), 1460 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1461 // Assumes constructor is called by AudioFlinger with it's mLock held, 1462 // but it would be safer to explicitly pass initial masterMute as parameter 1463 mMasterMute(audioFlinger->masterMute_l()), 1464 // mStreamTypes[] initialized in constructor body 1465 mOutput(output), 1466 // Assumes constructor is called by AudioFlinger with it's mLock held, 1467 // but it would be safer to explicitly pass initial masterVolume as parameter 1468 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1469 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1470{ 1471 snprintf(mName, kNameLength, "AudioOut_%d", id); 1472 1473 readOutputParameters(); 1474 1475 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1476 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1477 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1478 stream = (audio_stream_type_t) (stream + 1)) { 1479 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1480 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1481 // initialized by stream_type_t default constructor 1482 // mStreamTypes[stream].valid = true; 1483 } 1484 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1485 // because mAudioFlinger doesn't have one to copy from 1486} 1487 1488AudioFlinger::PlaybackThread::~PlaybackThread() 1489{ 1490 delete [] mMixBuffer; 1491} 1492 1493status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1494{ 1495 dumpInternals(fd, args); 1496 dumpTracks(fd, args); 1497 dumpEffectChains(fd, args); 1498 return NO_ERROR; 1499} 1500 1501status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1502{ 1503 const size_t SIZE = 256; 1504 char buffer[SIZE]; 1505 String8 result; 1506 1507 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1508 result.append(buffer); 1509 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1510 for (size_t i = 0; i < mTracks.size(); ++i) { 1511 sp<Track> track = mTracks[i]; 1512 if (track != 0) { 1513 track->dump(buffer, SIZE); 1514 result.append(buffer); 1515 } 1516 } 1517 1518 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1519 result.append(buffer); 1520 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1521 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1522 sp<Track> track = mActiveTracks[i].promote(); 1523 if (track != 0) { 1524 track->dump(buffer, SIZE); 1525 result.append(buffer); 1526 } 1527 } 1528 write(fd, result.string(), result.size()); 1529 return NO_ERROR; 1530} 1531 1532status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1533{ 1534 const size_t SIZE = 256; 1535 char buffer[SIZE]; 1536 String8 result; 1537 1538 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1541 result.append(buffer); 1542 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1545 result.append(buffer); 1546 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1547 result.append(buffer); 1548 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1549 result.append(buffer); 1550 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1551 result.append(buffer); 1552 write(fd, result.string(), result.size()); 1553 1554 dumpBase(fd, args); 1555 1556 return NO_ERROR; 1557} 1558 1559// Thread virtuals 1560status_t AudioFlinger::PlaybackThread::readyToRun() 1561{ 1562 status_t status = initCheck(); 1563 if (status == NO_ERROR) { 1564 ALOGI("AudioFlinger's thread %p ready to run", this); 1565 } else { 1566 ALOGE("No working audio driver found."); 1567 } 1568 return status; 1569} 1570 1571void AudioFlinger::PlaybackThread::onFirstRef() 1572{ 1573 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1574} 1575 1576// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1577sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1578 const sp<AudioFlinger::Client>& client, 1579 audio_stream_type_t streamType, 1580 uint32_t sampleRate, 1581 audio_format_t format, 1582 uint32_t channelMask, 1583 int frameCount, 1584 const sp<IMemory>& sharedBuffer, 1585 int sessionId, 1586 bool isTimed, 1587 status_t *status) 1588{ 1589 sp<Track> track; 1590 status_t lStatus; 1591 1592 if (mType == DIRECT) { 1593 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1594 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1595 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1596 "for output %p with format %d", 1597 sampleRate, format, channelMask, mOutput, mFormat); 1598 lStatus = BAD_VALUE; 1599 goto Exit; 1600 } 1601 } 1602 } else { 1603 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1604 if (sampleRate > mSampleRate*2) { 1605 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1606 lStatus = BAD_VALUE; 1607 goto Exit; 1608 } 1609 } 1610 1611 lStatus = initCheck(); 1612 if (lStatus != NO_ERROR) { 1613 ALOGE("Audio driver not initialized."); 1614 goto Exit; 1615 } 1616 1617 { // scope for mLock 1618 Mutex::Autolock _l(mLock); 1619 1620 // all tracks in same audio session must share the same routing strategy otherwise 1621 // conflicts will happen when tracks are moved from one output to another by audio policy 1622 // manager 1623 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1624 for (size_t i = 0; i < mTracks.size(); ++i) { 1625 sp<Track> t = mTracks[i]; 1626 if (t != 0) { 1627 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1628 if (sessionId == t->sessionId() && strategy != actual) { 1629 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1630 strategy, actual); 1631 lStatus = BAD_VALUE; 1632 goto Exit; 1633 } 1634 } 1635 } 1636 1637 if (!isTimed) { 1638 track = new Track(this, client, streamType, sampleRate, format, 1639 channelMask, frameCount, sharedBuffer, sessionId); 1640 } else { 1641 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1642 channelMask, frameCount, sharedBuffer, sessionId); 1643 } 1644 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1645 lStatus = NO_MEMORY; 1646 goto Exit; 1647 } 1648 mTracks.add(track); 1649 1650 sp<EffectChain> chain = getEffectChain_l(sessionId); 1651 if (chain != 0) { 1652 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1653 track->setMainBuffer(chain->inBuffer()); 1654 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1655 chain->incTrackCnt(); 1656 } 1657 1658 // invalidate track immediately if the stream type was moved to another thread since 1659 // createTrack() was called by the client process. 1660 if (!mStreamTypes[streamType].valid) { 1661 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1662 this, streamType); 1663 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1664 } 1665 } 1666 lStatus = NO_ERROR; 1667 1668Exit: 1669 if(status) { 1670 *status = lStatus; 1671 } 1672 return track; 1673} 1674 1675uint32_t AudioFlinger::PlaybackThread::latency() const 1676{ 1677 Mutex::Autolock _l(mLock); 1678 if (initCheck() == NO_ERROR) { 1679 return mOutput->stream->get_latency(mOutput->stream); 1680 } else { 1681 return 0; 1682 } 1683} 1684 1685void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1686{ 1687 Mutex::Autolock _l(mLock); 1688 mMasterVolume = value; 1689} 1690 1691void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1692{ 1693 Mutex::Autolock _l(mLock); 1694 setMasterMute_l(muted); 1695} 1696 1697void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1698{ 1699 Mutex::Autolock _l(mLock); 1700 mStreamTypes[stream].volume = value; 1701} 1702 1703void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1704{ 1705 Mutex::Autolock _l(mLock); 1706 mStreamTypes[stream].mute = muted; 1707} 1708 1709float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1710{ 1711 Mutex::Autolock _l(mLock); 1712 return mStreamTypes[stream].volume; 1713} 1714 1715// addTrack_l() must be called with ThreadBase::mLock held 1716status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1717{ 1718 status_t status = ALREADY_EXISTS; 1719 1720 // set retry count for buffer fill 1721 track->mRetryCount = kMaxTrackStartupRetries; 1722 if (mActiveTracks.indexOf(track) < 0) { 1723 // the track is newly added, make sure it fills up all its 1724 // buffers before playing. This is to ensure the client will 1725 // effectively get the latency it requested. 1726 track->mFillingUpStatus = Track::FS_FILLING; 1727 track->mResetDone = false; 1728 mActiveTracks.add(track); 1729 if (track->mainBuffer() != mMixBuffer) { 1730 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1731 if (chain != 0) { 1732 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1733 chain->incActiveTrackCnt(); 1734 } 1735 } 1736 1737 status = NO_ERROR; 1738 } 1739 1740 ALOGV("mWaitWorkCV.broadcast"); 1741 mWaitWorkCV.broadcast(); 1742 1743 return status; 1744} 1745 1746// destroyTrack_l() must be called with ThreadBase::mLock held 1747void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1748{ 1749 track->mState = TrackBase::TERMINATED; 1750 if (mActiveTracks.indexOf(track) < 0) { 1751 removeTrack_l(track); 1752 } 1753} 1754 1755void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1756{ 1757 mTracks.remove(track); 1758 deleteTrackName_l(track->name()); 1759 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1760 if (chain != 0) { 1761 chain->decTrackCnt(); 1762 } 1763} 1764 1765String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1766{ 1767 String8 out_s8 = String8(""); 1768 char *s; 1769 1770 Mutex::Autolock _l(mLock); 1771 if (initCheck() != NO_ERROR) { 1772 return out_s8; 1773 } 1774 1775 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1776 out_s8 = String8(s); 1777 free(s); 1778 return out_s8; 1779} 1780 1781// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1782void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1783 AudioSystem::OutputDescriptor desc; 1784 void *param2 = NULL; 1785 1786 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1787 1788 switch (event) { 1789 case AudioSystem::OUTPUT_OPENED: 1790 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1791 desc.channels = mChannelMask; 1792 desc.samplingRate = mSampleRate; 1793 desc.format = mFormat; 1794 desc.frameCount = mFrameCount; 1795 desc.latency = latency(); 1796 param2 = &desc; 1797 break; 1798 1799 case AudioSystem::STREAM_CONFIG_CHANGED: 1800 param2 = ¶m; 1801 case AudioSystem::OUTPUT_CLOSED: 1802 default: 1803 break; 1804 } 1805 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1806} 1807 1808void AudioFlinger::PlaybackThread::readOutputParameters() 1809{ 1810 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1811 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1812 mChannelCount = (uint16_t)popcount(mChannelMask); 1813 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1814 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1815 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1816 1817 // FIXME - Current mixer implementation only supports stereo output: Always 1818 // Allocate a stereo buffer even if HW output is mono. 1819 delete[] mMixBuffer; 1820 mMixBuffer = new int16_t[mFrameCount * 2]; 1821 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1822 1823 // force reconfiguration of effect chains and engines to take new buffer size and audio 1824 // parameters into account 1825 // Note that mLock is not held when readOutputParameters() is called from the constructor 1826 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1827 // matter. 1828 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1829 Vector< sp<EffectChain> > effectChains = mEffectChains; 1830 for (size_t i = 0; i < effectChains.size(); i ++) { 1831 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1832 } 1833} 1834 1835status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1836{ 1837 if (halFrames == NULL || dspFrames == NULL) { 1838 return BAD_VALUE; 1839 } 1840 Mutex::Autolock _l(mLock); 1841 if (initCheck() != NO_ERROR) { 1842 return INVALID_OPERATION; 1843 } 1844 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1845 1846 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1847} 1848 1849uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1850{ 1851 Mutex::Autolock _l(mLock); 1852 uint32_t result = 0; 1853 if (getEffectChain_l(sessionId) != 0) { 1854 result = EFFECT_SESSION; 1855 } 1856 1857 for (size_t i = 0; i < mTracks.size(); ++i) { 1858 sp<Track> track = mTracks[i]; 1859 if (sessionId == track->sessionId() && 1860 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1861 result |= TRACK_SESSION; 1862 break; 1863 } 1864 } 1865 1866 return result; 1867} 1868 1869uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1870{ 1871 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1872 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1873 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1874 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1875 } 1876 for (size_t i = 0; i < mTracks.size(); i++) { 1877 sp<Track> track = mTracks[i]; 1878 if (sessionId == track->sessionId() && 1879 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1880 return AudioSystem::getStrategyForStream(track->streamType()); 1881 } 1882 } 1883 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1884} 1885 1886 1887AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1888{ 1889 Mutex::Autolock _l(mLock); 1890 return mOutput; 1891} 1892 1893AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1894{ 1895 Mutex::Autolock _l(mLock); 1896 AudioStreamOut *output = mOutput; 1897 mOutput = NULL; 1898 return output; 1899} 1900 1901// this method must always be called either with ThreadBase mLock held or inside the thread loop 1902audio_stream_t* AudioFlinger::PlaybackThread::stream() 1903{ 1904 if (mOutput == NULL) { 1905 return NULL; 1906 } 1907 return &mOutput->stream->common; 1908} 1909 1910uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1911{ 1912 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1913 // decoding and transfer time. So sleeping for half of the latency would likely cause 1914 // underruns 1915 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1916 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1917 } else { 1918 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1919 } 1920} 1921 1922// ---------------------------------------------------------------------------- 1923 1924AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1925 audio_io_handle_t id, uint32_t device, type_t type) 1926 : PlaybackThread(audioFlinger, output, id, device, type), 1927 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1928 mPrevMixerStatus(MIXER_IDLE) 1929{ 1930 // FIXME - Current mixer implementation only supports stereo output 1931 if (mChannelCount == 1) { 1932 ALOGE("Invalid audio hardware channel count"); 1933 } 1934} 1935 1936AudioFlinger::MixerThread::~MixerThread() 1937{ 1938 delete mAudioMixer; 1939} 1940 1941bool AudioFlinger::MixerThread::threadLoop() 1942{ 1943 Vector< sp<Track> > tracksToRemove; 1944 mixer_state mixerStatus = MIXER_IDLE; 1945 nsecs_t standbyTime = systemTime(); 1946 size_t mixBufferSize = mFrameCount * mFrameSize; 1947 // FIXME: Relaxed timing because of a certain device that can't meet latency 1948 // Should be reduced to 2x after the vendor fixes the driver issue 1949 // increase threshold again due to low power audio mode. The way this warning threshold is 1950 // calculated and its usefulness should be reconsidered anyway. 1951 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1952 nsecs_t lastWarning = 0; 1953 bool longStandbyExit = false; 1954 uint32_t activeSleepTime = activeSleepTimeUs(); 1955 uint32_t idleSleepTime = idleSleepTimeUs(); 1956 uint32_t sleepTime = idleSleepTime; 1957 uint32_t sleepTimeShift = 0; 1958 Vector< sp<EffectChain> > effectChains; 1959#ifdef DEBUG_CPU_USAGE 1960 ThreadCpuUsage cpu; 1961 const CentralTendencyStatistics& stats = cpu.statistics(); 1962#endif 1963 1964 acquireWakeLock(); 1965 1966 while (!exitPending()) 1967 { 1968#ifdef DEBUG_CPU_USAGE 1969 cpu.sampleAndEnable(); 1970 unsigned n = stats.n(); 1971 // cpu.elapsed() is expensive, so don't call it every loop 1972 if ((n & 127) == 1) { 1973 long long elapsed = cpu.elapsed(); 1974 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1975 double perLoop = elapsed / (double) n; 1976 double perLoop100 = perLoop * 0.01; 1977 double mean = stats.mean(); 1978 double stddev = stats.stddev(); 1979 double minimum = stats.minimum(); 1980 double maximum = stats.maximum(); 1981 cpu.resetStatistics(); 1982 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", 1983 elapsed * .000000001, n, perLoop * .000001, 1984 mean * .001, 1985 stddev * .001, 1986 minimum * .001, 1987 maximum * .001, 1988 mean / perLoop100, 1989 stddev / perLoop100, 1990 minimum / perLoop100, 1991 maximum / perLoop100); 1992 } 1993 } 1994#endif 1995 processConfigEvents(); 1996 1997 mixerStatus = MIXER_IDLE; 1998 { // scope for mLock 1999 2000 Mutex::Autolock _l(mLock); 2001 2002 if (checkForNewParameters_l()) { 2003 mixBufferSize = mFrameCount * mFrameSize; 2004 // FIXME: Relaxed timing because of a certain device that can't meet latency 2005 // Should be reduced to 2x after the vendor fixes the driver issue 2006 // increase threshold again due to low power audio mode. The way this warning 2007 // threshold is calculated and its usefulness should be reconsidered anyway. 2008 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2009 activeSleepTime = activeSleepTimeUs(); 2010 idleSleepTime = idleSleepTimeUs(); 2011 } 2012 2013 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2014 2015 // put audio hardware into standby after short delay 2016 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2017 mSuspended)) { 2018 if (!mStandby) { 2019 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2020 mOutput->stream->common.standby(&mOutput->stream->common); 2021 mStandby = true; 2022 mBytesWritten = 0; 2023 } 2024 2025 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2026 // we're about to wait, flush the binder command buffer 2027 IPCThreadState::self()->flushCommands(); 2028 2029 if (exitPending()) break; 2030 2031 releaseWakeLock_l(); 2032 // wait until we have something to do... 2033 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 2034 mWaitWorkCV.wait(mLock); 2035 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 2036 acquireWakeLock_l(); 2037 2038 mPrevMixerStatus = MIXER_IDLE; 2039 if (!mMasterMute) { 2040 char value[PROPERTY_VALUE_MAX]; 2041 property_get("ro.audio.silent", value, "0"); 2042 if (atoi(value)) { 2043 ALOGD("Silence is golden"); 2044 setMasterMute_l(true); 2045 } 2046 } 2047 2048 standbyTime = systemTime() + mStandbyTimeInNsecs; 2049 sleepTime = idleSleepTime; 2050 sleepTimeShift = 0; 2051 continue; 2052 } 2053 } 2054 2055 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2056 2057 // prevent any changes in effect chain list and in each effect chain 2058 // during mixing and effect process as the audio buffers could be deleted 2059 // or modified if an effect is created or deleted 2060 lockEffectChains_l(effectChains); 2061 } 2062 2063 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2064 // obtain the presentation timestamp of the next output buffer 2065 int64_t pts; 2066 status_t status = INVALID_OPERATION; 2067 2068 if (NULL != mOutput->stream->get_next_write_timestamp) { 2069 status = mOutput->stream->get_next_write_timestamp( 2070 mOutput->stream, &pts); 2071 } 2072 2073 if (status != NO_ERROR) { 2074 pts = AudioBufferProvider::kInvalidPTS; 2075 } 2076 2077 // mix buffers... 2078 mAudioMixer->process(pts); 2079 // increase sleep time progressively when application underrun condition clears. 2080 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2081 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2082 // such that we would underrun the audio HAL. 2083 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2084 sleepTimeShift--; 2085 } 2086 sleepTime = 0; 2087 standbyTime = systemTime() + mStandbyTimeInNsecs; 2088 //TODO: delay standby when effects have a tail 2089 } else { 2090 // If no tracks are ready, sleep once for the duration of an output 2091 // buffer size, then write 0s to the output 2092 if (sleepTime == 0) { 2093 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2094 sleepTime = activeSleepTime >> sleepTimeShift; 2095 if (sleepTime < kMinThreadSleepTimeUs) { 2096 sleepTime = kMinThreadSleepTimeUs; 2097 } 2098 // reduce sleep time in case of consecutive application underruns to avoid 2099 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2100 // duration we would end up writing less data than needed by the audio HAL if 2101 // the condition persists. 2102 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2103 sleepTimeShift++; 2104 } 2105 } else { 2106 sleepTime = idleSleepTime; 2107 } 2108 } else if (mBytesWritten != 0 || 2109 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2110 memset (mMixBuffer, 0, mixBufferSize); 2111 sleepTime = 0; 2112 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2113 } 2114 // TODO add standby time extension fct of effect tail 2115 } 2116 2117 if (mSuspended) { 2118 sleepTime = suspendSleepTimeUs(); 2119 } 2120 // sleepTime == 0 means we must write to audio hardware 2121 if (sleepTime == 0) { 2122 for (size_t i = 0; i < effectChains.size(); i ++) { 2123 effectChains[i]->process_l(); 2124 } 2125 // enable changes in effect chain 2126 unlockEffectChains(effectChains); 2127 mLastWriteTime = systemTime(); 2128 mInWrite = true; 2129 mBytesWritten += mixBufferSize; 2130 2131 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2132 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2133 mNumWrites++; 2134 mInWrite = false; 2135 nsecs_t now = systemTime(); 2136 nsecs_t delta = now - mLastWriteTime; 2137 if (!mStandby && delta > maxPeriod) { 2138 mNumDelayedWrites++; 2139 if ((now - lastWarning) > kWarningThrottleNs) { 2140 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2141 ns2ms(delta), mNumDelayedWrites, this); 2142 lastWarning = now; 2143 } 2144 if (mStandby) { 2145 longStandbyExit = true; 2146 } 2147 } 2148 mStandby = false; 2149 } else { 2150 // enable changes in effect chain 2151 unlockEffectChains(effectChains); 2152 usleep(sleepTime); 2153 } 2154 2155 // finally let go of all our tracks, without the lock held 2156 // since we can't guarantee the destructors won't acquire that 2157 // same lock. 2158 tracksToRemove.clear(); 2159 2160 // Effect chains will be actually deleted here if they were removed from 2161 // mEffectChains list during mixing or effects processing 2162 effectChains.clear(); 2163 } 2164 2165 if (!mStandby) { 2166 mOutput->stream->common.standby(&mOutput->stream->common); 2167 } 2168 2169 releaseWakeLock(); 2170 2171 ALOGV("MixerThread %p exiting", this); 2172 return false; 2173} 2174 2175// prepareTracks_l() must be called with ThreadBase::mLock held 2176AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2177 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2178{ 2179 2180 mixer_state mixerStatus = MIXER_IDLE; 2181 // find out which tracks need to be processed 2182 size_t count = activeTracks.size(); 2183 size_t mixedTracks = 0; 2184 size_t tracksWithEffect = 0; 2185 2186 float masterVolume = mMasterVolume; 2187 bool masterMute = mMasterMute; 2188 2189 if (masterMute) { 2190 masterVolume = 0; 2191 } 2192 // Delegate master volume control to effect in output mix effect chain if needed 2193 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2194 if (chain != 0) { 2195 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2196 chain->setVolume_l(&v, &v); 2197 masterVolume = (float)((v + (1 << 23)) >> 24); 2198 chain.clear(); 2199 } 2200 2201 for (size_t i=0 ; i<count ; i++) { 2202 sp<Track> t = activeTracks[i].promote(); 2203 if (t == 0) continue; 2204 2205 // this const just means the local variable doesn't change 2206 Track* const track = t.get(); 2207 audio_track_cblk_t* cblk = track->cblk(); 2208 2209 // The first time a track is added we wait 2210 // for all its buffers to be filled before processing it 2211 int name = track->name(); 2212 // make sure that we have enough frames to mix one full buffer. 2213 // enforce this condition only once to enable draining the buffer in case the client 2214 // app does not call stop() and relies on underrun to stop: 2215 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2216 // during last round 2217 uint32_t minFrames = 1; 2218 if (!track->isStopped() && !track->isPausing() && 2219 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2220 if (t->sampleRate() == (int)mSampleRate) { 2221 minFrames = mFrameCount; 2222 } else { 2223 // +1 for rounding and +1 for additional sample needed for interpolation 2224 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2225 // add frames already consumed but not yet released by the resampler 2226 // because cblk->framesReady() will include these frames 2227 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2228 // the minimum track buffer size is normally twice the number of frames necessary 2229 // to fill one buffer and the resampler should not leave more than one buffer worth 2230 // of unreleased frames after each pass, but just in case... 2231 ALOG_ASSERT(minFrames <= cblk->frameCount); 2232 } 2233 } 2234 if ((track->framesReady() >= minFrames) && track->isReady() && 2235 !track->isPaused() && !track->isTerminated()) 2236 { 2237 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2238 2239 mixedTracks++; 2240 2241 // track->mainBuffer() != mMixBuffer means there is an effect chain 2242 // connected to the track 2243 chain.clear(); 2244 if (track->mainBuffer() != mMixBuffer) { 2245 chain = getEffectChain_l(track->sessionId()); 2246 // Delegate volume control to effect in track effect chain if needed 2247 if (chain != 0) { 2248 tracksWithEffect++; 2249 } else { 2250 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2251 name, track->sessionId()); 2252 } 2253 } 2254 2255 2256 int param = AudioMixer::VOLUME; 2257 if (track->mFillingUpStatus == Track::FS_FILLED) { 2258 // no ramp for the first volume setting 2259 track->mFillingUpStatus = Track::FS_ACTIVE; 2260 if (track->mState == TrackBase::RESUMING) { 2261 track->mState = TrackBase::ACTIVE; 2262 param = AudioMixer::RAMP_VOLUME; 2263 } 2264 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2265 } else if (cblk->server != 0) { 2266 // If the track is stopped before the first frame was mixed, 2267 // do not apply ramp 2268 param = AudioMixer::RAMP_VOLUME; 2269 } 2270 2271 // compute volume for this track 2272 uint32_t vl, vr, va; 2273 if (track->isMuted() || track->isPausing() || 2274 mStreamTypes[track->streamType()].mute) { 2275 vl = vr = va = 0; 2276 if (track->isPausing()) { 2277 track->setPaused(); 2278 } 2279 } else { 2280 2281 // read original volumes with volume control 2282 float typeVolume = mStreamTypes[track->streamType()].volume; 2283 float v = masterVolume * typeVolume; 2284 uint32_t vlr = cblk->getVolumeLR(); 2285 vl = vlr & 0xFFFF; 2286 vr = vlr >> 16; 2287 // track volumes come from shared memory, so can't be trusted and must be clamped 2288 if (vl > MAX_GAIN_INT) { 2289 ALOGV("Track left volume out of range: %04X", vl); 2290 vl = MAX_GAIN_INT; 2291 } 2292 if (vr > MAX_GAIN_INT) { 2293 ALOGV("Track right volume out of range: %04X", vr); 2294 vr = MAX_GAIN_INT; 2295 } 2296 // now apply the master volume and stream type volume 2297 vl = (uint32_t)(v * vl) << 12; 2298 vr = (uint32_t)(v * vr) << 12; 2299 // assuming master volume and stream type volume each go up to 1.0, 2300 // vl and vr are now in 8.24 format 2301 2302 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2303 // send level comes from shared memory and so may be corrupt 2304 if (sendLevel > MAX_GAIN_INT) { 2305 ALOGV("Track send level out of range: %04X", sendLevel); 2306 sendLevel = MAX_GAIN_INT; 2307 } 2308 va = (uint32_t)(v * sendLevel); 2309 } 2310 // Delegate volume control to effect in track effect chain if needed 2311 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2312 // Do not ramp volume if volume is controlled by effect 2313 param = AudioMixer::VOLUME; 2314 track->mHasVolumeController = true; 2315 } else { 2316 // force no volume ramp when volume controller was just disabled or removed 2317 // from effect chain to avoid volume spike 2318 if (track->mHasVolumeController) { 2319 param = AudioMixer::VOLUME; 2320 } 2321 track->mHasVolumeController = false; 2322 } 2323 2324 // Convert volumes from 8.24 to 4.12 format 2325 // This additional clamping is needed in case chain->setVolume_l() overshot 2326 vl = (vl + (1 << 11)) >> 12; 2327 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2328 vr = (vr + (1 << 11)) >> 12; 2329 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2330 2331 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2332 2333 // XXX: these things DON'T need to be done each time 2334 mAudioMixer->setBufferProvider(name, track); 2335 mAudioMixer->enable(name); 2336 2337 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2338 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2339 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2340 mAudioMixer->setParameter( 2341 name, 2342 AudioMixer::TRACK, 2343 AudioMixer::FORMAT, (void *)track->format()); 2344 mAudioMixer->setParameter( 2345 name, 2346 AudioMixer::TRACK, 2347 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2348 mAudioMixer->setParameter( 2349 name, 2350 AudioMixer::RESAMPLE, 2351 AudioMixer::SAMPLE_RATE, 2352 (void *)(cblk->sampleRate)); 2353 mAudioMixer->setParameter( 2354 name, 2355 AudioMixer::TRACK, 2356 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2357 mAudioMixer->setParameter( 2358 name, 2359 AudioMixer::TRACK, 2360 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2361 2362 // reset retry count 2363 track->mRetryCount = kMaxTrackRetries; 2364 // If one track is ready, set the mixer ready if: 2365 // - the mixer was not ready during previous round OR 2366 // - no other track is not ready 2367 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2368 mixerStatus != MIXER_TRACKS_ENABLED) { 2369 mixerStatus = MIXER_TRACKS_READY; 2370 } 2371 } else { 2372 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2373 if (track->isStopped()) { 2374 track->reset(); 2375 } 2376 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2377 // We have consumed all the buffers of this track. 2378 // Remove it from the list of active tracks. 2379 tracksToRemove->add(track); 2380 } else { 2381 // No buffers for this track. Give it a few chances to 2382 // fill a buffer, then remove it from active list. 2383 if (--(track->mRetryCount) <= 0) { 2384 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2385 tracksToRemove->add(track); 2386 // indicate to client process that the track was disabled because of underrun 2387 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2388 // If one track is not ready, mark the mixer also not ready if: 2389 // - the mixer was ready during previous round OR 2390 // - no other track is ready 2391 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2392 mixerStatus != MIXER_TRACKS_READY) { 2393 mixerStatus = MIXER_TRACKS_ENABLED; 2394 } 2395 } 2396 mAudioMixer->disable(name); 2397 } 2398 } 2399 2400 // remove all the tracks that need to be... 2401 count = tracksToRemove->size(); 2402 if (CC_UNLIKELY(count)) { 2403 for (size_t i=0 ; i<count ; i++) { 2404 const sp<Track>& track = tracksToRemove->itemAt(i); 2405 mActiveTracks.remove(track); 2406 if (track->mainBuffer() != mMixBuffer) { 2407 chain = getEffectChain_l(track->sessionId()); 2408 if (chain != 0) { 2409 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2410 chain->decActiveTrackCnt(); 2411 } 2412 } 2413 if (track->isTerminated()) { 2414 removeTrack_l(track); 2415 } 2416 } 2417 } 2418 2419 // mix buffer must be cleared if all tracks are connected to an 2420 // effect chain as in this case the mixer will not write to 2421 // mix buffer and track effects will accumulate into it 2422 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2423 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2424 } 2425 2426 mPrevMixerStatus = mixerStatus; 2427 return mixerStatus; 2428} 2429 2430void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2431{ 2432 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2433 this, streamType, mTracks.size()); 2434 Mutex::Autolock _l(mLock); 2435 2436 size_t size = mTracks.size(); 2437 for (size_t i = 0; i < size; i++) { 2438 sp<Track> t = mTracks[i]; 2439 if (t->streamType() == streamType) { 2440 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2441 t->mCblk->cv.signal(); 2442 } 2443 } 2444} 2445 2446void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2447{ 2448 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2449 this, streamType, valid); 2450 Mutex::Autolock _l(mLock); 2451 2452 mStreamTypes[streamType].valid = valid; 2453} 2454 2455// getTrackName_l() must be called with ThreadBase::mLock held 2456int AudioFlinger::MixerThread::getTrackName_l() 2457{ 2458 return mAudioMixer->getTrackName(); 2459} 2460 2461// deleteTrackName_l() must be called with ThreadBase::mLock held 2462void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2463{ 2464 ALOGV("remove track (%d) and delete from mixer", name); 2465 mAudioMixer->deleteTrackName(name); 2466} 2467 2468// checkForNewParameters_l() must be called with ThreadBase::mLock held 2469bool AudioFlinger::MixerThread::checkForNewParameters_l() 2470{ 2471 bool reconfig = false; 2472 2473 while (!mNewParameters.isEmpty()) { 2474 status_t status = NO_ERROR; 2475 String8 keyValuePair = mNewParameters[0]; 2476 AudioParameter param = AudioParameter(keyValuePair); 2477 int value; 2478 2479 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2480 reconfig = true; 2481 } 2482 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2483 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2484 status = BAD_VALUE; 2485 } else { 2486 reconfig = true; 2487 } 2488 } 2489 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2490 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2491 status = BAD_VALUE; 2492 } else { 2493 reconfig = true; 2494 } 2495 } 2496 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2497 // do not accept frame count changes if tracks are open as the track buffer 2498 // size depends on frame count and correct behavior would not be guaranteed 2499 // if frame count is changed after track creation 2500 if (!mTracks.isEmpty()) { 2501 status = INVALID_OPERATION; 2502 } else { 2503 reconfig = true; 2504 } 2505 } 2506 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2507 // when changing the audio output device, call addBatteryData to notify 2508 // the change 2509 if ((int)mDevice != value) { 2510 uint32_t params = 0; 2511 // check whether speaker is on 2512 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2513 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2514 } 2515 2516 int deviceWithoutSpeaker 2517 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2518 // check if any other device (except speaker) is on 2519 if (value & deviceWithoutSpeaker ) { 2520 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2521 } 2522 2523 if (params != 0) { 2524 addBatteryData(params); 2525 } 2526 } 2527 2528 // forward device change to effects that have requested to be 2529 // aware of attached audio device. 2530 mDevice = (uint32_t)value; 2531 for (size_t i = 0; i < mEffectChains.size(); i++) { 2532 mEffectChains[i]->setDevice_l(mDevice); 2533 } 2534 } 2535 2536 if (status == NO_ERROR) { 2537 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2538 keyValuePair.string()); 2539 if (!mStandby && status == INVALID_OPERATION) { 2540 mOutput->stream->common.standby(&mOutput->stream->common); 2541 mStandby = true; 2542 mBytesWritten = 0; 2543 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2544 keyValuePair.string()); 2545 } 2546 if (status == NO_ERROR && reconfig) { 2547 delete mAudioMixer; 2548 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2549 mAudioMixer = NULL; 2550 readOutputParameters(); 2551 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2552 for (size_t i = 0; i < mTracks.size() ; i++) { 2553 int name = getTrackName_l(); 2554 if (name < 0) break; 2555 mTracks[i]->mName = name; 2556 // limit track sample rate to 2 x new output sample rate 2557 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2558 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2559 } 2560 } 2561 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2562 } 2563 } 2564 2565 mNewParameters.removeAt(0); 2566 2567 mParamStatus = status; 2568 mParamCond.signal(); 2569 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2570 // already timed out waiting for the status and will never signal the condition. 2571 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2572 } 2573 return reconfig; 2574} 2575 2576status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2577{ 2578 const size_t SIZE = 256; 2579 char buffer[SIZE]; 2580 String8 result; 2581 2582 PlaybackThread::dumpInternals(fd, args); 2583 2584 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2585 result.append(buffer); 2586 write(fd, result.string(), result.size()); 2587 return NO_ERROR; 2588} 2589 2590uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2591{ 2592 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2593} 2594 2595uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2596{ 2597 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2598} 2599 2600// ---------------------------------------------------------------------------- 2601AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2602 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2603 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2604 // mLeftVolFloat, mRightVolFloat 2605 // mLeftVolShort, mRightVolShort 2606{ 2607} 2608 2609AudioFlinger::DirectOutputThread::~DirectOutputThread() 2610{ 2611} 2612 2613void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2614{ 2615 // Do not apply volume on compressed audio 2616 if (!audio_is_linear_pcm(mFormat)) { 2617 return; 2618 } 2619 2620 // convert to signed 16 bit before volume calculation 2621 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2622 size_t count = mFrameCount * mChannelCount; 2623 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2624 int16_t *dst = mMixBuffer + count-1; 2625 while(count--) { 2626 *dst-- = (int16_t)(*src--^0x80) << 8; 2627 } 2628 } 2629 2630 size_t frameCount = mFrameCount; 2631 int16_t *out = mMixBuffer; 2632 if (ramp) { 2633 if (mChannelCount == 1) { 2634 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2635 int32_t vlInc = d / (int32_t)frameCount; 2636 int32_t vl = ((int32_t)mLeftVolShort << 16); 2637 do { 2638 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2639 out++; 2640 vl += vlInc; 2641 } while (--frameCount); 2642 2643 } else { 2644 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2645 int32_t vlInc = d / (int32_t)frameCount; 2646 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2647 int32_t vrInc = d / (int32_t)frameCount; 2648 int32_t vl = ((int32_t)mLeftVolShort << 16); 2649 int32_t vr = ((int32_t)mRightVolShort << 16); 2650 do { 2651 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2652 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2653 out += 2; 2654 vl += vlInc; 2655 vr += vrInc; 2656 } while (--frameCount); 2657 } 2658 } else { 2659 if (mChannelCount == 1) { 2660 do { 2661 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2662 out++; 2663 } while (--frameCount); 2664 } else { 2665 do { 2666 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2667 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2668 out += 2; 2669 } while (--frameCount); 2670 } 2671 } 2672 2673 // convert back to unsigned 8 bit after volume calculation 2674 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2675 size_t count = mFrameCount * mChannelCount; 2676 int16_t *src = mMixBuffer; 2677 uint8_t *dst = (uint8_t *)mMixBuffer; 2678 while(count--) { 2679 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2680 } 2681 } 2682 2683 mLeftVolShort = leftVol; 2684 mRightVolShort = rightVol; 2685} 2686 2687bool AudioFlinger::DirectOutputThread::threadLoop() 2688{ 2689 mixer_state mixerStatus = MIXER_IDLE; 2690 sp<Track> trackToRemove; 2691 sp<Track> activeTrack; 2692 nsecs_t standbyTime = systemTime(); 2693 size_t mixBufferSize = mFrameCount*mFrameSize; 2694 uint32_t activeSleepTime = activeSleepTimeUs(); 2695 uint32_t idleSleepTime = idleSleepTimeUs(); 2696 uint32_t sleepTime = idleSleepTime; 2697 // use shorter standby delay as on normal output to release 2698 // hardware resources as soon as possible 2699 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2700 2701 acquireWakeLock(); 2702 2703 while (!exitPending()) 2704 { 2705 bool rampVolume; 2706 uint16_t leftVol; 2707 uint16_t rightVol; 2708 Vector< sp<EffectChain> > effectChains; 2709 2710 processConfigEvents(); 2711 2712 mixerStatus = MIXER_IDLE; 2713 2714 { // scope for the mLock 2715 2716 Mutex::Autolock _l(mLock); 2717 2718 if (checkForNewParameters_l()) { 2719 mixBufferSize = mFrameCount*mFrameSize; 2720 activeSleepTime = activeSleepTimeUs(); 2721 idleSleepTime = idleSleepTimeUs(); 2722 standbyDelay = microseconds(activeSleepTime*2); 2723 } 2724 2725 // put audio hardware into standby after short delay 2726 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2727 mSuspended)) { 2728 // wait until we have something to do... 2729 if (!mStandby) { 2730 ALOGV("Audio hardware entering standby, mixer %p", this); 2731 mOutput->stream->common.standby(&mOutput->stream->common); 2732 mStandby = true; 2733 mBytesWritten = 0; 2734 } 2735 2736 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2737 // we're about to wait, flush the binder command buffer 2738 IPCThreadState::self()->flushCommands(); 2739 2740 if (exitPending()) break; 2741 2742 releaseWakeLock_l(); 2743 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2744 mWaitWorkCV.wait(mLock); 2745 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2746 acquireWakeLock_l(); 2747 2748 if (!mMasterMute) { 2749 char value[PROPERTY_VALUE_MAX]; 2750 property_get("ro.audio.silent", value, "0"); 2751 if (atoi(value)) { 2752 ALOGD("Silence is golden"); 2753 setMasterMute_l(true); 2754 } 2755 } 2756 2757 standbyTime = systemTime() + standbyDelay; 2758 sleepTime = idleSleepTime; 2759 continue; 2760 } 2761 } 2762 2763 effectChains = mEffectChains; 2764 2765 // find out which tracks need to be processed 2766 if (mActiveTracks.size() != 0) { 2767 sp<Track> t = mActiveTracks[0].promote(); 2768 if (t == 0) continue; 2769 2770 Track* const track = t.get(); 2771 audio_track_cblk_t* cblk = track->cblk(); 2772 2773 // The first time a track is added we wait 2774 // for all its buffers to be filled before processing it 2775 if (cblk->framesReady() && track->isReady() && 2776 !track->isPaused() && !track->isTerminated()) 2777 { 2778 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2779 2780 if (track->mFillingUpStatus == Track::FS_FILLED) { 2781 track->mFillingUpStatus = Track::FS_ACTIVE; 2782 mLeftVolFloat = mRightVolFloat = 0; 2783 mLeftVolShort = mRightVolShort = 0; 2784 if (track->mState == TrackBase::RESUMING) { 2785 track->mState = TrackBase::ACTIVE; 2786 rampVolume = true; 2787 } 2788 } else if (cblk->server != 0) { 2789 // If the track is stopped before the first frame was mixed, 2790 // do not apply ramp 2791 rampVolume = true; 2792 } 2793 // compute volume for this track 2794 float left, right; 2795 if (track->isMuted() || mMasterMute || track->isPausing() || 2796 mStreamTypes[track->streamType()].mute) { 2797 left = right = 0; 2798 if (track->isPausing()) { 2799 track->setPaused(); 2800 } 2801 } else { 2802 float typeVolume = mStreamTypes[track->streamType()].volume; 2803 float v = mMasterVolume * typeVolume; 2804 uint32_t vlr = cblk->getVolumeLR(); 2805 float v_clamped = v * (vlr & 0xFFFF); 2806 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2807 left = v_clamped/MAX_GAIN; 2808 v_clamped = v * (vlr >> 16); 2809 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2810 right = v_clamped/MAX_GAIN; 2811 } 2812 2813 if (left != mLeftVolFloat || right != mRightVolFloat) { 2814 mLeftVolFloat = left; 2815 mRightVolFloat = right; 2816 2817 // If audio HAL implements volume control, 2818 // force software volume to nominal value 2819 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2820 left = 1.0f; 2821 right = 1.0f; 2822 } 2823 2824 // Convert volumes from float to 8.24 2825 uint32_t vl = (uint32_t)(left * (1 << 24)); 2826 uint32_t vr = (uint32_t)(right * (1 << 24)); 2827 2828 // Delegate volume control to effect in track effect chain if needed 2829 // only one effect chain can be present on DirectOutputThread, so if 2830 // there is one, the track is connected to it 2831 if (!effectChains.isEmpty()) { 2832 // Do not ramp volume if volume is controlled by effect 2833 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2834 rampVolume = false; 2835 } 2836 } 2837 2838 // Convert volumes from 8.24 to 4.12 format 2839 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2840 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2841 leftVol = (uint16_t)v_clamped; 2842 v_clamped = (vr + (1 << 11)) >> 12; 2843 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2844 rightVol = (uint16_t)v_clamped; 2845 } else { 2846 leftVol = mLeftVolShort; 2847 rightVol = mRightVolShort; 2848 rampVolume = false; 2849 } 2850 2851 // reset retry count 2852 track->mRetryCount = kMaxTrackRetriesDirect; 2853 activeTrack = t; 2854 mixerStatus = MIXER_TRACKS_READY; 2855 } else { 2856 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2857 if (track->isStopped()) { 2858 track->reset(); 2859 } 2860 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2861 // We have consumed all the buffers of this track. 2862 // Remove it from the list of active tracks. 2863 trackToRemove = track; 2864 } else { 2865 // No buffers for this track. Give it a few chances to 2866 // fill a buffer, then remove it from active list. 2867 if (--(track->mRetryCount) <= 0) { 2868 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2869 trackToRemove = track; 2870 } else { 2871 mixerStatus = MIXER_TRACKS_ENABLED; 2872 } 2873 } 2874 } 2875 } 2876 2877 // remove all the tracks that need to be... 2878 if (CC_UNLIKELY(trackToRemove != 0)) { 2879 mActiveTracks.remove(trackToRemove); 2880 if (!effectChains.isEmpty()) { 2881 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2882 trackToRemove->sessionId()); 2883 effectChains[0]->decActiveTrackCnt(); 2884 } 2885 if (trackToRemove->isTerminated()) { 2886 removeTrack_l(trackToRemove); 2887 } 2888 } 2889 2890 lockEffectChains_l(effectChains); 2891 } 2892 2893 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2894 AudioBufferProvider::Buffer buffer; 2895 size_t frameCount = mFrameCount; 2896 int8_t *curBuf = (int8_t *)mMixBuffer; 2897 // output audio to hardware 2898 while (frameCount) { 2899 buffer.frameCount = frameCount; 2900 activeTrack->getNextBuffer(&buffer, 2901 AudioBufferProvider::kInvalidPTS); 2902 if (CC_UNLIKELY(buffer.raw == NULL)) { 2903 memset(curBuf, 0, frameCount * mFrameSize); 2904 break; 2905 } 2906 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2907 frameCount -= buffer.frameCount; 2908 curBuf += buffer.frameCount * mFrameSize; 2909 activeTrack->releaseBuffer(&buffer); 2910 } 2911 sleepTime = 0; 2912 standbyTime = systemTime() + standbyDelay; 2913 } else { 2914 if (sleepTime == 0) { 2915 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2916 sleepTime = activeSleepTime; 2917 } else { 2918 sleepTime = idleSleepTime; 2919 } 2920 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2921 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2922 sleepTime = 0; 2923 } 2924 } 2925 2926 if (mSuspended) { 2927 sleepTime = suspendSleepTimeUs(); 2928 } 2929 // sleepTime == 0 means we must write to audio hardware 2930 if (sleepTime == 0) { 2931 if (mixerStatus == MIXER_TRACKS_READY) { 2932 applyVolume(leftVol, rightVol, rampVolume); 2933 } 2934 for (size_t i = 0; i < effectChains.size(); i ++) { 2935 effectChains[i]->process_l(); 2936 } 2937 unlockEffectChains(effectChains); 2938 2939 mLastWriteTime = systemTime(); 2940 mInWrite = true; 2941 mBytesWritten += mixBufferSize; 2942 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2943 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2944 mNumWrites++; 2945 mInWrite = false; 2946 mStandby = false; 2947 } else { 2948 unlockEffectChains(effectChains); 2949 usleep(sleepTime); 2950 } 2951 2952 // finally let go of removed track, without the lock held 2953 // since we can't guarantee the destructors won't acquire that 2954 // same lock. 2955 trackToRemove.clear(); 2956 activeTrack.clear(); 2957 2958 // Effect chains will be actually deleted here if they were removed from 2959 // mEffectChains list during mixing or effects processing 2960 effectChains.clear(); 2961 } 2962 2963 if (!mStandby) { 2964 mOutput->stream->common.standby(&mOutput->stream->common); 2965 } 2966 2967 releaseWakeLock(); 2968 2969 ALOGV("DirectOutputThread %p exiting", this); 2970 return false; 2971} 2972 2973// getTrackName_l() must be called with ThreadBase::mLock held 2974int AudioFlinger::DirectOutputThread::getTrackName_l() 2975{ 2976 return 0; 2977} 2978 2979// deleteTrackName_l() must be called with ThreadBase::mLock held 2980void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2981{ 2982} 2983 2984// checkForNewParameters_l() must be called with ThreadBase::mLock held 2985bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2986{ 2987 bool reconfig = false; 2988 2989 while (!mNewParameters.isEmpty()) { 2990 status_t status = NO_ERROR; 2991 String8 keyValuePair = mNewParameters[0]; 2992 AudioParameter param = AudioParameter(keyValuePair); 2993 int value; 2994 2995 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2996 // do not accept frame count changes if tracks are open as the track buffer 2997 // size depends on frame count and correct behavior would not be garantied 2998 // if frame count is changed after track creation 2999 if (!mTracks.isEmpty()) { 3000 status = INVALID_OPERATION; 3001 } else { 3002 reconfig = true; 3003 } 3004 } 3005 if (status == NO_ERROR) { 3006 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3007 keyValuePair.string()); 3008 if (!mStandby && status == INVALID_OPERATION) { 3009 mOutput->stream->common.standby(&mOutput->stream->common); 3010 mStandby = true; 3011 mBytesWritten = 0; 3012 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3013 keyValuePair.string()); 3014 } 3015 if (status == NO_ERROR && reconfig) { 3016 readOutputParameters(); 3017 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3018 } 3019 } 3020 3021 mNewParameters.removeAt(0); 3022 3023 mParamStatus = status; 3024 mParamCond.signal(); 3025 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3026 // already timed out waiting for the status and will never signal the condition. 3027 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3028 } 3029 return reconfig; 3030} 3031 3032uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3033{ 3034 uint32_t time; 3035 if (audio_is_linear_pcm(mFormat)) { 3036 time = PlaybackThread::activeSleepTimeUs(); 3037 } else { 3038 time = 10000; 3039 } 3040 return time; 3041} 3042 3043uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3044{ 3045 uint32_t time; 3046 if (audio_is_linear_pcm(mFormat)) { 3047 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3048 } else { 3049 time = 10000; 3050 } 3051 return time; 3052} 3053 3054uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3055{ 3056 uint32_t time; 3057 if (audio_is_linear_pcm(mFormat)) { 3058 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3059 } else { 3060 time = 10000; 3061 } 3062 return time; 3063} 3064 3065 3066// ---------------------------------------------------------------------------- 3067 3068AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3069 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3070 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3071 mWaitTimeMs(UINT_MAX) 3072{ 3073 addOutputTrack(mainThread); 3074} 3075 3076AudioFlinger::DuplicatingThread::~DuplicatingThread() 3077{ 3078 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3079 mOutputTracks[i]->destroy(); 3080 } 3081} 3082 3083bool AudioFlinger::DuplicatingThread::threadLoop() 3084{ 3085 Vector< sp<Track> > tracksToRemove; 3086 mixer_state mixerStatus = MIXER_IDLE; 3087 nsecs_t standbyTime = systemTime(); 3088 size_t mixBufferSize = mFrameCount*mFrameSize; 3089 SortedVector< sp<OutputTrack> > outputTracks; 3090 uint32_t writeFrames = 0; 3091 uint32_t activeSleepTime = activeSleepTimeUs(); 3092 uint32_t idleSleepTime = idleSleepTimeUs(); 3093 uint32_t sleepTime = idleSleepTime; 3094 Vector< sp<EffectChain> > effectChains; 3095 3096 acquireWakeLock(); 3097 3098 while (!exitPending()) 3099 { 3100 processConfigEvents(); 3101 3102 mixerStatus = MIXER_IDLE; 3103 { // scope for the mLock 3104 3105 Mutex::Autolock _l(mLock); 3106 3107 if (checkForNewParameters_l()) { 3108 mixBufferSize = mFrameCount*mFrameSize; 3109 updateWaitTime(); 3110 activeSleepTime = activeSleepTimeUs(); 3111 idleSleepTime = idleSleepTimeUs(); 3112 } 3113 3114 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3115 3116 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3117 outputTracks.add(mOutputTracks[i]); 3118 } 3119 3120 // put audio hardware into standby after short delay 3121 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3122 mSuspended)) { 3123 if (!mStandby) { 3124 for (size_t i = 0; i < outputTracks.size(); i++) { 3125 outputTracks[i]->stop(); 3126 } 3127 mStandby = true; 3128 mBytesWritten = 0; 3129 } 3130 3131 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3132 // we're about to wait, flush the binder command buffer 3133 IPCThreadState::self()->flushCommands(); 3134 outputTracks.clear(); 3135 3136 if (exitPending()) break; 3137 3138 releaseWakeLock_l(); 3139 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3140 mWaitWorkCV.wait(mLock); 3141 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3142 acquireWakeLock_l(); 3143 3144 mPrevMixerStatus = MIXER_IDLE; 3145 if (!mMasterMute) { 3146 char value[PROPERTY_VALUE_MAX]; 3147 property_get("ro.audio.silent", value, "0"); 3148 if (atoi(value)) { 3149 ALOGD("Silence is golden"); 3150 setMasterMute_l(true); 3151 } 3152 } 3153 3154 standbyTime = systemTime() + mStandbyTimeInNsecs; 3155 sleepTime = idleSleepTime; 3156 continue; 3157 } 3158 } 3159 3160 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3161 3162 // prevent any changes in effect chain list and in each effect chain 3163 // during mixing and effect process as the audio buffers could be deleted 3164 // or modified if an effect is created or deleted 3165 lockEffectChains_l(effectChains); 3166 } 3167 3168 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3169 // mix buffers... 3170 if (outputsReady(outputTracks)) { 3171 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3172 } else { 3173 memset(mMixBuffer, 0, mixBufferSize); 3174 } 3175 sleepTime = 0; 3176 writeFrames = mFrameCount; 3177 } else { 3178 if (sleepTime == 0) { 3179 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3180 sleepTime = activeSleepTime; 3181 } else { 3182 sleepTime = idleSleepTime; 3183 } 3184 } else if (mBytesWritten != 0) { 3185 // flush remaining overflow buffers in output tracks 3186 for (size_t i = 0; i < outputTracks.size(); i++) { 3187 if (outputTracks[i]->isActive()) { 3188 sleepTime = 0; 3189 writeFrames = 0; 3190 memset(mMixBuffer, 0, mixBufferSize); 3191 break; 3192 } 3193 } 3194 } 3195 } 3196 3197 if (mSuspended) { 3198 sleepTime = suspendSleepTimeUs(); 3199 } 3200 // sleepTime == 0 means we must write to audio hardware 3201 if (sleepTime == 0) { 3202 for (size_t i = 0; i < effectChains.size(); i ++) { 3203 effectChains[i]->process_l(); 3204 } 3205 // enable changes in effect chain 3206 unlockEffectChains(effectChains); 3207 3208 standbyTime = systemTime() + mStandbyTimeInNsecs; 3209 for (size_t i = 0; i < outputTracks.size(); i++) { 3210 outputTracks[i]->write(mMixBuffer, writeFrames); 3211 } 3212 mStandby = false; 3213 mBytesWritten += mixBufferSize; 3214 } else { 3215 // enable changes in effect chain 3216 unlockEffectChains(effectChains); 3217 usleep(sleepTime); 3218 } 3219 3220 // finally let go of all our tracks, without the lock held 3221 // since we can't guarantee the destructors won't acquire that 3222 // same lock. 3223 tracksToRemove.clear(); 3224 outputTracks.clear(); 3225 3226 // Effect chains will be actually deleted here if they were removed from 3227 // mEffectChains list during mixing or effects processing 3228 effectChains.clear(); 3229 } 3230 3231 releaseWakeLock(); 3232 3233 return false; 3234} 3235 3236void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3237{ 3238 // FIXME explain this formula 3239 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3240 OutputTrack *outputTrack = new OutputTrack(thread, 3241 this, 3242 mSampleRate, 3243 mFormat, 3244 mChannelMask, 3245 frameCount); 3246 if (outputTrack->cblk() != NULL) { 3247 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3248 mOutputTracks.add(outputTrack); 3249 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3250 updateWaitTime(); 3251 } 3252} 3253 3254void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3255{ 3256 Mutex::Autolock _l(mLock); 3257 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3258 if (mOutputTracks[i]->thread() == thread) { 3259 mOutputTracks[i]->destroy(); 3260 mOutputTracks.removeAt(i); 3261 updateWaitTime(); 3262 return; 3263 } 3264 } 3265 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3266} 3267 3268void AudioFlinger::DuplicatingThread::updateWaitTime() 3269{ 3270 mWaitTimeMs = UINT_MAX; 3271 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3272 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3273 if (strong != 0) { 3274 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3275 if (waitTimeMs < mWaitTimeMs) { 3276 mWaitTimeMs = waitTimeMs; 3277 } 3278 } 3279 } 3280} 3281 3282 3283bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3284{ 3285 for (size_t i = 0; i < outputTracks.size(); i++) { 3286 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3287 if (thread == 0) { 3288 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3289 return false; 3290 } 3291 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3292 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3293 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3294 return false; 3295 } 3296 } 3297 return true; 3298} 3299 3300uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3301{ 3302 return (mWaitTimeMs * 1000) / 2; 3303} 3304 3305// ---------------------------------------------------------------------------- 3306 3307// TrackBase constructor must be called with AudioFlinger::mLock held 3308AudioFlinger::ThreadBase::TrackBase::TrackBase( 3309 ThreadBase *thread, 3310 const sp<Client>& client, 3311 uint32_t sampleRate, 3312 audio_format_t format, 3313 uint32_t channelMask, 3314 int frameCount, 3315 uint32_t flags, 3316 const sp<IMemory>& sharedBuffer, 3317 int sessionId) 3318 : RefBase(), 3319 mThread(thread), 3320 mClient(client), 3321 mCblk(NULL), 3322 // mBuffer 3323 // mBufferEnd 3324 mFrameCount(0), 3325 mState(IDLE), 3326 mFormat(format), 3327 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3328 mSessionId(sessionId) 3329 // mChannelCount 3330 // mChannelMask 3331{ 3332 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3333 3334 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3335 size_t size = sizeof(audio_track_cblk_t); 3336 uint8_t channelCount = popcount(channelMask); 3337 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3338 if (sharedBuffer == 0) { 3339 size += bufferSize; 3340 } 3341 3342 if (client != NULL) { 3343 mCblkMemory = client->heap()->allocate(size); 3344 if (mCblkMemory != 0) { 3345 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3346 if (mCblk != NULL) { // construct the shared structure in-place. 3347 new(mCblk) audio_track_cblk_t(); 3348 // clear all buffers 3349 mCblk->frameCount = frameCount; 3350 mCblk->sampleRate = sampleRate; 3351 mChannelCount = channelCount; 3352 mChannelMask = channelMask; 3353 if (sharedBuffer == 0) { 3354 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3355 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3356 // Force underrun condition to avoid false underrun callback until first data is 3357 // written to buffer (other flags are cleared) 3358 mCblk->flags = CBLK_UNDERRUN_ON; 3359 } else { 3360 mBuffer = sharedBuffer->pointer(); 3361 } 3362 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3363 } 3364 } else { 3365 ALOGE("not enough memory for AudioTrack size=%u", size); 3366 client->heap()->dump("AudioTrack"); 3367 return; 3368 } 3369 } else { 3370 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3371 // construct the shared structure in-place. 3372 new(mCblk) audio_track_cblk_t(); 3373 // clear all buffers 3374 mCblk->frameCount = frameCount; 3375 mCblk->sampleRate = sampleRate; 3376 mChannelCount = channelCount; 3377 mChannelMask = channelMask; 3378 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3379 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3380 // Force underrun condition to avoid false underrun callback until first data is 3381 // written to buffer (other flags are cleared) 3382 mCblk->flags = CBLK_UNDERRUN_ON; 3383 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3384 } 3385} 3386 3387AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3388{ 3389 if (mCblk != NULL) { 3390 if (mClient == 0) { 3391 delete mCblk; 3392 } else { 3393 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3394 } 3395 } 3396 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3397 if (mClient != 0) { 3398 // Client destructor must run with AudioFlinger mutex locked 3399 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3400 // If the client's reference count drops to zero, the associated destructor 3401 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3402 // relying on the automatic clear() at end of scope. 3403 mClient.clear(); 3404 } 3405} 3406 3407void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3408{ 3409 buffer->raw = NULL; 3410 mFrameCount = buffer->frameCount; 3411 step(); 3412 buffer->frameCount = 0; 3413} 3414 3415bool AudioFlinger::ThreadBase::TrackBase::step() { 3416 bool result; 3417 audio_track_cblk_t* cblk = this->cblk(); 3418 3419 result = cblk->stepServer(mFrameCount); 3420 if (!result) { 3421 ALOGV("stepServer failed acquiring cblk mutex"); 3422 mFlags |= STEPSERVER_FAILED; 3423 } 3424 return result; 3425} 3426 3427void AudioFlinger::ThreadBase::TrackBase::reset() { 3428 audio_track_cblk_t* cblk = this->cblk(); 3429 3430 cblk->user = 0; 3431 cblk->server = 0; 3432 cblk->userBase = 0; 3433 cblk->serverBase = 0; 3434 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3435 ALOGV("TrackBase::reset"); 3436} 3437 3438int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3439 return (int)mCblk->sampleRate; 3440} 3441 3442void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3443 audio_track_cblk_t* cblk = this->cblk(); 3444 size_t frameSize = cblk->frameSize; 3445 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3446 int8_t *bufferEnd = bufferStart + frames * frameSize; 3447 3448 // Check validity of returned pointer in case the track control block would have been corrupted. 3449 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3450 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3451 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3452 server %d, serverBase %d, user %d, userBase %d", 3453 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3454 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3455 return NULL; 3456 } 3457 3458 return bufferStart; 3459} 3460 3461// ---------------------------------------------------------------------------- 3462 3463// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3464AudioFlinger::PlaybackThread::Track::Track( 3465 PlaybackThread *thread, 3466 const sp<Client>& client, 3467 audio_stream_type_t streamType, 3468 uint32_t sampleRate, 3469 audio_format_t format, 3470 uint32_t channelMask, 3471 int frameCount, 3472 const sp<IMemory>& sharedBuffer, 3473 int sessionId) 3474 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3475 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3476 mAuxEffectId(0), mHasVolumeController(false) 3477{ 3478 if (mCblk != NULL) { 3479 if (thread != NULL) { 3480 mName = thread->getTrackName_l(); 3481 mMainBuffer = thread->mixBuffer(); 3482 } 3483 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3484 if (mName < 0) { 3485 ALOGE("no more track names available"); 3486 } 3487 mStreamType = streamType; 3488 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3489 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3490 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3491 } 3492} 3493 3494AudioFlinger::PlaybackThread::Track::~Track() 3495{ 3496 ALOGV("PlaybackThread::Track destructor"); 3497 sp<ThreadBase> thread = mThread.promote(); 3498 if (thread != 0) { 3499 Mutex::Autolock _l(thread->mLock); 3500 mState = TERMINATED; 3501 } 3502} 3503 3504void AudioFlinger::PlaybackThread::Track::destroy() 3505{ 3506 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3507 // by removing it from mTracks vector, so there is a risk that this Tracks's 3508 // destructor is called. As the destructor needs to lock mLock, 3509 // we must acquire a strong reference on this Track before locking mLock 3510 // here so that the destructor is called only when exiting this function. 3511 // On the other hand, as long as Track::destroy() is only called by 3512 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3513 // this Track with its member mTrack. 3514 sp<Track> keep(this); 3515 { // scope for mLock 3516 sp<ThreadBase> thread = mThread.promote(); 3517 if (thread != 0) { 3518 if (!isOutputTrack()) { 3519 if (mState == ACTIVE || mState == RESUMING) { 3520 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3521 3522 // to track the speaker usage 3523 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3524 } 3525 AudioSystem::releaseOutput(thread->id()); 3526 } 3527 Mutex::Autolock _l(thread->mLock); 3528 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3529 playbackThread->destroyTrack_l(this); 3530 } 3531 } 3532} 3533 3534void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3535{ 3536 uint32_t vlr = mCblk->getVolumeLR(); 3537 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", 3538 mName - AudioMixer::TRACK0, 3539 (mClient == 0) ? getpid_cached : mClient->pid(), 3540 mStreamType, 3541 mFormat, 3542 mChannelMask, 3543 mSessionId, 3544 mFrameCount, 3545 mState, 3546 mMute, 3547 mFillingUpStatus, 3548 mCblk->sampleRate, 3549 vlr & 0xFFFF, 3550 vlr >> 16, 3551 mCblk->server, 3552 mCblk->user, 3553 (int)mMainBuffer, 3554 (int)mAuxBuffer); 3555} 3556 3557status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3558 AudioBufferProvider::Buffer* buffer, int64_t pts) 3559{ 3560 audio_track_cblk_t* cblk = this->cblk(); 3561 uint32_t framesReady; 3562 uint32_t framesReq = buffer->frameCount; 3563 3564 // Check if last stepServer failed, try to step now 3565 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3566 if (!step()) goto getNextBuffer_exit; 3567 ALOGV("stepServer recovered"); 3568 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3569 } 3570 3571 framesReady = cblk->framesReady(); 3572 3573 if (CC_LIKELY(framesReady)) { 3574 uint32_t s = cblk->server; 3575 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3576 3577 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3578 if (framesReq > framesReady) { 3579 framesReq = framesReady; 3580 } 3581 if (s + framesReq > bufferEnd) { 3582 framesReq = bufferEnd - s; 3583 } 3584 3585 buffer->raw = getBuffer(s, framesReq); 3586 if (buffer->raw == NULL) goto getNextBuffer_exit; 3587 3588 buffer->frameCount = framesReq; 3589 return NO_ERROR; 3590 } 3591 3592getNextBuffer_exit: 3593 buffer->raw = NULL; 3594 buffer->frameCount = 0; 3595 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3596 return NOT_ENOUGH_DATA; 3597} 3598 3599uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3600 return mCblk->framesReady(); 3601} 3602 3603bool AudioFlinger::PlaybackThread::Track::isReady() const { 3604 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3605 3606 if (framesReady() >= mCblk->frameCount || 3607 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3608 mFillingUpStatus = FS_FILLED; 3609 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3610 return true; 3611 } 3612 return false; 3613} 3614 3615status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3616{ 3617 status_t status = NO_ERROR; 3618 ALOGV("start(%d), calling pid %d session %d tid %d", 3619 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3620 sp<ThreadBase> thread = mThread.promote(); 3621 if (thread != 0) { 3622 Mutex::Autolock _l(thread->mLock); 3623 track_state state = mState; 3624 // here the track could be either new, or restarted 3625 // in both cases "unstop" the track 3626 if (mState == PAUSED) { 3627 mState = TrackBase::RESUMING; 3628 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3629 } else { 3630 mState = TrackBase::ACTIVE; 3631 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3632 } 3633 3634 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3635 thread->mLock.unlock(); 3636 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3637 thread->mLock.lock(); 3638 3639 // to track the speaker usage 3640 if (status == NO_ERROR) { 3641 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3642 } 3643 } 3644 if (status == NO_ERROR) { 3645 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3646 playbackThread->addTrack_l(this); 3647 } else { 3648 mState = state; 3649 } 3650 } else { 3651 status = BAD_VALUE; 3652 } 3653 return status; 3654} 3655 3656void AudioFlinger::PlaybackThread::Track::stop() 3657{ 3658 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3659 sp<ThreadBase> thread = mThread.promote(); 3660 if (thread != 0) { 3661 Mutex::Autolock _l(thread->mLock); 3662 track_state state = mState; 3663 if (mState > STOPPED) { 3664 mState = STOPPED; 3665 // If the track is not active (PAUSED and buffers full), flush buffers 3666 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3667 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3668 reset(); 3669 } 3670 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3671 } 3672 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3673 thread->mLock.unlock(); 3674 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3675 thread->mLock.lock(); 3676 3677 // to track the speaker usage 3678 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3679 } 3680 } 3681} 3682 3683void AudioFlinger::PlaybackThread::Track::pause() 3684{ 3685 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3686 sp<ThreadBase> thread = mThread.promote(); 3687 if (thread != 0) { 3688 Mutex::Autolock _l(thread->mLock); 3689 if (mState == ACTIVE || mState == RESUMING) { 3690 mState = PAUSING; 3691 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3692 if (!isOutputTrack()) { 3693 thread->mLock.unlock(); 3694 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3695 thread->mLock.lock(); 3696 3697 // to track the speaker usage 3698 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3699 } 3700 } 3701 } 3702} 3703 3704void AudioFlinger::PlaybackThread::Track::flush() 3705{ 3706 ALOGV("flush(%d)", mName); 3707 sp<ThreadBase> thread = mThread.promote(); 3708 if (thread != 0) { 3709 Mutex::Autolock _l(thread->mLock); 3710 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3711 return; 3712 } 3713 // No point remaining in PAUSED state after a flush => go to 3714 // STOPPED state 3715 mState = STOPPED; 3716 3717 // do not reset the track if it is still in the process of being stopped or paused. 3718 // this will be done by prepareTracks_l() when the track is stopped. 3719 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3720 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3721 reset(); 3722 } 3723 } 3724} 3725 3726void AudioFlinger::PlaybackThread::Track::reset() 3727{ 3728 // Do not reset twice to avoid discarding data written just after a flush and before 3729 // the audioflinger thread detects the track is stopped. 3730 if (!mResetDone) { 3731 TrackBase::reset(); 3732 // Force underrun condition to avoid false underrun callback until first data is 3733 // written to buffer 3734 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3735 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3736 mFillingUpStatus = FS_FILLING; 3737 mResetDone = true; 3738 } 3739} 3740 3741void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3742{ 3743 mMute = muted; 3744} 3745 3746status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3747{ 3748 status_t status = DEAD_OBJECT; 3749 sp<ThreadBase> thread = mThread.promote(); 3750 if (thread != 0) { 3751 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3752 status = playbackThread->attachAuxEffect(this, EffectId); 3753 } 3754 return status; 3755} 3756 3757void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3758{ 3759 mAuxEffectId = EffectId; 3760 mAuxBuffer = buffer; 3761} 3762 3763// timed audio tracks 3764 3765sp<AudioFlinger::PlaybackThread::TimedTrack> 3766AudioFlinger::PlaybackThread::TimedTrack::create( 3767 PlaybackThread *thread, 3768 const sp<Client>& client, 3769 audio_stream_type_t streamType, 3770 uint32_t sampleRate, 3771 audio_format_t format, 3772 uint32_t channelMask, 3773 int frameCount, 3774 const sp<IMemory>& sharedBuffer, 3775 int sessionId) { 3776 if (!client->reserveTimedTrack()) 3777 return NULL; 3778 3779 sp<TimedTrack> track = new TimedTrack( 3780 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3781 sharedBuffer, sessionId); 3782 3783 if (track == NULL) { 3784 client->releaseTimedTrack(); 3785 return NULL; 3786 } 3787 3788 return track; 3789} 3790 3791AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3792 PlaybackThread *thread, 3793 const sp<Client>& client, 3794 audio_stream_type_t streamType, 3795 uint32_t sampleRate, 3796 audio_format_t format, 3797 uint32_t channelMask, 3798 int frameCount, 3799 const sp<IMemory>& sharedBuffer, 3800 int sessionId) 3801 : Track(thread, client, streamType, sampleRate, format, channelMask, 3802 frameCount, sharedBuffer, sessionId), 3803 mTimedSilenceBuffer(NULL), 3804 mTimedSilenceBufferSize(0), 3805 mTimedAudioOutputOnTime(false), 3806 mMediaTimeTransformValid(false) 3807{ 3808 LocalClock lc; 3809 mLocalTimeFreq = lc.getLocalFreq(); 3810 3811 mLocalTimeToSampleTransform.a_zero = 0; 3812 mLocalTimeToSampleTransform.b_zero = 0; 3813 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3814 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3815 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3816 &mLocalTimeToSampleTransform.a_to_b_denom); 3817} 3818 3819AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3820 mClient->releaseTimedTrack(); 3821 delete [] mTimedSilenceBuffer; 3822} 3823 3824status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3825 size_t size, sp<IMemory>* buffer) { 3826 3827 Mutex::Autolock _l(mTimedBufferQueueLock); 3828 3829 trimTimedBufferQueue_l(); 3830 3831 // lazily initialize the shared memory heap for timed buffers 3832 if (mTimedMemoryDealer == NULL) { 3833 const int kTimedBufferHeapSize = 512 << 10; 3834 3835 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3836 "AudioFlingerTimed"); 3837 if (mTimedMemoryDealer == NULL) 3838 return NO_MEMORY; 3839 } 3840 3841 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3842 if (newBuffer == NULL) { 3843 newBuffer = mTimedMemoryDealer->allocate(size); 3844 if (newBuffer == NULL) 3845 return NO_MEMORY; 3846 } 3847 3848 *buffer = newBuffer; 3849 return NO_ERROR; 3850} 3851 3852// caller must hold mTimedBufferQueueLock 3853void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3854 int64_t mediaTimeNow; 3855 { 3856 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3857 if (!mMediaTimeTransformValid) 3858 return; 3859 3860 int64_t targetTimeNow; 3861 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3862 ? mCCHelper.getCommonTime(&targetTimeNow) 3863 : mCCHelper.getLocalTime(&targetTimeNow); 3864 3865 if (OK != res) 3866 return; 3867 3868 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3869 &mediaTimeNow)) { 3870 return; 3871 } 3872 } 3873 3874 size_t trimIndex; 3875 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3876 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3877 break; 3878 } 3879 3880 if (trimIndex) { 3881 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3882 } 3883} 3884 3885status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3886 const sp<IMemory>& buffer, int64_t pts) { 3887 3888 { 3889 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3890 if (!mMediaTimeTransformValid) 3891 return INVALID_OPERATION; 3892 } 3893 3894 Mutex::Autolock _l(mTimedBufferQueueLock); 3895 3896 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3897 3898 return NO_ERROR; 3899} 3900 3901status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3902 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3903 3904 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3905 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3906 target); 3907 3908 if (!(target == TimedAudioTrack::LOCAL_TIME || 3909 target == TimedAudioTrack::COMMON_TIME)) { 3910 return BAD_VALUE; 3911 } 3912 3913 Mutex::Autolock lock(mMediaTimeTransformLock); 3914 mMediaTimeTransform = xform; 3915 mMediaTimeTransformTarget = target; 3916 mMediaTimeTransformValid = true; 3917 3918 return NO_ERROR; 3919} 3920 3921#define min(a, b) ((a) < (b) ? (a) : (b)) 3922 3923// implementation of getNextBuffer for tracks whose buffers have timestamps 3924status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3925 AudioBufferProvider::Buffer* buffer, int64_t pts) 3926{ 3927 if (pts == AudioBufferProvider::kInvalidPTS) { 3928 buffer->raw = 0; 3929 buffer->frameCount = 0; 3930 return INVALID_OPERATION; 3931 } 3932 3933 Mutex::Autolock _l(mTimedBufferQueueLock); 3934 3935 while (true) { 3936 3937 // if we have no timed buffers, then fail 3938 if (mTimedBufferQueue.isEmpty()) { 3939 buffer->raw = 0; 3940 buffer->frameCount = 0; 3941 return NOT_ENOUGH_DATA; 3942 } 3943 3944 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3945 3946 // calculate the PTS of the head of the timed buffer queue expressed in 3947 // local time 3948 int64_t headLocalPTS; 3949 { 3950 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3951 3952 assert(mMediaTimeTransformValid); 3953 3954 if (mMediaTimeTransform.a_to_b_denom == 0) { 3955 // the transform represents a pause, so yield silence 3956 timedYieldSilence(buffer->frameCount, buffer); 3957 return NO_ERROR; 3958 } 3959 3960 int64_t transformedPTS; 3961 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3962 &transformedPTS)) { 3963 // the transform failed. this shouldn't happen, but if it does 3964 // then just drop this buffer 3965 ALOGW("timedGetNextBuffer transform failed"); 3966 buffer->raw = 0; 3967 buffer->frameCount = 0; 3968 mTimedBufferQueue.removeAt(0); 3969 return NO_ERROR; 3970 } 3971 3972 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3973 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3974 &headLocalPTS)) { 3975 buffer->raw = 0; 3976 buffer->frameCount = 0; 3977 return INVALID_OPERATION; 3978 } 3979 } else { 3980 headLocalPTS = transformedPTS; 3981 } 3982 } 3983 3984 // adjust the head buffer's PTS to reflect the portion of the head buffer 3985 // that has already been consumed 3986 int64_t effectivePTS = headLocalPTS + 3987 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3988 3989 // Calculate the delta in samples between the head of the input buffer 3990 // queue and the start of the next output buffer that will be written. 3991 // If the transformation fails because of over or underflow, it means 3992 // that the sample's position in the output stream is so far out of 3993 // whack that it should just be dropped. 3994 int64_t sampleDelta; 3995 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3996 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3997 mTimedBufferQueue.removeAt(0); 3998 continue; 3999 } 4000 if (!mLocalTimeToSampleTransform.doForwardTransform( 4001 (effectivePTS - pts) << 32, &sampleDelta)) { 4002 ALOGV("*** too late during sample rate transform: dropped buffer"); 4003 mTimedBufferQueue.removeAt(0); 4004 continue; 4005 } 4006 4007 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4008 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4009 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4010 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4011 4012 // if the delta between the ideal placement for the next input sample and 4013 // the current output position is within this threshold, then we will 4014 // concatenate the next input samples to the previous output 4015 const int64_t kSampleContinuityThreshold = 4016 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4017 4018 // if this is the first buffer of audio that we're emitting from this track 4019 // then it should be almost exactly on time. 4020 const int64_t kSampleStartupThreshold = 1LL << 32; 4021 4022 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4023 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4024 // the next input is close enough to being on time, so concatenate it 4025 // with the last output 4026 timedYieldSamples(buffer); 4027 4028 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4029 return NO_ERROR; 4030 } else if (sampleDelta > 0) { 4031 // the gap between the current output position and the proper start of 4032 // the next input sample is too big, so fill it with silence 4033 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4034 4035 timedYieldSilence(framesUntilNextInput, buffer); 4036 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4037 return NO_ERROR; 4038 } else { 4039 // the next input sample is late 4040 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4041 size_t onTimeSamplePosition = 4042 head.position() + lateFrames * mCblk->frameSize; 4043 4044 if (onTimeSamplePosition > head.buffer()->size()) { 4045 // all the remaining samples in the head are too late, so 4046 // drop it and move on 4047 ALOGV("*** too late: dropped buffer"); 4048 mTimedBufferQueue.removeAt(0); 4049 continue; 4050 } else { 4051 // skip over the late samples 4052 head.setPosition(onTimeSamplePosition); 4053 4054 // yield the available samples 4055 timedYieldSamples(buffer); 4056 4057 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4058 return NO_ERROR; 4059 } 4060 } 4061 } 4062} 4063 4064// Yield samples from the timed buffer queue head up to the given output 4065// buffer's capacity. 4066// 4067// Caller must hold mTimedBufferQueueLock 4068void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4069 AudioBufferProvider::Buffer* buffer) { 4070 4071 const TimedBuffer& head = mTimedBufferQueue[0]; 4072 4073 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4074 head.position()); 4075 4076 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4077 mCblk->frameSize); 4078 size_t framesRequested = buffer->frameCount; 4079 buffer->frameCount = min(framesLeftInHead, framesRequested); 4080 4081 mTimedAudioOutputOnTime = true; 4082} 4083 4084// Yield samples of silence up to the given output buffer's capacity 4085// 4086// Caller must hold mTimedBufferQueueLock 4087void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4088 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4089 4090 // lazily allocate a buffer filled with silence 4091 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4092 delete [] mTimedSilenceBuffer; 4093 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4094 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4095 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4096 } 4097 4098 buffer->raw = mTimedSilenceBuffer; 4099 size_t framesRequested = buffer->frameCount; 4100 buffer->frameCount = min(numFrames, framesRequested); 4101 4102 mTimedAudioOutputOnTime = false; 4103} 4104 4105void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4106 AudioBufferProvider::Buffer* buffer) { 4107 4108 Mutex::Autolock _l(mTimedBufferQueueLock); 4109 4110 // If the buffer which was just released is part of the buffer at the head 4111 // of the queue, be sure to update the amt of the buffer which has been 4112 // consumed. If the buffer being returned is not part of the head of the 4113 // queue, its either because the buffer is part of the silence buffer, or 4114 // because the head of the timed queue was trimmed after the mixer called 4115 // getNextBuffer but before the mixer called releaseBuffer. 4116 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4117 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4118 4119 void* start = head.buffer()->pointer(); 4120 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4121 4122 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4123 head.setPosition(head.position() + 4124 (buffer->frameCount * mCblk->frameSize)); 4125 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4126 mTimedBufferQueue.removeAt(0); 4127 } 4128 } 4129 } 4130 4131 buffer->raw = 0; 4132 buffer->frameCount = 0; 4133} 4134 4135uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4136 Mutex::Autolock _l(mTimedBufferQueueLock); 4137 4138 uint32_t frames = 0; 4139 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4140 const TimedBuffer& tb = mTimedBufferQueue[i]; 4141 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4142 } 4143 4144 return frames; 4145} 4146 4147AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4148 : mPTS(0), mPosition(0) {} 4149 4150AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4151 const sp<IMemory>& buffer, int64_t pts) 4152 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4153 4154// ---------------------------------------------------------------------------- 4155 4156// RecordTrack constructor must be called with AudioFlinger::mLock held 4157AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4158 RecordThread *thread, 4159 const sp<Client>& client, 4160 uint32_t sampleRate, 4161 audio_format_t format, 4162 uint32_t channelMask, 4163 int frameCount, 4164 uint32_t flags, 4165 int sessionId) 4166 : TrackBase(thread, client, sampleRate, format, 4167 channelMask, frameCount, flags, 0, sessionId), 4168 mOverflow(false) 4169{ 4170 if (mCblk != NULL) { 4171 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4172 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4173 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4174 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4175 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4176 } else { 4177 mCblk->frameSize = sizeof(int8_t); 4178 } 4179 } 4180} 4181 4182AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4183{ 4184 sp<ThreadBase> thread = mThread.promote(); 4185 if (thread != 0) { 4186 AudioSystem::releaseInput(thread->id()); 4187 } 4188} 4189 4190status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4191{ 4192 audio_track_cblk_t* cblk = this->cblk(); 4193 uint32_t framesAvail; 4194 uint32_t framesReq = buffer->frameCount; 4195 4196 // Check if last stepServer failed, try to step now 4197 if (mFlags & TrackBase::STEPSERVER_FAILED) { 4198 if (!step()) goto getNextBuffer_exit; 4199 ALOGV("stepServer recovered"); 4200 mFlags &= ~TrackBase::STEPSERVER_FAILED; 4201 } 4202 4203 framesAvail = cblk->framesAvailable_l(); 4204 4205 if (CC_LIKELY(framesAvail)) { 4206 uint32_t s = cblk->server; 4207 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4208 4209 if (framesReq > framesAvail) { 4210 framesReq = framesAvail; 4211 } 4212 if (s + framesReq > bufferEnd) { 4213 framesReq = bufferEnd - s; 4214 } 4215 4216 buffer->raw = getBuffer(s, framesReq); 4217 if (buffer->raw == NULL) goto getNextBuffer_exit; 4218 4219 buffer->frameCount = framesReq; 4220 return NO_ERROR; 4221 } 4222 4223getNextBuffer_exit: 4224 buffer->raw = NULL; 4225 buffer->frameCount = 0; 4226 return NOT_ENOUGH_DATA; 4227} 4228 4229status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4230{ 4231 sp<ThreadBase> thread = mThread.promote(); 4232 if (thread != 0) { 4233 RecordThread *recordThread = (RecordThread *)thread.get(); 4234 return recordThread->start(this, tid); 4235 } else { 4236 return BAD_VALUE; 4237 } 4238} 4239 4240void AudioFlinger::RecordThread::RecordTrack::stop() 4241{ 4242 sp<ThreadBase> thread = mThread.promote(); 4243 if (thread != 0) { 4244 RecordThread *recordThread = (RecordThread *)thread.get(); 4245 recordThread->stop(this); 4246 TrackBase::reset(); 4247 // Force overerrun condition to avoid false overrun callback until first data is 4248 // read from buffer 4249 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4250 } 4251} 4252 4253void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4254{ 4255 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4256 (mClient == 0) ? getpid_cached : mClient->pid(), 4257 mFormat, 4258 mChannelMask, 4259 mSessionId, 4260 mFrameCount, 4261 mState, 4262 mCblk->sampleRate, 4263 mCblk->server, 4264 mCblk->user); 4265} 4266 4267 4268// ---------------------------------------------------------------------------- 4269 4270AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4271 PlaybackThread *playbackThread, 4272 DuplicatingThread *sourceThread, 4273 uint32_t sampleRate, 4274 audio_format_t format, 4275 uint32_t channelMask, 4276 int frameCount) 4277 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4278 mActive(false), mSourceThread(sourceThread) 4279{ 4280 4281 if (mCblk != NULL) { 4282 mCblk->flags |= CBLK_DIRECTION_OUT; 4283 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4284 mOutBuffer.frameCount = 0; 4285 playbackThread->mTracks.add(this); 4286 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4287 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4288 mCblk, mBuffer, mCblk->buffers, 4289 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4290 } else { 4291 ALOGW("Error creating output track on thread %p", playbackThread); 4292 } 4293} 4294 4295AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4296{ 4297 clearBufferQueue(); 4298} 4299 4300status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4301{ 4302 status_t status = Track::start(tid); 4303 if (status != NO_ERROR) { 4304 return status; 4305 } 4306 4307 mActive = true; 4308 mRetryCount = 127; 4309 return status; 4310} 4311 4312void AudioFlinger::PlaybackThread::OutputTrack::stop() 4313{ 4314 Track::stop(); 4315 clearBufferQueue(); 4316 mOutBuffer.frameCount = 0; 4317 mActive = false; 4318} 4319 4320bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4321{ 4322 Buffer *pInBuffer; 4323 Buffer inBuffer; 4324 uint32_t channelCount = mChannelCount; 4325 bool outputBufferFull = false; 4326 inBuffer.frameCount = frames; 4327 inBuffer.i16 = data; 4328 4329 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4330 4331 if (!mActive && frames != 0) { 4332 start(0); 4333 sp<ThreadBase> thread = mThread.promote(); 4334 if (thread != 0) { 4335 MixerThread *mixerThread = (MixerThread *)thread.get(); 4336 if (mCblk->frameCount > frames){ 4337 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4338 uint32_t startFrames = (mCblk->frameCount - frames); 4339 pInBuffer = new Buffer; 4340 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4341 pInBuffer->frameCount = startFrames; 4342 pInBuffer->i16 = pInBuffer->mBuffer; 4343 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4344 mBufferQueue.add(pInBuffer); 4345 } else { 4346 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4347 } 4348 } 4349 } 4350 } 4351 4352 while (waitTimeLeftMs) { 4353 // First write pending buffers, then new data 4354 if (mBufferQueue.size()) { 4355 pInBuffer = mBufferQueue.itemAt(0); 4356 } else { 4357 pInBuffer = &inBuffer; 4358 } 4359 4360 if (pInBuffer->frameCount == 0) { 4361 break; 4362 } 4363 4364 if (mOutBuffer.frameCount == 0) { 4365 mOutBuffer.frameCount = pInBuffer->frameCount; 4366 nsecs_t startTime = systemTime(); 4367 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4368 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4369 outputBufferFull = true; 4370 break; 4371 } 4372 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4373 if (waitTimeLeftMs >= waitTimeMs) { 4374 waitTimeLeftMs -= waitTimeMs; 4375 } else { 4376 waitTimeLeftMs = 0; 4377 } 4378 } 4379 4380 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4381 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4382 mCblk->stepUser(outFrames); 4383 pInBuffer->frameCount -= outFrames; 4384 pInBuffer->i16 += outFrames * channelCount; 4385 mOutBuffer.frameCount -= outFrames; 4386 mOutBuffer.i16 += outFrames * channelCount; 4387 4388 if (pInBuffer->frameCount == 0) { 4389 if (mBufferQueue.size()) { 4390 mBufferQueue.removeAt(0); 4391 delete [] pInBuffer->mBuffer; 4392 delete pInBuffer; 4393 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4394 } else { 4395 break; 4396 } 4397 } 4398 } 4399 4400 // If we could not write all frames, allocate a buffer and queue it for next time. 4401 if (inBuffer.frameCount) { 4402 sp<ThreadBase> thread = mThread.promote(); 4403 if (thread != 0 && !thread->standby()) { 4404 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4405 pInBuffer = new Buffer; 4406 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4407 pInBuffer->frameCount = inBuffer.frameCount; 4408 pInBuffer->i16 = pInBuffer->mBuffer; 4409 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4410 mBufferQueue.add(pInBuffer); 4411 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4412 } else { 4413 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4414 } 4415 } 4416 } 4417 4418 // Calling write() with a 0 length buffer, means that no more data will be written: 4419 // If no more buffers are pending, fill output track buffer to make sure it is started 4420 // by output mixer. 4421 if (frames == 0 && mBufferQueue.size() == 0) { 4422 if (mCblk->user < mCblk->frameCount) { 4423 frames = mCblk->frameCount - mCblk->user; 4424 pInBuffer = new Buffer; 4425 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4426 pInBuffer->frameCount = frames; 4427 pInBuffer->i16 = pInBuffer->mBuffer; 4428 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4429 mBufferQueue.add(pInBuffer); 4430 } else if (mActive) { 4431 stop(); 4432 } 4433 } 4434 4435 return outputBufferFull; 4436} 4437 4438status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4439{ 4440 int active; 4441 status_t result; 4442 audio_track_cblk_t* cblk = mCblk; 4443 uint32_t framesReq = buffer->frameCount; 4444 4445// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4446 buffer->frameCount = 0; 4447 4448 uint32_t framesAvail = cblk->framesAvailable(); 4449 4450 4451 if (framesAvail == 0) { 4452 Mutex::Autolock _l(cblk->lock); 4453 goto start_loop_here; 4454 while (framesAvail == 0) { 4455 active = mActive; 4456 if (CC_UNLIKELY(!active)) { 4457 ALOGV("Not active and NO_MORE_BUFFERS"); 4458 return NO_MORE_BUFFERS; 4459 } 4460 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4461 if (result != NO_ERROR) { 4462 return NO_MORE_BUFFERS; 4463 } 4464 // read the server count again 4465 start_loop_here: 4466 framesAvail = cblk->framesAvailable_l(); 4467 } 4468 } 4469 4470// if (framesAvail < framesReq) { 4471// return NO_MORE_BUFFERS; 4472// } 4473 4474 if (framesReq > framesAvail) { 4475 framesReq = framesAvail; 4476 } 4477 4478 uint32_t u = cblk->user; 4479 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4480 4481 if (u + framesReq > bufferEnd) { 4482 framesReq = bufferEnd - u; 4483 } 4484 4485 buffer->frameCount = framesReq; 4486 buffer->raw = (void *)cblk->buffer(u); 4487 return NO_ERROR; 4488} 4489 4490 4491void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4492{ 4493 size_t size = mBufferQueue.size(); 4494 4495 for (size_t i = 0; i < size; i++) { 4496 Buffer *pBuffer = mBufferQueue.itemAt(i); 4497 delete [] pBuffer->mBuffer; 4498 delete pBuffer; 4499 } 4500 mBufferQueue.clear(); 4501} 4502 4503// ---------------------------------------------------------------------------- 4504 4505AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4506 : RefBase(), 4507 mAudioFlinger(audioFlinger), 4508 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4509 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4510 mPid(pid), 4511 mTimedTrackCount(0) 4512{ 4513 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4514} 4515 4516// Client destructor must be called with AudioFlinger::mLock held 4517AudioFlinger::Client::~Client() 4518{ 4519 mAudioFlinger->removeClient_l(mPid); 4520} 4521 4522sp<MemoryDealer> AudioFlinger::Client::heap() const 4523{ 4524 return mMemoryDealer; 4525} 4526 4527// Reserve one of the limited slots for a timed audio track associated 4528// with this client 4529bool AudioFlinger::Client::reserveTimedTrack() 4530{ 4531 const int kMaxTimedTracksPerClient = 4; 4532 4533 Mutex::Autolock _l(mTimedTrackLock); 4534 4535 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4536 ALOGW("can not create timed track - pid %d has exceeded the limit", 4537 mPid); 4538 return false; 4539 } 4540 4541 mTimedTrackCount++; 4542 return true; 4543} 4544 4545// Release a slot for a timed audio track 4546void AudioFlinger::Client::releaseTimedTrack() 4547{ 4548 Mutex::Autolock _l(mTimedTrackLock); 4549 mTimedTrackCount--; 4550} 4551 4552// ---------------------------------------------------------------------------- 4553 4554AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4555 const sp<IAudioFlingerClient>& client, 4556 pid_t pid) 4557 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4558{ 4559} 4560 4561AudioFlinger::NotificationClient::~NotificationClient() 4562{ 4563} 4564 4565void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4566{ 4567 sp<NotificationClient> keep(this); 4568 mAudioFlinger->removeNotificationClient(mPid); 4569} 4570 4571// ---------------------------------------------------------------------------- 4572 4573AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4574 : BnAudioTrack(), 4575 mTrack(track) 4576{ 4577} 4578 4579AudioFlinger::TrackHandle::~TrackHandle() { 4580 // just stop the track on deletion, associated resources 4581 // will be freed from the main thread once all pending buffers have 4582 // been played. Unless it's not in the active track list, in which 4583 // case we free everything now... 4584 mTrack->destroy(); 4585} 4586 4587sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4588 return mTrack->getCblk(); 4589} 4590 4591status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4592 return mTrack->start(tid); 4593} 4594 4595void AudioFlinger::TrackHandle::stop() { 4596 mTrack->stop(); 4597} 4598 4599void AudioFlinger::TrackHandle::flush() { 4600 mTrack->flush(); 4601} 4602 4603void AudioFlinger::TrackHandle::mute(bool e) { 4604 mTrack->mute(e); 4605} 4606 4607void AudioFlinger::TrackHandle::pause() { 4608 mTrack->pause(); 4609} 4610 4611status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4612{ 4613 return mTrack->attachAuxEffect(EffectId); 4614} 4615 4616status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4617 sp<IMemory>* buffer) { 4618 if (!mTrack->isTimedTrack()) 4619 return INVALID_OPERATION; 4620 4621 PlaybackThread::TimedTrack* tt = 4622 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4623 return tt->allocateTimedBuffer(size, buffer); 4624} 4625 4626status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4627 int64_t pts) { 4628 if (!mTrack->isTimedTrack()) 4629 return INVALID_OPERATION; 4630 4631 PlaybackThread::TimedTrack* tt = 4632 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4633 return tt->queueTimedBuffer(buffer, pts); 4634} 4635 4636status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4637 const LinearTransform& xform, int target) { 4638 4639 if (!mTrack->isTimedTrack()) 4640 return INVALID_OPERATION; 4641 4642 PlaybackThread::TimedTrack* tt = 4643 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4644 return tt->setMediaTimeTransform( 4645 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4646} 4647 4648status_t AudioFlinger::TrackHandle::onTransact( 4649 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4650{ 4651 return BnAudioTrack::onTransact(code, data, reply, flags); 4652} 4653 4654// ---------------------------------------------------------------------------- 4655 4656sp<IAudioRecord> AudioFlinger::openRecord( 4657 pid_t pid, 4658 audio_io_handle_t input, 4659 uint32_t sampleRate, 4660 audio_format_t format, 4661 uint32_t channelMask, 4662 int frameCount, 4663 uint32_t flags, 4664 int *sessionId, 4665 status_t *status) 4666{ 4667 sp<RecordThread::RecordTrack> recordTrack; 4668 sp<RecordHandle> recordHandle; 4669 sp<Client> client; 4670 status_t lStatus; 4671 RecordThread *thread; 4672 size_t inFrameCount; 4673 int lSessionId; 4674 4675 // check calling permissions 4676 if (!recordingAllowed()) { 4677 lStatus = PERMISSION_DENIED; 4678 goto Exit; 4679 } 4680 4681 // add client to list 4682 { // scope for mLock 4683 Mutex::Autolock _l(mLock); 4684 thread = checkRecordThread_l(input); 4685 if (thread == NULL) { 4686 lStatus = BAD_VALUE; 4687 goto Exit; 4688 } 4689 4690 client = registerPid_l(pid); 4691 4692 // If no audio session id is provided, create one here 4693 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4694 lSessionId = *sessionId; 4695 } else { 4696 lSessionId = nextUniqueId(); 4697 if (sessionId != NULL) { 4698 *sessionId = lSessionId; 4699 } 4700 } 4701 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4702 recordTrack = thread->createRecordTrack_l(client, 4703 sampleRate, 4704 format, 4705 channelMask, 4706 frameCount, 4707 flags, 4708 lSessionId, 4709 &lStatus); 4710 } 4711 if (lStatus != NO_ERROR) { 4712 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4713 // destructor is called by the TrackBase destructor with mLock held 4714 client.clear(); 4715 recordTrack.clear(); 4716 goto Exit; 4717 } 4718 4719 // return to handle to client 4720 recordHandle = new RecordHandle(recordTrack); 4721 lStatus = NO_ERROR; 4722 4723Exit: 4724 if (status) { 4725 *status = lStatus; 4726 } 4727 return recordHandle; 4728} 4729 4730// ---------------------------------------------------------------------------- 4731 4732AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4733 : BnAudioRecord(), 4734 mRecordTrack(recordTrack) 4735{ 4736} 4737 4738AudioFlinger::RecordHandle::~RecordHandle() { 4739 stop(); 4740} 4741 4742sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4743 return mRecordTrack->getCblk(); 4744} 4745 4746status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4747 ALOGV("RecordHandle::start()"); 4748 return mRecordTrack->start(tid); 4749} 4750 4751void AudioFlinger::RecordHandle::stop() { 4752 ALOGV("RecordHandle::stop()"); 4753 mRecordTrack->stop(); 4754} 4755 4756status_t AudioFlinger::RecordHandle::onTransact( 4757 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4758{ 4759 return BnAudioRecord::onTransact(code, data, reply, flags); 4760} 4761 4762// ---------------------------------------------------------------------------- 4763 4764AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4765 AudioStreamIn *input, 4766 uint32_t sampleRate, 4767 uint32_t channels, 4768 audio_io_handle_t id, 4769 uint32_t device) : 4770 ThreadBase(audioFlinger, id, device, RECORD), 4771 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4772 // mRsmpInIndex and mInputBytes set by readInputParameters() 4773 mReqChannelCount(popcount(channels)), 4774 mReqSampleRate(sampleRate) 4775 // mBytesRead is only meaningful while active, and so is cleared in start() 4776 // (but might be better to also clear here for dump?) 4777{ 4778 snprintf(mName, kNameLength, "AudioIn_%d", id); 4779 4780 readInputParameters(); 4781} 4782 4783 4784AudioFlinger::RecordThread::~RecordThread() 4785{ 4786 delete[] mRsmpInBuffer; 4787 delete mResampler; 4788 delete[] mRsmpOutBuffer; 4789} 4790 4791void AudioFlinger::RecordThread::onFirstRef() 4792{ 4793 run(mName, PRIORITY_URGENT_AUDIO); 4794} 4795 4796status_t AudioFlinger::RecordThread::readyToRun() 4797{ 4798 status_t status = initCheck(); 4799 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4800 return status; 4801} 4802 4803bool AudioFlinger::RecordThread::threadLoop() 4804{ 4805 AudioBufferProvider::Buffer buffer; 4806 sp<RecordTrack> activeTrack; 4807 Vector< sp<EffectChain> > effectChains; 4808 4809 nsecs_t lastWarning = 0; 4810 4811 acquireWakeLock(); 4812 4813 // start recording 4814 while (!exitPending()) { 4815 4816 processConfigEvents(); 4817 4818 { // scope for mLock 4819 Mutex::Autolock _l(mLock); 4820 checkForNewParameters_l(); 4821 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4822 if (!mStandby) { 4823 mInput->stream->common.standby(&mInput->stream->common); 4824 mStandby = true; 4825 } 4826 4827 if (exitPending()) break; 4828 4829 releaseWakeLock_l(); 4830 ALOGV("RecordThread: loop stopping"); 4831 // go to sleep 4832 mWaitWorkCV.wait(mLock); 4833 ALOGV("RecordThread: loop starting"); 4834 acquireWakeLock_l(); 4835 continue; 4836 } 4837 if (mActiveTrack != 0) { 4838 if (mActiveTrack->mState == TrackBase::PAUSING) { 4839 if (!mStandby) { 4840 mInput->stream->common.standby(&mInput->stream->common); 4841 mStandby = true; 4842 } 4843 mActiveTrack.clear(); 4844 mStartStopCond.broadcast(); 4845 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4846 if (mReqChannelCount != mActiveTrack->channelCount()) { 4847 mActiveTrack.clear(); 4848 mStartStopCond.broadcast(); 4849 } else if (mBytesRead != 0) { 4850 // record start succeeds only if first read from audio input 4851 // succeeds 4852 if (mBytesRead > 0) { 4853 mActiveTrack->mState = TrackBase::ACTIVE; 4854 } else { 4855 mActiveTrack.clear(); 4856 } 4857 mStartStopCond.broadcast(); 4858 } 4859 mStandby = false; 4860 } 4861 } 4862 lockEffectChains_l(effectChains); 4863 } 4864 4865 if (mActiveTrack != 0) { 4866 if (mActiveTrack->mState != TrackBase::ACTIVE && 4867 mActiveTrack->mState != TrackBase::RESUMING) { 4868 unlockEffectChains(effectChains); 4869 usleep(kRecordThreadSleepUs); 4870 continue; 4871 } 4872 for (size_t i = 0; i < effectChains.size(); i ++) { 4873 effectChains[i]->process_l(); 4874 } 4875 4876 buffer.frameCount = mFrameCount; 4877 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4878 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4879 size_t framesOut = buffer.frameCount; 4880 if (mResampler == NULL) { 4881 // no resampling 4882 while (framesOut) { 4883 size_t framesIn = mFrameCount - mRsmpInIndex; 4884 if (framesIn) { 4885 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4886 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4887 if (framesIn > framesOut) 4888 framesIn = framesOut; 4889 mRsmpInIndex += framesIn; 4890 framesOut -= framesIn; 4891 if ((int)mChannelCount == mReqChannelCount || 4892 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4893 memcpy(dst, src, framesIn * mFrameSize); 4894 } else { 4895 int16_t *src16 = (int16_t *)src; 4896 int16_t *dst16 = (int16_t *)dst; 4897 if (mChannelCount == 1) { 4898 while (framesIn--) { 4899 *dst16++ = *src16; 4900 *dst16++ = *src16++; 4901 } 4902 } else { 4903 while (framesIn--) { 4904 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4905 src16 += 2; 4906 } 4907 } 4908 } 4909 } 4910 if (framesOut && mFrameCount == mRsmpInIndex) { 4911 if (framesOut == mFrameCount && 4912 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4913 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4914 framesOut = 0; 4915 } else { 4916 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4917 mRsmpInIndex = 0; 4918 } 4919 if (mBytesRead < 0) { 4920 ALOGE("Error reading audio input"); 4921 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4922 // Force input into standby so that it tries to 4923 // recover at next read attempt 4924 mInput->stream->common.standby(&mInput->stream->common); 4925 usleep(kRecordThreadSleepUs); 4926 } 4927 mRsmpInIndex = mFrameCount; 4928 framesOut = 0; 4929 buffer.frameCount = 0; 4930 } 4931 } 4932 } 4933 } else { 4934 // resampling 4935 4936 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4937 // alter output frame count as if we were expecting stereo samples 4938 if (mChannelCount == 1 && mReqChannelCount == 1) { 4939 framesOut >>= 1; 4940 } 4941 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4942 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4943 // are 32 bit aligned which should be always true. 4944 if (mChannelCount == 2 && mReqChannelCount == 1) { 4945 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4946 // the resampler always outputs stereo samples: do post stereo to mono conversion 4947 int16_t *src = (int16_t *)mRsmpOutBuffer; 4948 int16_t *dst = buffer.i16; 4949 while (framesOut--) { 4950 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4951 src += 2; 4952 } 4953 } else { 4954 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4955 } 4956 4957 } 4958 mActiveTrack->releaseBuffer(&buffer); 4959 mActiveTrack->overflow(); 4960 } 4961 // client isn't retrieving buffers fast enough 4962 else { 4963 if (!mActiveTrack->setOverflow()) { 4964 nsecs_t now = systemTime(); 4965 if ((now - lastWarning) > kWarningThrottleNs) { 4966 ALOGW("RecordThread: buffer overflow"); 4967 lastWarning = now; 4968 } 4969 } 4970 // Release the processor for a while before asking for a new buffer. 4971 // This will give the application more chance to read from the buffer and 4972 // clear the overflow. 4973 usleep(kRecordThreadSleepUs); 4974 } 4975 } 4976 // enable changes in effect chain 4977 unlockEffectChains(effectChains); 4978 effectChains.clear(); 4979 } 4980 4981 if (!mStandby) { 4982 mInput->stream->common.standby(&mInput->stream->common); 4983 } 4984 mActiveTrack.clear(); 4985 4986 mStartStopCond.broadcast(); 4987 4988 releaseWakeLock(); 4989 4990 ALOGV("RecordThread %p exiting", this); 4991 return false; 4992} 4993 4994 4995sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4996 const sp<AudioFlinger::Client>& client, 4997 uint32_t sampleRate, 4998 audio_format_t format, 4999 int channelMask, 5000 int frameCount, 5001 uint32_t flags, 5002 int sessionId, 5003 status_t *status) 5004{ 5005 sp<RecordTrack> track; 5006 status_t lStatus; 5007 5008 lStatus = initCheck(); 5009 if (lStatus != NO_ERROR) { 5010 ALOGE("Audio driver not initialized."); 5011 goto Exit; 5012 } 5013 5014 { // scope for mLock 5015 Mutex::Autolock _l(mLock); 5016 5017 track = new RecordTrack(this, client, sampleRate, 5018 format, channelMask, frameCount, flags, sessionId); 5019 5020 if (track->getCblk() == 0) { 5021 lStatus = NO_MEMORY; 5022 goto Exit; 5023 } 5024 5025 mTrack = track.get(); 5026 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5027 bool suspend = audio_is_bluetooth_sco_device( 5028 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5029 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5030 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5031 } 5032 lStatus = NO_ERROR; 5033 5034Exit: 5035 if (status) { 5036 *status = lStatus; 5037 } 5038 return track; 5039} 5040 5041status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5042{ 5043 ALOGV("RecordThread::start tid=%d", tid); 5044 sp <ThreadBase> strongMe = this; 5045 status_t status = NO_ERROR; 5046 { 5047 AutoMutex lock(mLock); 5048 if (mActiveTrack != 0) { 5049 if (recordTrack != mActiveTrack.get()) { 5050 status = -EBUSY; 5051 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5052 mActiveTrack->mState = TrackBase::ACTIVE; 5053 } 5054 return status; 5055 } 5056 5057 recordTrack->mState = TrackBase::IDLE; 5058 mActiveTrack = recordTrack; 5059 mLock.unlock(); 5060 status_t status = AudioSystem::startInput(mId); 5061 mLock.lock(); 5062 if (status != NO_ERROR) { 5063 mActiveTrack.clear(); 5064 return status; 5065 } 5066 mRsmpInIndex = mFrameCount; 5067 mBytesRead = 0; 5068 if (mResampler != NULL) { 5069 mResampler->reset(); 5070 } 5071 mActiveTrack->mState = TrackBase::RESUMING; 5072 // signal thread to start 5073 ALOGV("Signal record thread"); 5074 mWaitWorkCV.signal(); 5075 // do not wait for mStartStopCond if exiting 5076 if (exitPending()) { 5077 mActiveTrack.clear(); 5078 status = INVALID_OPERATION; 5079 goto startError; 5080 } 5081 mStartStopCond.wait(mLock); 5082 if (mActiveTrack == 0) { 5083 ALOGV("Record failed to start"); 5084 status = BAD_VALUE; 5085 goto startError; 5086 } 5087 ALOGV("Record started OK"); 5088 return status; 5089 } 5090startError: 5091 AudioSystem::stopInput(mId); 5092 return status; 5093} 5094 5095void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5096 ALOGV("RecordThread::stop"); 5097 sp <ThreadBase> strongMe = this; 5098 { 5099 AutoMutex lock(mLock); 5100 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5101 mActiveTrack->mState = TrackBase::PAUSING; 5102 // do not wait for mStartStopCond if exiting 5103 if (exitPending()) { 5104 return; 5105 } 5106 mStartStopCond.wait(mLock); 5107 // if we have been restarted, recordTrack == mActiveTrack.get() here 5108 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5109 mLock.unlock(); 5110 AudioSystem::stopInput(mId); 5111 mLock.lock(); 5112 ALOGV("Record stopped OK"); 5113 } 5114 } 5115 } 5116} 5117 5118status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5119{ 5120 const size_t SIZE = 256; 5121 char buffer[SIZE]; 5122 String8 result; 5123 5124 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5125 result.append(buffer); 5126 5127 if (mActiveTrack != 0) { 5128 result.append("Active Track:\n"); 5129 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5130 mActiveTrack->dump(buffer, SIZE); 5131 result.append(buffer); 5132 5133 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5134 result.append(buffer); 5135 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5136 result.append(buffer); 5137 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5138 result.append(buffer); 5139 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5140 result.append(buffer); 5141 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5142 result.append(buffer); 5143 5144 5145 } else { 5146 result.append("No record client\n"); 5147 } 5148 write(fd, result.string(), result.size()); 5149 5150 dumpBase(fd, args); 5151 dumpEffectChains(fd, args); 5152 5153 return NO_ERROR; 5154} 5155 5156status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5157{ 5158 size_t framesReq = buffer->frameCount; 5159 size_t framesReady = mFrameCount - mRsmpInIndex; 5160 int channelCount; 5161 5162 if (framesReady == 0) { 5163 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5164 if (mBytesRead < 0) { 5165 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5166 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5167 // Force input into standby so that it tries to 5168 // recover at next read attempt 5169 mInput->stream->common.standby(&mInput->stream->common); 5170 usleep(kRecordThreadSleepUs); 5171 } 5172 buffer->raw = NULL; 5173 buffer->frameCount = 0; 5174 return NOT_ENOUGH_DATA; 5175 } 5176 mRsmpInIndex = 0; 5177 framesReady = mFrameCount; 5178 } 5179 5180 if (framesReq > framesReady) { 5181 framesReq = framesReady; 5182 } 5183 5184 if (mChannelCount == 1 && mReqChannelCount == 2) { 5185 channelCount = 1; 5186 } else { 5187 channelCount = 2; 5188 } 5189 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5190 buffer->frameCount = framesReq; 5191 return NO_ERROR; 5192} 5193 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 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5439 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5440 uint32_t channels = pChannels ? *pChannels : 0; 5441 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5442 audio_stream_out_t *outStream; 5443 audio_hw_device_t *outHwDev; 5444 5445 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5446 pDevices ? *pDevices : 0, 5447 samplingRate, 5448 format, 5449 channels, 5450 flags); 5451 5452 if (pDevices == NULL || *pDevices == 0) { 5453 return 0; 5454 } 5455 5456 Mutex::Autolock _l(mLock); 5457 5458 outHwDev = findSuitableHwDev_l(*pDevices); 5459 if (outHwDev == NULL) 5460 return 0; 5461 5462 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5463 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5464 &channels, &samplingRate, &outStream); 5465 mHardwareStatus = AUDIO_HW_IDLE; 5466 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5467 outStream, 5468 samplingRate, 5469 format, 5470 channels, 5471 status); 5472 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