AudioFlinger.cpp revision 5d4eeeaf76ebe177b43e87b2a9df5e55e39021f0
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_INIT; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_INIT; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 uint32_t flags, 436 const sp<IMemory>& sharedBuffer, 437 audio_io_handle_t output, 438 bool isTimed, 439 int *sessionId, 440 status_t *status) 441{ 442 sp<PlaybackThread::Track> track; 443 sp<TrackHandle> trackHandle; 444 sp<Client> client; 445 status_t lStatus; 446 int lSessionId; 447 448 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 449 // but if someone uses binder directly they could bypass that and cause us to crash 450 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 451 ALOGE("createTrack() invalid stream type %d", streamType); 452 lStatus = BAD_VALUE; 453 goto Exit; 454 } 455 456 { 457 Mutex::Autolock _l(mLock); 458 PlaybackThread *thread = checkPlaybackThread_l(output); 459 PlaybackThread *effectThread = NULL; 460 if (thread == NULL) { 461 ALOGE("unknown output thread"); 462 lStatus = BAD_VALUE; 463 goto Exit; 464 } 465 466 client = registerPid_l(pid); 467 468 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 469 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 470 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 471 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 472 if (mPlaybackThreads.keyAt(i) != output) { 473 // prevent same audio session on different output threads 474 uint32_t sessions = t->hasAudioSession(*sessionId); 475 if (sessions & PlaybackThread::TRACK_SESSION) { 476 ALOGE("createTrack() session ID %d already in use", *sessionId); 477 lStatus = BAD_VALUE; 478 goto Exit; 479 } 480 // check if an effect with same session ID is waiting for a track to be created 481 if (sessions & PlaybackThread::EFFECT_SESSION) { 482 effectThread = t.get(); 483 } 484 } 485 } 486 lSessionId = *sessionId; 487 } else { 488 // if no audio session id is provided, create one here 489 lSessionId = nextUniqueId(); 490 if (sessionId != NULL) { 491 *sessionId = lSessionId; 492 } 493 } 494 ALOGV("createTrack() lSessionId: %d", lSessionId); 495 496 track = thread->createTrack_l(client, streamType, sampleRate, format, 497 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 498 499 // move effect chain to this output thread if an effect on same session was waiting 500 // for a track to be created 501 if (lStatus == NO_ERROR && effectThread != NULL) { 502 Mutex::Autolock _dl(thread->mLock); 503 Mutex::Autolock _sl(effectThread->mLock); 504 moveEffectChain_l(lSessionId, effectThread, thread, true); 505 } 506 } 507 if (lStatus == NO_ERROR) { 508 trackHandle = new TrackHandle(track); 509 } else { 510 // remove local strong reference to Client before deleting the Track so that the Client 511 // destructor is called by the TrackBase destructor with mLock held 512 client.clear(); 513 track.clear(); 514 } 515 516Exit: 517 if(status) { 518 *status = lStatus; 519 } 520 return trackHandle; 521} 522 523uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 524{ 525 Mutex::Autolock _l(mLock); 526 PlaybackThread *thread = checkPlaybackThread_l(output); 527 if (thread == NULL) { 528 ALOGW("sampleRate() unknown thread %d", output); 529 return 0; 530 } 531 return thread->sampleRate(); 532} 533 534int AudioFlinger::channelCount(audio_io_handle_t output) const 535{ 536 Mutex::Autolock _l(mLock); 537 PlaybackThread *thread = checkPlaybackThread_l(output); 538 if (thread == NULL) { 539 ALOGW("channelCount() unknown thread %d", output); 540 return 0; 541 } 542 return thread->channelCount(); 543} 544 545audio_format_t AudioFlinger::format(audio_io_handle_t output) const 546{ 547 Mutex::Autolock _l(mLock); 548 PlaybackThread *thread = checkPlaybackThread_l(output); 549 if (thread == NULL) { 550 ALOGW("format() unknown thread %d", output); 551 return AUDIO_FORMAT_INVALID; 552 } 553 return thread->format(); 554} 555 556size_t AudioFlinger::frameCount(audio_io_handle_t output) const 557{ 558 Mutex::Autolock _l(mLock); 559 PlaybackThread *thread = checkPlaybackThread_l(output); 560 if (thread == NULL) { 561 ALOGW("frameCount() unknown thread %d", output); 562 return 0; 563 } 564 return thread->frameCount(); 565} 566 567uint32_t AudioFlinger::latency(audio_io_handle_t output) const 568{ 569 Mutex::Autolock _l(mLock); 570 PlaybackThread *thread = checkPlaybackThread_l(output); 571 if (thread == NULL) { 572 ALOGW("latency() unknown thread %d", output); 573 return 0; 574 } 575 return thread->latency(); 576} 577 578status_t AudioFlinger::setMasterVolume(float value) 579{ 580 status_t ret = initCheck(); 581 if (ret != NO_ERROR) { 582 return ret; 583 } 584 585 // check calling permissions 586 if (!settingsAllowed()) { 587 return PERMISSION_DENIED; 588 } 589 590 float swmv = value; 591 592 // when hw supports master volume, don't scale in sw mixer 593 if (MVS_NONE != mMasterVolumeSupportLvl) { 594 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 595 AutoMutex lock(mHardwareLock); 596 audio_hw_device_t *dev = mAudioHwDevs[i]; 597 598 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 599 if (NULL != dev->set_master_volume) { 600 dev->set_master_volume(dev, value); 601 } 602 mHardwareStatus = AUDIO_HW_IDLE; 603 } 604 605 swmv = 1.0; 606 } 607 608 Mutex::Autolock _l(mLock); 609 mMasterVolume = value; 610 mMasterVolumeSW = swmv; 611 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 612 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 613 614 return NO_ERROR; 615} 616 617status_t AudioFlinger::setMode(audio_mode_t mode) 618{ 619 status_t ret = initCheck(); 620 if (ret != NO_ERROR) { 621 return ret; 622 } 623 624 // check calling permissions 625 if (!settingsAllowed()) { 626 return PERMISSION_DENIED; 627 } 628 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 629 ALOGW("Illegal value: setMode(%d)", mode); 630 return BAD_VALUE; 631 } 632 633 { // scope for the lock 634 AutoMutex lock(mHardwareLock); 635 mHardwareStatus = AUDIO_HW_SET_MODE; 636 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 637 mHardwareStatus = AUDIO_HW_IDLE; 638 } 639 640 if (NO_ERROR == ret) { 641 Mutex::Autolock _l(mLock); 642 mMode = mode; 643 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 644 mPlaybackThreads.valueAt(i)->setMode(mode); 645 } 646 647 return ret; 648} 649 650status_t AudioFlinger::setMicMute(bool state) 651{ 652 status_t ret = initCheck(); 653 if (ret != NO_ERROR) { 654 return ret; 655 } 656 657 // check calling permissions 658 if (!settingsAllowed()) { 659 return PERMISSION_DENIED; 660 } 661 662 AutoMutex lock(mHardwareLock); 663 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 664 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 665 mHardwareStatus = AUDIO_HW_IDLE; 666 return ret; 667} 668 669bool AudioFlinger::getMicMute() const 670{ 671 status_t ret = initCheck(); 672 if (ret != NO_ERROR) { 673 return false; 674 } 675 676 bool state = AUDIO_MODE_INVALID; 677 AutoMutex lock(mHardwareLock); 678 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 679 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 680 mHardwareStatus = AUDIO_HW_IDLE; 681 return state; 682} 683 684status_t AudioFlinger::setMasterMute(bool muted) 685{ 686 // check calling permissions 687 if (!settingsAllowed()) { 688 return PERMISSION_DENIED; 689 } 690 691 Mutex::Autolock _l(mLock); 692 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 693 mMasterMute = muted; 694 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 695 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 696 697 return NO_ERROR; 698} 699 700float AudioFlinger::masterVolume() const 701{ 702 Mutex::Autolock _l(mLock); 703 return masterVolume_l(); 704} 705 706float AudioFlinger::masterVolumeSW() const 707{ 708 Mutex::Autolock _l(mLock); 709 return masterVolumeSW_l(); 710} 711 712bool AudioFlinger::masterMute() const 713{ 714 Mutex::Autolock _l(mLock); 715 return masterMute_l(); 716} 717 718float AudioFlinger::masterVolume_l() const 719{ 720 if (MVS_FULL == mMasterVolumeSupportLvl) { 721 float ret_val; 722 AutoMutex lock(mHardwareLock); 723 724 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 725 assert(NULL != mPrimaryHardwareDev); 726 assert(NULL != mPrimaryHardwareDev->get_master_volume); 727 728 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 729 mHardwareStatus = AUDIO_HW_IDLE; 730 return ret_val; 731 } 732 733 return mMasterVolume; 734} 735 736status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 737 audio_io_handle_t output) 738{ 739 // check calling permissions 740 if (!settingsAllowed()) { 741 return PERMISSION_DENIED; 742 } 743 744 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 745 ALOGE("setStreamVolume() invalid stream %d", stream); 746 return BAD_VALUE; 747 } 748 749 AutoMutex lock(mLock); 750 PlaybackThread *thread = NULL; 751 if (output) { 752 thread = checkPlaybackThread_l(output); 753 if (thread == NULL) { 754 return BAD_VALUE; 755 } 756 } 757 758 mStreamTypes[stream].volume = value; 759 760 if (thread == NULL) { 761 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 762 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 763 } 764 } else { 765 thread->setStreamVolume(stream, value); 766 } 767 768 return NO_ERROR; 769} 770 771status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 772{ 773 // check calling permissions 774 if (!settingsAllowed()) { 775 return PERMISSION_DENIED; 776 } 777 778 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 779 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 780 ALOGE("setStreamMute() invalid stream %d", stream); 781 return BAD_VALUE; 782 } 783 784 AutoMutex lock(mLock); 785 mStreamTypes[stream].mute = muted; 786 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 787 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 788 789 return NO_ERROR; 790} 791 792float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 793{ 794 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 795 return 0.0f; 796 } 797 798 AutoMutex lock(mLock); 799 float volume; 800 if (output) { 801 PlaybackThread *thread = checkPlaybackThread_l(output); 802 if (thread == NULL) { 803 return 0.0f; 804 } 805 volume = thread->streamVolume(stream); 806 } else { 807 volume = streamVolume_l(stream); 808 } 809 810 return volume; 811} 812 813bool AudioFlinger::streamMute(audio_stream_type_t stream) const 814{ 815 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 816 return true; 817 } 818 819 AutoMutex lock(mLock); 820 return streamMute_l(stream); 821} 822 823status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 824{ 825 status_t result; 826 827 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 828 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 829 // check calling permissions 830 if (!settingsAllowed()) { 831 return PERMISSION_DENIED; 832 } 833 834 // ioHandle == 0 means the parameters are global to the audio hardware interface 835 if (ioHandle == 0) { 836 AutoMutex lock(mHardwareLock); 837 mHardwareStatus = AUDIO_SET_PARAMETER; 838 status_t final_result = NO_ERROR; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs[i]; 841 result = dev->set_parameters(dev, keyValuePairs.string()); 842 final_result = result ?: final_result; 843 } 844 mHardwareStatus = AUDIO_HW_IDLE; 845 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 846 AudioParameter param = AudioParameter(keyValuePairs); 847 String8 value; 848 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 849 Mutex::Autolock _l(mLock); 850 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 851 if (mBtNrecIsOff != btNrecIsOff) { 852 for (size_t i = 0; i < mRecordThreads.size(); i++) { 853 sp<RecordThread> thread = mRecordThreads.valueAt(i); 854 RecordThread::RecordTrack *track = thread->track(); 855 if (track != NULL) { 856 audio_devices_t device = (audio_devices_t)( 857 thread->device() & AUDIO_DEVICE_IN_ALL); 858 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 859 thread->setEffectSuspended(FX_IID_AEC, 860 suspend, 861 track->sessionId()); 862 thread->setEffectSuspended(FX_IID_NS, 863 suspend, 864 track->sessionId()); 865 } 866 } 867 mBtNrecIsOff = btNrecIsOff; 868 } 869 } 870 return final_result; 871 } 872 873 // hold a strong ref on thread in case closeOutput() or closeInput() is called 874 // and the thread is exited once the lock is released 875 sp<ThreadBase> thread; 876 { 877 Mutex::Autolock _l(mLock); 878 thread = checkPlaybackThread_l(ioHandle); 879 if (thread == NULL) { 880 thread = checkRecordThread_l(ioHandle); 881 } else if (thread == primaryPlaybackThread_l()) { 882 // indicate output device change to all input threads for pre processing 883 AudioParameter param = AudioParameter(keyValuePairs); 884 int value; 885 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 886 for (size_t i = 0; i < mRecordThreads.size(); i++) { 887 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 888 } 889 } 890 } 891 } 892 if (thread != 0) { 893 return thread->setParameters(keyValuePairs); 894 } 895 return BAD_VALUE; 896} 897 898String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 899{ 900// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 901// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 902 903 if (ioHandle == 0) { 904 String8 out_s8; 905 906 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 907 audio_hw_device_t *dev = mAudioHwDevs[i]; 908 char *s = dev->get_parameters(dev, keys.string()); 909 out_s8 += String8(s ? s : ""); 910 free(s); 911 } 912 return out_s8; 913 } 914 915 Mutex::Autolock _l(mLock); 916 917 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 918 if (playbackThread != NULL) { 919 return playbackThread->getParameters(keys); 920 } 921 RecordThread *recordThread = checkRecordThread_l(ioHandle); 922 if (recordThread != NULL) { 923 return recordThread->getParameters(keys); 924 } 925 return String8(""); 926} 927 928size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 929{ 930 status_t ret = initCheck(); 931 if (ret != NO_ERROR) { 932 return 0; 933 } 934 935 AutoMutex lock(mHardwareLock); 936 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 937 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 938 mHardwareStatus = AUDIO_HW_IDLE; 939 return size; 940} 941 942unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 943{ 944 if (ioHandle == 0) { 945 return 0; 946 } 947 948 Mutex::Autolock _l(mLock); 949 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getInputFramesLost(); 953 } 954 return 0; 955} 956 957status_t AudioFlinger::setVoiceVolume(float value) 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return ret; 962 } 963 964 // check calling permissions 965 if (!settingsAllowed()) { 966 return PERMISSION_DENIED; 967 } 968 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 971 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 972 mHardwareStatus = AUDIO_HW_IDLE; 973 974 return ret; 975} 976 977status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 978 audio_io_handle_t output) const 979{ 980 status_t status; 981 982 Mutex::Autolock _l(mLock); 983 984 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 985 if (playbackThread != NULL) { 986 return playbackThread->getRenderPosition(halFrames, dspFrames); 987 } 988 989 return BAD_VALUE; 990} 991 992void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 993{ 994 995 Mutex::Autolock _l(mLock); 996 997 pid_t pid = IPCThreadState::self()->getCallingPid(); 998 if (mNotificationClients.indexOfKey(pid) < 0) { 999 sp<NotificationClient> notificationClient = new NotificationClient(this, 1000 client, 1001 pid); 1002 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1003 1004 mNotificationClients.add(pid, notificationClient); 1005 1006 sp<IBinder> binder = client->asBinder(); 1007 binder->linkToDeath(notificationClient); 1008 1009 // the config change is always sent from playback or record threads to avoid deadlock 1010 // with AudioSystem::gLock 1011 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1012 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1013 } 1014 1015 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1016 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1017 } 1018 } 1019} 1020 1021void AudioFlinger::removeNotificationClient(pid_t pid) 1022{ 1023 Mutex::Autolock _l(mLock); 1024 1025 ssize_t index = mNotificationClients.indexOfKey(pid); 1026 if (index >= 0) { 1027 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 1028 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 1029 mNotificationClients.removeItem(pid); 1030 } 1031 1032 ALOGV("%d died, releasing its sessions", pid); 1033 size_t num = mAudioSessionRefs.size(); 1034 bool removed = false; 1035 for (size_t i = 0; i< num; ) { 1036 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1037 ALOGV(" pid %d @ %d", ref->pid, i); 1038 if (ref->pid == pid) { 1039 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1040 mAudioSessionRefs.removeAt(i); 1041 delete ref; 1042 removed = true; 1043 num--; 1044 } else { 1045 i++; 1046 } 1047 } 1048 if (removed) { 1049 purgeStaleEffects_l(); 1050 } 1051} 1052 1053// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1054void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1055{ 1056 size_t size = mNotificationClients.size(); 1057 for (size_t i = 0; i < size; i++) { 1058 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1059 param2); 1060 } 1061} 1062 1063// removeClient_l() must be called with AudioFlinger::mLock held 1064void AudioFlinger::removeClient_l(pid_t pid) 1065{ 1066 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1067 mClients.removeItem(pid); 1068} 1069 1070 1071// ---------------------------------------------------------------------------- 1072 1073AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1074 uint32_t device, type_t type) 1075 : Thread(false), 1076 mType(type), 1077 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1078 // mChannelMask 1079 mChannelCount(0), 1080 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1081 mParamStatus(NO_ERROR), 1082 mStandby(false), mId(id), 1083 mDevice(device), 1084 mDeathRecipient(new PMDeathRecipient(this)) 1085{ 1086} 1087 1088AudioFlinger::ThreadBase::~ThreadBase() 1089{ 1090 mParamCond.broadcast(); 1091 // do not lock the mutex in destructor 1092 releaseWakeLock_l(); 1093 if (mPowerManager != 0) { 1094 sp<IBinder> binder = mPowerManager->asBinder(); 1095 binder->unlinkToDeath(mDeathRecipient); 1096 } 1097} 1098 1099void AudioFlinger::ThreadBase::exit() 1100{ 1101 ALOGV("ThreadBase::exit"); 1102 { 1103 // This lock prevents the following race in thread (uniprocessor for illustration): 1104 // if (!exitPending()) { 1105 // // context switch from here to exit() 1106 // // exit() calls requestExit(), what exitPending() observes 1107 // // exit() calls signal(), which is dropped since no waiters 1108 // // context switch back from exit() to here 1109 // mWaitWorkCV.wait(...); 1110 // // now thread is hung 1111 // } 1112 AutoMutex lock(mLock); 1113 requestExit(); 1114 mWaitWorkCV.signal(); 1115 } 1116 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1117 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1118 requestExitAndWait(); 1119} 1120 1121status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1122{ 1123 status_t status; 1124 1125 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1126 Mutex::Autolock _l(mLock); 1127 1128 mNewParameters.add(keyValuePairs); 1129 mWaitWorkCV.signal(); 1130 // wait condition with timeout in case the thread loop has exited 1131 // before the request could be processed 1132 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1133 status = mParamStatus; 1134 mWaitWorkCV.signal(); 1135 } else { 1136 status = TIMED_OUT; 1137 } 1138 return status; 1139} 1140 1141void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1142{ 1143 Mutex::Autolock _l(mLock); 1144 sendConfigEvent_l(event, param); 1145} 1146 1147// sendConfigEvent_l() must be called with ThreadBase::mLock held 1148void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1149{ 1150 ConfigEvent configEvent; 1151 configEvent.mEvent = event; 1152 configEvent.mParam = param; 1153 mConfigEvents.add(configEvent); 1154 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1155 mWaitWorkCV.signal(); 1156} 1157 1158void AudioFlinger::ThreadBase::processConfigEvents() 1159{ 1160 mLock.lock(); 1161 while(!mConfigEvents.isEmpty()) { 1162 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1163 ConfigEvent configEvent = mConfigEvents[0]; 1164 mConfigEvents.removeAt(0); 1165 // release mLock before locking AudioFlinger mLock: lock order is always 1166 // AudioFlinger then ThreadBase to avoid cross deadlock 1167 mLock.unlock(); 1168 mAudioFlinger->mLock.lock(); 1169 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1170 mAudioFlinger->mLock.unlock(); 1171 mLock.lock(); 1172 } 1173 mLock.unlock(); 1174} 1175 1176status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1177{ 1178 const size_t SIZE = 256; 1179 char buffer[SIZE]; 1180 String8 result; 1181 1182 bool locked = tryLock(mLock); 1183 if (!locked) { 1184 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1185 write(fd, buffer, strlen(buffer)); 1186 } 1187 1188 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1189 result.append(buffer); 1190 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1191 result.append(buffer); 1192 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1193 result.append(buffer); 1194 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1197 result.append(buffer); 1198 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1199 result.append(buffer); 1200 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1201 result.append(buffer); 1202 1203 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1204 result.append(buffer); 1205 result.append(" Index Command"); 1206 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1207 snprintf(buffer, SIZE, "\n %02d ", i); 1208 result.append(buffer); 1209 result.append(mNewParameters[i]); 1210 } 1211 1212 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, " Index event param\n"); 1215 result.append(buffer); 1216 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1217 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1218 result.append(buffer); 1219 } 1220 result.append("\n"); 1221 1222 write(fd, result.string(), result.size()); 1223 1224 if (locked) { 1225 mLock.unlock(); 1226 } 1227 return NO_ERROR; 1228} 1229 1230status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1231{ 1232 const size_t SIZE = 256; 1233 char buffer[SIZE]; 1234 String8 result; 1235 1236 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1237 write(fd, buffer, strlen(buffer)); 1238 1239 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1240 sp<EffectChain> chain = mEffectChains[i]; 1241 if (chain != 0) { 1242 chain->dump(fd, args); 1243 } 1244 } 1245 return NO_ERROR; 1246} 1247 1248void AudioFlinger::ThreadBase::acquireWakeLock() 1249{ 1250 Mutex::Autolock _l(mLock); 1251 acquireWakeLock_l(); 1252} 1253 1254void AudioFlinger::ThreadBase::acquireWakeLock_l() 1255{ 1256 if (mPowerManager == 0) { 1257 // use checkService() to avoid blocking if power service is not up yet 1258 sp<IBinder> binder = 1259 defaultServiceManager()->checkService(String16("power")); 1260 if (binder == 0) { 1261 ALOGW("Thread %s cannot connect to the power manager service", mName); 1262 } else { 1263 mPowerManager = interface_cast<IPowerManager>(binder); 1264 binder->linkToDeath(mDeathRecipient); 1265 } 1266 } 1267 if (mPowerManager != 0) { 1268 sp<IBinder> binder = new BBinder(); 1269 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1270 binder, 1271 String16(mName)); 1272 if (status == NO_ERROR) { 1273 mWakeLockToken = binder; 1274 } 1275 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1276 } 1277} 1278 1279void AudioFlinger::ThreadBase::releaseWakeLock() 1280{ 1281 Mutex::Autolock _l(mLock); 1282 releaseWakeLock_l(); 1283} 1284 1285void AudioFlinger::ThreadBase::releaseWakeLock_l() 1286{ 1287 if (mWakeLockToken != 0) { 1288 ALOGV("releaseWakeLock_l() %s", mName); 1289 if (mPowerManager != 0) { 1290 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1291 } 1292 mWakeLockToken.clear(); 1293 } 1294} 1295 1296void AudioFlinger::ThreadBase::clearPowerManager() 1297{ 1298 Mutex::Autolock _l(mLock); 1299 releaseWakeLock_l(); 1300 mPowerManager.clear(); 1301} 1302 1303void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1304{ 1305 sp<ThreadBase> thread = mThread.promote(); 1306 if (thread != 0) { 1307 thread->clearPowerManager(); 1308 } 1309 ALOGW("power manager service died !!!"); 1310} 1311 1312void AudioFlinger::ThreadBase::setEffectSuspended( 1313 const effect_uuid_t *type, bool suspend, int sessionId) 1314{ 1315 Mutex::Autolock _l(mLock); 1316 setEffectSuspended_l(type, suspend, sessionId); 1317} 1318 1319void AudioFlinger::ThreadBase::setEffectSuspended_l( 1320 const effect_uuid_t *type, bool suspend, int sessionId) 1321{ 1322 sp<EffectChain> chain = getEffectChain_l(sessionId); 1323 if (chain != 0) { 1324 if (type != NULL) { 1325 chain->setEffectSuspended_l(type, suspend); 1326 } else { 1327 chain->setEffectSuspendedAll_l(suspend); 1328 } 1329 } 1330 1331 updateSuspendedSessions_l(type, suspend, sessionId); 1332} 1333 1334void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1335{ 1336 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1337 if (index < 0) { 1338 return; 1339 } 1340 1341 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1342 mSuspendedSessions.editValueAt(index); 1343 1344 for (size_t i = 0; i < sessionEffects.size(); i++) { 1345 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1346 for (int j = 0; j < desc->mRefCount; j++) { 1347 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1348 chain->setEffectSuspendedAll_l(true); 1349 } else { 1350 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1351 desc->mType.timeLow); 1352 chain->setEffectSuspended_l(&desc->mType, true); 1353 } 1354 } 1355 } 1356} 1357 1358void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1359 bool suspend, 1360 int sessionId) 1361{ 1362 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1363 1364 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1365 1366 if (suspend) { 1367 if (index >= 0) { 1368 sessionEffects = mSuspendedSessions.editValueAt(index); 1369 } else { 1370 mSuspendedSessions.add(sessionId, sessionEffects); 1371 } 1372 } else { 1373 if (index < 0) { 1374 return; 1375 } 1376 sessionEffects = mSuspendedSessions.editValueAt(index); 1377 } 1378 1379 1380 int key = EffectChain::kKeyForSuspendAll; 1381 if (type != NULL) { 1382 key = type->timeLow; 1383 } 1384 index = sessionEffects.indexOfKey(key); 1385 1386 sp <SuspendedSessionDesc> desc; 1387 if (suspend) { 1388 if (index >= 0) { 1389 desc = sessionEffects.valueAt(index); 1390 } else { 1391 desc = new SuspendedSessionDesc(); 1392 if (type != NULL) { 1393 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1394 } 1395 sessionEffects.add(key, desc); 1396 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1397 } 1398 desc->mRefCount++; 1399 } else { 1400 if (index < 0) { 1401 return; 1402 } 1403 desc = sessionEffects.valueAt(index); 1404 if (--desc->mRefCount == 0) { 1405 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1406 sessionEffects.removeItemsAt(index); 1407 if (sessionEffects.isEmpty()) { 1408 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1409 sessionId); 1410 mSuspendedSessions.removeItem(sessionId); 1411 } 1412 } 1413 } 1414 if (!sessionEffects.isEmpty()) { 1415 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1416 } 1417} 1418 1419void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1420 bool enabled, 1421 int sessionId) 1422{ 1423 Mutex::Autolock _l(mLock); 1424 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1425} 1426 1427void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1428 bool enabled, 1429 int sessionId) 1430{ 1431 if (mType != RECORD) { 1432 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1433 // another session. This gives the priority to well behaved effect control panels 1434 // and applications not using global effects. 1435 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1436 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1437 } 1438 } 1439 1440 sp<EffectChain> chain = getEffectChain_l(sessionId); 1441 if (chain != 0) { 1442 chain->checkSuspendOnEffectEnabled(effect, enabled); 1443 } 1444} 1445 1446// ---------------------------------------------------------------------------- 1447 1448AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1449 AudioStreamOut* output, 1450 audio_io_handle_t id, 1451 uint32_t device, 1452 type_t type) 1453 : ThreadBase(audioFlinger, id, device, type), 1454 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1455 // Assumes constructor is called by AudioFlinger with it's mLock held, 1456 // but it would be safer to explicitly pass initial masterMute as parameter 1457 mMasterMute(audioFlinger->masterMute_l()), 1458 // mStreamTypes[] initialized in constructor body 1459 mOutput(output), 1460 // Assumes constructor is called by AudioFlinger with it's mLock held, 1461 // but it would be safer to explicitly pass initial masterVolume as parameter 1462 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1463 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1464{ 1465 snprintf(mName, kNameLength, "AudioOut_%d", id); 1466 1467 readOutputParameters(); 1468 1469 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1470 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1471 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1472 stream = (audio_stream_type_t) (stream + 1)) { 1473 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1474 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1475 // initialized by stream_type_t default constructor 1476 // mStreamTypes[stream].valid = true; 1477 } 1478 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1479 // because mAudioFlinger doesn't have one to copy from 1480} 1481 1482AudioFlinger::PlaybackThread::~PlaybackThread() 1483{ 1484 delete [] mMixBuffer; 1485} 1486 1487status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1488{ 1489 dumpInternals(fd, args); 1490 dumpTracks(fd, args); 1491 dumpEffectChains(fd, args); 1492 return NO_ERROR; 1493} 1494 1495status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1496{ 1497 const size_t SIZE = 256; 1498 char buffer[SIZE]; 1499 String8 result; 1500 1501 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1502 result.append(buffer); 1503 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1504 for (size_t i = 0; i < mTracks.size(); ++i) { 1505 sp<Track> track = mTracks[i]; 1506 if (track != 0) { 1507 track->dump(buffer, SIZE); 1508 result.append(buffer); 1509 } 1510 } 1511 1512 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1513 result.append(buffer); 1514 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1515 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1516 sp<Track> track = mActiveTracks[i].promote(); 1517 if (track != 0) { 1518 track->dump(buffer, SIZE); 1519 result.append(buffer); 1520 } 1521 } 1522 write(fd, result.string(), result.size()); 1523 return NO_ERROR; 1524} 1525 1526status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1527{ 1528 const size_t SIZE = 256; 1529 char buffer[SIZE]; 1530 String8 result; 1531 1532 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1533 result.append(buffer); 1534 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1535 result.append(buffer); 1536 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1537 result.append(buffer); 1538 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1541 result.append(buffer); 1542 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1545 result.append(buffer); 1546 write(fd, result.string(), result.size()); 1547 1548 dumpBase(fd, args); 1549 1550 return NO_ERROR; 1551} 1552 1553// Thread virtuals 1554status_t AudioFlinger::PlaybackThread::readyToRun() 1555{ 1556 status_t status = initCheck(); 1557 if (status == NO_ERROR) { 1558 ALOGI("AudioFlinger's thread %p ready to run", this); 1559 } else { 1560 ALOGE("No working audio driver found."); 1561 } 1562 return status; 1563} 1564 1565void AudioFlinger::PlaybackThread::onFirstRef() 1566{ 1567 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1568} 1569 1570// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1571sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1572 const sp<AudioFlinger::Client>& client, 1573 audio_stream_type_t streamType, 1574 uint32_t sampleRate, 1575 audio_format_t format, 1576 uint32_t channelMask, 1577 int frameCount, 1578 const sp<IMemory>& sharedBuffer, 1579 int sessionId, 1580 bool isTimed, 1581 status_t *status) 1582{ 1583 sp<Track> track; 1584 status_t lStatus; 1585 1586 if (mType == DIRECT) { 1587 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1588 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1589 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1590 "for output %p with format %d", 1591 sampleRate, format, channelMask, mOutput, mFormat); 1592 lStatus = BAD_VALUE; 1593 goto Exit; 1594 } 1595 } 1596 } else { 1597 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1598 if (sampleRate > mSampleRate*2) { 1599 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1600 lStatus = BAD_VALUE; 1601 goto Exit; 1602 } 1603 } 1604 1605 lStatus = initCheck(); 1606 if (lStatus != NO_ERROR) { 1607 ALOGE("Audio driver not initialized."); 1608 goto Exit; 1609 } 1610 1611 { // scope for mLock 1612 Mutex::Autolock _l(mLock); 1613 1614 // all tracks in same audio session must share the same routing strategy otherwise 1615 // conflicts will happen when tracks are moved from one output to another by audio policy 1616 // manager 1617 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1618 for (size_t i = 0; i < mTracks.size(); ++i) { 1619 sp<Track> t = mTracks[i]; 1620 if (t != 0) { 1621 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1622 if (sessionId == t->sessionId() && strategy != actual) { 1623 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1624 strategy, actual); 1625 lStatus = BAD_VALUE; 1626 goto Exit; 1627 } 1628 } 1629 } 1630 1631 if (!isTimed) { 1632 track = new Track(this, client, streamType, sampleRate, format, 1633 channelMask, frameCount, sharedBuffer, sessionId); 1634 } else { 1635 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1636 channelMask, frameCount, sharedBuffer, sessionId); 1637 } 1638 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1639 lStatus = NO_MEMORY; 1640 goto Exit; 1641 } 1642 mTracks.add(track); 1643 1644 sp<EffectChain> chain = getEffectChain_l(sessionId); 1645 if (chain != 0) { 1646 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1647 track->setMainBuffer(chain->inBuffer()); 1648 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1649 chain->incTrackCnt(); 1650 } 1651 1652 // invalidate track immediately if the stream type was moved to another thread since 1653 // createTrack() was called by the client process. 1654 if (!mStreamTypes[streamType].valid) { 1655 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1656 this, streamType); 1657 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1658 } 1659 } 1660 lStatus = NO_ERROR; 1661 1662Exit: 1663 if(status) { 1664 *status = lStatus; 1665 } 1666 return track; 1667} 1668 1669uint32_t AudioFlinger::PlaybackThread::latency() const 1670{ 1671 Mutex::Autolock _l(mLock); 1672 if (initCheck() == NO_ERROR) { 1673 return mOutput->stream->get_latency(mOutput->stream); 1674 } else { 1675 return 0; 1676 } 1677} 1678 1679void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1680{ 1681 Mutex::Autolock _l(mLock); 1682 mMasterVolume = value; 1683} 1684 1685void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1686{ 1687 Mutex::Autolock _l(mLock); 1688 setMasterMute_l(muted); 1689} 1690 1691void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1692{ 1693 Mutex::Autolock _l(mLock); 1694 mStreamTypes[stream].volume = value; 1695} 1696 1697void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1698{ 1699 Mutex::Autolock _l(mLock); 1700 mStreamTypes[stream].mute = muted; 1701} 1702 1703float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1704{ 1705 Mutex::Autolock _l(mLock); 1706 return mStreamTypes[stream].volume; 1707} 1708 1709// addTrack_l() must be called with ThreadBase::mLock held 1710status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1711{ 1712 status_t status = ALREADY_EXISTS; 1713 1714 // set retry count for buffer fill 1715 track->mRetryCount = kMaxTrackStartupRetries; 1716 if (mActiveTracks.indexOf(track) < 0) { 1717 // the track is newly added, make sure it fills up all its 1718 // buffers before playing. This is to ensure the client will 1719 // effectively get the latency it requested. 1720 track->mFillingUpStatus = Track::FS_FILLING; 1721 track->mResetDone = false; 1722 mActiveTracks.add(track); 1723 if (track->mainBuffer() != mMixBuffer) { 1724 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1725 if (chain != 0) { 1726 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1727 chain->incActiveTrackCnt(); 1728 } 1729 } 1730 1731 status = NO_ERROR; 1732 } 1733 1734 ALOGV("mWaitWorkCV.broadcast"); 1735 mWaitWorkCV.broadcast(); 1736 1737 return status; 1738} 1739 1740// destroyTrack_l() must be called with ThreadBase::mLock held 1741void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1742{ 1743 track->mState = TrackBase::TERMINATED; 1744 if (mActiveTracks.indexOf(track) < 0) { 1745 removeTrack_l(track); 1746 } 1747} 1748 1749void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1750{ 1751 mTracks.remove(track); 1752 deleteTrackName_l(track->name()); 1753 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1754 if (chain != 0) { 1755 chain->decTrackCnt(); 1756 } 1757} 1758 1759String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1760{ 1761 String8 out_s8 = String8(""); 1762 char *s; 1763 1764 Mutex::Autolock _l(mLock); 1765 if (initCheck() != NO_ERROR) { 1766 return out_s8; 1767 } 1768 1769 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1770 out_s8 = String8(s); 1771 free(s); 1772 return out_s8; 1773} 1774 1775// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1776void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1777 AudioSystem::OutputDescriptor desc; 1778 void *param2 = NULL; 1779 1780 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1781 1782 switch (event) { 1783 case AudioSystem::OUTPUT_OPENED: 1784 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1785 desc.channels = mChannelMask; 1786 desc.samplingRate = mSampleRate; 1787 desc.format = mFormat; 1788 desc.frameCount = mFrameCount; 1789 desc.latency = latency(); 1790 param2 = &desc; 1791 break; 1792 1793 case AudioSystem::STREAM_CONFIG_CHANGED: 1794 param2 = ¶m; 1795 case AudioSystem::OUTPUT_CLOSED: 1796 default: 1797 break; 1798 } 1799 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1800} 1801 1802void AudioFlinger::PlaybackThread::readOutputParameters() 1803{ 1804 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1805 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1806 mChannelCount = (uint16_t)popcount(mChannelMask); 1807 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1808 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1809 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1810 1811 // FIXME - Current mixer implementation only supports stereo output: Always 1812 // Allocate a stereo buffer even if HW output is mono. 1813 delete[] mMixBuffer; 1814 mMixBuffer = new int16_t[mFrameCount * 2]; 1815 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1816 1817 // force reconfiguration of effect chains and engines to take new buffer size and audio 1818 // parameters into account 1819 // Note that mLock is not held when readOutputParameters() is called from the constructor 1820 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1821 // matter. 1822 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1823 Vector< sp<EffectChain> > effectChains = mEffectChains; 1824 for (size_t i = 0; i < effectChains.size(); i ++) { 1825 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1826 } 1827} 1828 1829status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1830{ 1831 if (halFrames == NULL || dspFrames == NULL) { 1832 return BAD_VALUE; 1833 } 1834 Mutex::Autolock _l(mLock); 1835 if (initCheck() != NO_ERROR) { 1836 return INVALID_OPERATION; 1837 } 1838 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1839 1840 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1841} 1842 1843uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1844{ 1845 Mutex::Autolock _l(mLock); 1846 uint32_t result = 0; 1847 if (getEffectChain_l(sessionId) != 0) { 1848 result = EFFECT_SESSION; 1849 } 1850 1851 for (size_t i = 0; i < mTracks.size(); ++i) { 1852 sp<Track> track = mTracks[i]; 1853 if (sessionId == track->sessionId() && 1854 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1855 result |= TRACK_SESSION; 1856 break; 1857 } 1858 } 1859 1860 return result; 1861} 1862 1863uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1864{ 1865 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1866 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1867 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1868 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1869 } 1870 for (size_t i = 0; i < mTracks.size(); i++) { 1871 sp<Track> track = mTracks[i]; 1872 if (sessionId == track->sessionId() && 1873 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1874 return AudioSystem::getStrategyForStream(track->streamType()); 1875 } 1876 } 1877 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1878} 1879 1880 1881AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1882{ 1883 Mutex::Autolock _l(mLock); 1884 return mOutput; 1885} 1886 1887AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1888{ 1889 Mutex::Autolock _l(mLock); 1890 AudioStreamOut *output = mOutput; 1891 mOutput = NULL; 1892 return output; 1893} 1894 1895// this method must always be called either with ThreadBase mLock held or inside the thread loop 1896audio_stream_t* AudioFlinger::PlaybackThread::stream() 1897{ 1898 if (mOutput == NULL) { 1899 return NULL; 1900 } 1901 return &mOutput->stream->common; 1902} 1903 1904uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1905{ 1906 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1907 // decoding and transfer time. So sleeping for half of the latency would likely cause 1908 // underruns 1909 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1910 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1911 } else { 1912 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1913 } 1914} 1915 1916// ---------------------------------------------------------------------------- 1917 1918AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1919 audio_io_handle_t id, uint32_t device, type_t type) 1920 : PlaybackThread(audioFlinger, output, id, device, type), 1921 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1922 mPrevMixerStatus(MIXER_IDLE) 1923{ 1924 // FIXME - Current mixer implementation only supports stereo output 1925 if (mChannelCount == 1) { 1926 ALOGE("Invalid audio hardware channel count"); 1927 } 1928} 1929 1930AudioFlinger::MixerThread::~MixerThread() 1931{ 1932 delete mAudioMixer; 1933} 1934 1935bool AudioFlinger::MixerThread::threadLoop() 1936{ 1937 Vector< sp<Track> > tracksToRemove; 1938 nsecs_t standbyTime = systemTime(); 1939 size_t mixBufferSize = mFrameCount * mFrameSize; 1940 // FIXME: Relaxed timing because of a certain device that can't meet latency 1941 // Should be reduced to 2x after the vendor fixes the driver issue 1942 // increase threshold again due to low power audio mode. The way this warning threshold is 1943 // calculated and its usefulness should be reconsidered anyway. 1944 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1945 nsecs_t lastWarning = 0; 1946 bool longStandbyExit = false; 1947 uint32_t activeSleepTime = activeSleepTimeUs(); 1948 uint32_t idleSleepTime = idleSleepTimeUs(); 1949 uint32_t sleepTime = idleSleepTime; 1950 uint32_t sleepTimeShift = 0; 1951 Vector< sp<EffectChain> > effectChains; 1952#ifdef DEBUG_CPU_USAGE 1953 ThreadCpuUsage cpu; 1954 const CentralTendencyStatistics& stats = cpu.statistics(); 1955#endif 1956 1957 acquireWakeLock(); 1958 1959 while (!exitPending()) 1960 { 1961#ifdef DEBUG_CPU_USAGE 1962 cpu.sampleAndEnable(); 1963 unsigned n = stats.n(); 1964 // cpu.elapsed() is expensive, so don't call it every loop 1965 if ((n & 127) == 1) { 1966 long long elapsed = cpu.elapsed(); 1967 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1968 double perLoop = elapsed / (double) n; 1969 double perLoop100 = perLoop * 0.01; 1970 double mean = stats.mean(); 1971 double stddev = stats.stddev(); 1972 double minimum = stats.minimum(); 1973 double maximum = stats.maximum(); 1974 cpu.resetStatistics(); 1975 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", 1976 elapsed * .000000001, n, perLoop * .000001, 1977 mean * .001, 1978 stddev * .001, 1979 minimum * .001, 1980 maximum * .001, 1981 mean / perLoop100, 1982 stddev / perLoop100, 1983 minimum / perLoop100, 1984 maximum / perLoop100); 1985 } 1986 } 1987#endif 1988 processConfigEvents(); 1989 1990 mixer_state mixerStatus = MIXER_IDLE; 1991 { // scope for mLock 1992 1993 Mutex::Autolock _l(mLock); 1994 1995 if (checkForNewParameters_l()) { 1996 mixBufferSize = mFrameCount * mFrameSize; 1997 // FIXME: Relaxed timing because of a certain device that can't meet latency 1998 // Should be reduced to 2x after the vendor fixes the driver issue 1999 // increase threshold again due to low power audio mode. The way this warning 2000 // threshold is calculated and its usefulness should be reconsidered anyway. 2001 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2002 activeSleepTime = activeSleepTimeUs(); 2003 idleSleepTime = idleSleepTimeUs(); 2004 } 2005 2006 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2007 2008 // put audio hardware into standby after short delay 2009 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2010 mSuspended)) { 2011 if (!mStandby) { 2012 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2013 mOutput->stream->common.standby(&mOutput->stream->common); 2014 mStandby = true; 2015 mBytesWritten = 0; 2016 } 2017 2018 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2019 // we're about to wait, flush the binder command buffer 2020 IPCThreadState::self()->flushCommands(); 2021 2022 if (exitPending()) break; 2023 2024 releaseWakeLock_l(); 2025 // wait until we have something to do... 2026 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 2027 mWaitWorkCV.wait(mLock); 2028 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 2029 acquireWakeLock_l(); 2030 2031 mPrevMixerStatus = MIXER_IDLE; 2032 if (!mMasterMute) { 2033 char value[PROPERTY_VALUE_MAX]; 2034 property_get("ro.audio.silent", value, "0"); 2035 if (atoi(value)) { 2036 ALOGD("Silence is golden"); 2037 setMasterMute_l(true); 2038 } 2039 } 2040 2041 standbyTime = systemTime() + mStandbyTimeInNsecs; 2042 sleepTime = idleSleepTime; 2043 sleepTimeShift = 0; 2044 continue; 2045 } 2046 } 2047 2048 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2049 2050 // prevent any changes in effect chain list and in each effect chain 2051 // during mixing and effect process as the audio buffers could be deleted 2052 // or modified if an effect is created or deleted 2053 lockEffectChains_l(effectChains); 2054 } 2055 2056 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2057 // obtain the presentation timestamp of the next output buffer 2058 int64_t pts; 2059 status_t status = INVALID_OPERATION; 2060 2061 if (NULL != mOutput->stream->get_next_write_timestamp) { 2062 status = mOutput->stream->get_next_write_timestamp( 2063 mOutput->stream, &pts); 2064 } 2065 2066 if (status != NO_ERROR) { 2067 pts = AudioBufferProvider::kInvalidPTS; 2068 } 2069 2070 // mix buffers... 2071 mAudioMixer->process(pts); 2072 // increase sleep time progressively when application underrun condition clears. 2073 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2074 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2075 // such that we would underrun the audio HAL. 2076 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2077 sleepTimeShift--; 2078 } 2079 sleepTime = 0; 2080 standbyTime = systemTime() + mStandbyTimeInNsecs; 2081 //TODO: delay standby when effects have a tail 2082 } else { 2083 // If no tracks are ready, sleep once for the duration of an output 2084 // buffer size, then write 0s to the output 2085 if (sleepTime == 0) { 2086 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2087 sleepTime = activeSleepTime >> sleepTimeShift; 2088 if (sleepTime < kMinThreadSleepTimeUs) { 2089 sleepTime = kMinThreadSleepTimeUs; 2090 } 2091 // reduce sleep time in case of consecutive application underruns to avoid 2092 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2093 // duration we would end up writing less data than needed by the audio HAL if 2094 // the condition persists. 2095 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2096 sleepTimeShift++; 2097 } 2098 } else { 2099 sleepTime = idleSleepTime; 2100 } 2101 } else if (mBytesWritten != 0 || 2102 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2103 memset (mMixBuffer, 0, mixBufferSize); 2104 sleepTime = 0; 2105 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2106 } 2107 // TODO add standby time extension fct of effect tail 2108 } 2109 2110 if (mSuspended) { 2111 sleepTime = suspendSleepTimeUs(); 2112 } 2113 // sleepTime == 0 means we must write to audio hardware 2114 if (sleepTime == 0) { 2115 for (size_t i = 0; i < effectChains.size(); i ++) { 2116 effectChains[i]->process_l(); 2117 } 2118 // enable changes in effect chain 2119 unlockEffectChains(effectChains); 2120 mLastWriteTime = systemTime(); 2121 mInWrite = true; 2122 mBytesWritten += mixBufferSize; 2123 2124 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2125 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2126 mNumWrites++; 2127 mInWrite = false; 2128 nsecs_t now = systemTime(); 2129 nsecs_t delta = now - mLastWriteTime; 2130 if (!mStandby && delta > maxPeriod) { 2131 mNumDelayedWrites++; 2132 if ((now - lastWarning) > kWarningThrottleNs) { 2133 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2134 ns2ms(delta), mNumDelayedWrites, this); 2135 lastWarning = now; 2136 } 2137 if (mStandby) { 2138 longStandbyExit = true; 2139 } 2140 } 2141 mStandby = false; 2142 } else { 2143 // enable changes in effect chain 2144 unlockEffectChains(effectChains); 2145 usleep(sleepTime); 2146 } 2147 2148 // finally let go of all our tracks, without the lock held 2149 // since we can't guarantee the destructors won't acquire that 2150 // same lock. 2151 tracksToRemove.clear(); 2152 2153 // Effect chains will be actually deleted here if they were removed from 2154 // mEffectChains list during mixing or effects processing 2155 effectChains.clear(); 2156 } 2157 2158 if (!mStandby) { 2159 mOutput->stream->common.standby(&mOutput->stream->common); 2160 } 2161 2162 releaseWakeLock(); 2163 2164 ALOGV("MixerThread %p exiting", this); 2165 return false; 2166} 2167 2168// prepareTracks_l() must be called with ThreadBase::mLock held 2169AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2170 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2171{ 2172 2173 mixer_state mixerStatus = MIXER_IDLE; 2174 // find out which tracks need to be processed 2175 size_t count = activeTracks.size(); 2176 size_t mixedTracks = 0; 2177 size_t tracksWithEffect = 0; 2178 2179 float masterVolume = mMasterVolume; 2180 bool masterMute = mMasterMute; 2181 2182 if (masterMute) { 2183 masterVolume = 0; 2184 } 2185 // Delegate master volume control to effect in output mix effect chain if needed 2186 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2187 if (chain != 0) { 2188 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2189 chain->setVolume_l(&v, &v); 2190 masterVolume = (float)((v + (1 << 23)) >> 24); 2191 chain.clear(); 2192 } 2193 2194 for (size_t i=0 ; i<count ; i++) { 2195 sp<Track> t = activeTracks[i].promote(); 2196 if (t == 0) continue; 2197 2198 // this const just means the local variable doesn't change 2199 Track* const track = t.get(); 2200 audio_track_cblk_t* cblk = track->cblk(); 2201 2202 // The first time a track is added we wait 2203 // for all its buffers to be filled before processing it 2204 int name = track->name(); 2205 // make sure that we have enough frames to mix one full buffer. 2206 // enforce this condition only once to enable draining the buffer in case the client 2207 // app does not call stop() and relies on underrun to stop: 2208 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2209 // during last round 2210 uint32_t minFrames = 1; 2211 if (!track->isStopped() && !track->isPausing() && 2212 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2213 if (t->sampleRate() == (int)mSampleRate) { 2214 minFrames = mFrameCount; 2215 } else { 2216 // +1 for rounding and +1 for additional sample needed for interpolation 2217 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2218 // add frames already consumed but not yet released by the resampler 2219 // because cblk->framesReady() will include these frames 2220 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2221 // the minimum track buffer size is normally twice the number of frames necessary 2222 // to fill one buffer and the resampler should not leave more than one buffer worth 2223 // of unreleased frames after each pass, but just in case... 2224 ALOG_ASSERT(minFrames <= cblk->frameCount); 2225 } 2226 } 2227 if ((track->framesReady() >= minFrames) && track->isReady() && 2228 !track->isPaused() && !track->isTerminated()) 2229 { 2230 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2231 2232 mixedTracks++; 2233 2234 // track->mainBuffer() != mMixBuffer means there is an effect chain 2235 // connected to the track 2236 chain.clear(); 2237 if (track->mainBuffer() != mMixBuffer) { 2238 chain = getEffectChain_l(track->sessionId()); 2239 // Delegate volume control to effect in track effect chain if needed 2240 if (chain != 0) { 2241 tracksWithEffect++; 2242 } else { 2243 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2244 name, track->sessionId()); 2245 } 2246 } 2247 2248 2249 int param = AudioMixer::VOLUME; 2250 if (track->mFillingUpStatus == Track::FS_FILLED) { 2251 // no ramp for the first volume setting 2252 track->mFillingUpStatus = Track::FS_ACTIVE; 2253 if (track->mState == TrackBase::RESUMING) { 2254 track->mState = TrackBase::ACTIVE; 2255 param = AudioMixer::RAMP_VOLUME; 2256 } 2257 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2258 } else if (cblk->server != 0) { 2259 // If the track is stopped before the first frame was mixed, 2260 // do not apply ramp 2261 param = AudioMixer::RAMP_VOLUME; 2262 } 2263 2264 // compute volume for this track 2265 uint32_t vl, vr, va; 2266 if (track->isMuted() || track->isPausing() || 2267 mStreamTypes[track->streamType()].mute) { 2268 vl = vr = va = 0; 2269 if (track->isPausing()) { 2270 track->setPaused(); 2271 } 2272 } else { 2273 2274 // read original volumes with volume control 2275 float typeVolume = mStreamTypes[track->streamType()].volume; 2276 float v = masterVolume * typeVolume; 2277 uint32_t vlr = cblk->getVolumeLR(); 2278 vl = vlr & 0xFFFF; 2279 vr = vlr >> 16; 2280 // track volumes come from shared memory, so can't be trusted and must be clamped 2281 if (vl > MAX_GAIN_INT) { 2282 ALOGV("Track left volume out of range: %04X", vl); 2283 vl = MAX_GAIN_INT; 2284 } 2285 if (vr > MAX_GAIN_INT) { 2286 ALOGV("Track right volume out of range: %04X", vr); 2287 vr = MAX_GAIN_INT; 2288 } 2289 // now apply the master volume and stream type volume 2290 vl = (uint32_t)(v * vl) << 12; 2291 vr = (uint32_t)(v * vr) << 12; 2292 // assuming master volume and stream type volume each go up to 1.0, 2293 // vl and vr are now in 8.24 format 2294 2295 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2296 // send level comes from shared memory and so may be corrupt 2297 if (sendLevel > MAX_GAIN_INT) { 2298 ALOGV("Track send level out of range: %04X", sendLevel); 2299 sendLevel = MAX_GAIN_INT; 2300 } 2301 va = (uint32_t)(v * sendLevel); 2302 } 2303 // Delegate volume control to effect in track effect chain if needed 2304 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2305 // Do not ramp volume if volume is controlled by effect 2306 param = AudioMixer::VOLUME; 2307 track->mHasVolumeController = true; 2308 } else { 2309 // force no volume ramp when volume controller was just disabled or removed 2310 // from effect chain to avoid volume spike 2311 if (track->mHasVolumeController) { 2312 param = AudioMixer::VOLUME; 2313 } 2314 track->mHasVolumeController = false; 2315 } 2316 2317 // Convert volumes from 8.24 to 4.12 format 2318 // This additional clamping is needed in case chain->setVolume_l() overshot 2319 vl = (vl + (1 << 11)) >> 12; 2320 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2321 vr = (vr + (1 << 11)) >> 12; 2322 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2323 2324 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2325 2326 // XXX: these things DON'T need to be done each time 2327 mAudioMixer->setBufferProvider(name, track); 2328 mAudioMixer->enable(name); 2329 2330 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2331 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2332 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2333 mAudioMixer->setParameter( 2334 name, 2335 AudioMixer::TRACK, 2336 AudioMixer::FORMAT, (void *)track->format()); 2337 mAudioMixer->setParameter( 2338 name, 2339 AudioMixer::TRACK, 2340 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2341 mAudioMixer->setParameter( 2342 name, 2343 AudioMixer::RESAMPLE, 2344 AudioMixer::SAMPLE_RATE, 2345 (void *)(cblk->sampleRate)); 2346 mAudioMixer->setParameter( 2347 name, 2348 AudioMixer::TRACK, 2349 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2350 mAudioMixer->setParameter( 2351 name, 2352 AudioMixer::TRACK, 2353 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2354 2355 // reset retry count 2356 track->mRetryCount = kMaxTrackRetries; 2357 // If one track is ready, set the mixer ready if: 2358 // - the mixer was not ready during previous round OR 2359 // - no other track is not ready 2360 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2361 mixerStatus != MIXER_TRACKS_ENABLED) { 2362 mixerStatus = MIXER_TRACKS_READY; 2363 } 2364 } else { 2365 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2366 if (track->isStopped()) { 2367 track->reset(); 2368 } 2369 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2370 // We have consumed all the buffers of this track. 2371 // Remove it from the list of active tracks. 2372 tracksToRemove->add(track); 2373 } else { 2374 // No buffers for this track. Give it a few chances to 2375 // fill a buffer, then remove it from active list. 2376 if (--(track->mRetryCount) <= 0) { 2377 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2378 tracksToRemove->add(track); 2379 // indicate to client process that the track was disabled because of underrun 2380 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2381 // If one track is not ready, mark the mixer also not ready if: 2382 // - the mixer was ready during previous round OR 2383 // - no other track is ready 2384 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2385 mixerStatus != MIXER_TRACKS_READY) { 2386 mixerStatus = MIXER_TRACKS_ENABLED; 2387 } 2388 } 2389 mAudioMixer->disable(name); 2390 } 2391 } 2392 2393 // remove all the tracks that need to be... 2394 count = tracksToRemove->size(); 2395 if (CC_UNLIKELY(count)) { 2396 for (size_t i=0 ; i<count ; i++) { 2397 const sp<Track>& track = tracksToRemove->itemAt(i); 2398 mActiveTracks.remove(track); 2399 if (track->mainBuffer() != mMixBuffer) { 2400 chain = getEffectChain_l(track->sessionId()); 2401 if (chain != 0) { 2402 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2403 chain->decActiveTrackCnt(); 2404 } 2405 } 2406 if (track->isTerminated()) { 2407 removeTrack_l(track); 2408 } 2409 } 2410 } 2411 2412 // mix buffer must be cleared if all tracks are connected to an 2413 // effect chain as in this case the mixer will not write to 2414 // mix buffer and track effects will accumulate into it 2415 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2416 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2417 } 2418 2419 mPrevMixerStatus = mixerStatus; 2420 return mixerStatus; 2421} 2422 2423void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2424{ 2425 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2426 this, streamType, mTracks.size()); 2427 Mutex::Autolock _l(mLock); 2428 2429 size_t size = mTracks.size(); 2430 for (size_t i = 0; i < size; i++) { 2431 sp<Track> t = mTracks[i]; 2432 if (t->streamType() == streamType) { 2433 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2434 t->mCblk->cv.signal(); 2435 } 2436 } 2437} 2438 2439void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2440{ 2441 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2442 this, streamType, valid); 2443 Mutex::Autolock _l(mLock); 2444 2445 mStreamTypes[streamType].valid = valid; 2446} 2447 2448// getTrackName_l() must be called with ThreadBase::mLock held 2449int AudioFlinger::MixerThread::getTrackName_l() 2450{ 2451 return mAudioMixer->getTrackName(); 2452} 2453 2454// deleteTrackName_l() must be called with ThreadBase::mLock held 2455void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2456{ 2457 ALOGV("remove track (%d) and delete from mixer", name); 2458 mAudioMixer->deleteTrackName(name); 2459} 2460 2461// checkForNewParameters_l() must be called with ThreadBase::mLock held 2462bool AudioFlinger::MixerThread::checkForNewParameters_l() 2463{ 2464 bool reconfig = false; 2465 2466 while (!mNewParameters.isEmpty()) { 2467 status_t status = NO_ERROR; 2468 String8 keyValuePair = mNewParameters[0]; 2469 AudioParameter param = AudioParameter(keyValuePair); 2470 int value; 2471 2472 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2473 reconfig = true; 2474 } 2475 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2476 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2477 status = BAD_VALUE; 2478 } else { 2479 reconfig = true; 2480 } 2481 } 2482 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2483 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2484 status = BAD_VALUE; 2485 } else { 2486 reconfig = true; 2487 } 2488 } 2489 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2490 // do not accept frame count changes if tracks are open as the track buffer 2491 // size depends on frame count and correct behavior would not be guaranteed 2492 // if frame count is changed after track creation 2493 if (!mTracks.isEmpty()) { 2494 status = INVALID_OPERATION; 2495 } else { 2496 reconfig = true; 2497 } 2498 } 2499 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2500 // when changing the audio output device, call addBatteryData to notify 2501 // the change 2502 if ((int)mDevice != value) { 2503 uint32_t params = 0; 2504 // check whether speaker is on 2505 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2506 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2507 } 2508 2509 int deviceWithoutSpeaker 2510 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2511 // check if any other device (except speaker) is on 2512 if (value & deviceWithoutSpeaker ) { 2513 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2514 } 2515 2516 if (params != 0) { 2517 addBatteryData(params); 2518 } 2519 } 2520 2521 // forward device change to effects that have requested to be 2522 // aware of attached audio device. 2523 mDevice = (uint32_t)value; 2524 for (size_t i = 0; i < mEffectChains.size(); i++) { 2525 mEffectChains[i]->setDevice_l(mDevice); 2526 } 2527 } 2528 2529 if (status == NO_ERROR) { 2530 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2531 keyValuePair.string()); 2532 if (!mStandby && status == INVALID_OPERATION) { 2533 mOutput->stream->common.standby(&mOutput->stream->common); 2534 mStandby = true; 2535 mBytesWritten = 0; 2536 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2537 keyValuePair.string()); 2538 } 2539 if (status == NO_ERROR && reconfig) { 2540 delete mAudioMixer; 2541 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2542 mAudioMixer = NULL; 2543 readOutputParameters(); 2544 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2545 for (size_t i = 0; i < mTracks.size() ; i++) { 2546 int name = getTrackName_l(); 2547 if (name < 0) break; 2548 mTracks[i]->mName = name; 2549 // limit track sample rate to 2 x new output sample rate 2550 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2551 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2552 } 2553 } 2554 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2555 } 2556 } 2557 2558 mNewParameters.removeAt(0); 2559 2560 mParamStatus = status; 2561 mParamCond.signal(); 2562 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2563 // already timed out waiting for the status and will never signal the condition. 2564 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2565 } 2566 return reconfig; 2567} 2568 2569status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2570{ 2571 const size_t SIZE = 256; 2572 char buffer[SIZE]; 2573 String8 result; 2574 2575 PlaybackThread::dumpInternals(fd, args); 2576 2577 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2578 result.append(buffer); 2579 write(fd, result.string(), result.size()); 2580 return NO_ERROR; 2581} 2582 2583uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2584{ 2585 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2586} 2587 2588uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2589{ 2590 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2591} 2592 2593// ---------------------------------------------------------------------------- 2594AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2595 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2596 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2597 // mLeftVolFloat, mRightVolFloat 2598 // mLeftVolShort, mRightVolShort 2599{ 2600} 2601 2602AudioFlinger::DirectOutputThread::~DirectOutputThread() 2603{ 2604} 2605 2606void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2607{ 2608 // Do not apply volume on compressed audio 2609 if (!audio_is_linear_pcm(mFormat)) { 2610 return; 2611 } 2612 2613 // convert to signed 16 bit before volume calculation 2614 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2615 size_t count = mFrameCount * mChannelCount; 2616 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2617 int16_t *dst = mMixBuffer + count-1; 2618 while(count--) { 2619 *dst-- = (int16_t)(*src--^0x80) << 8; 2620 } 2621 } 2622 2623 size_t frameCount = mFrameCount; 2624 int16_t *out = mMixBuffer; 2625 if (ramp) { 2626 if (mChannelCount == 1) { 2627 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2628 int32_t vlInc = d / (int32_t)frameCount; 2629 int32_t vl = ((int32_t)mLeftVolShort << 16); 2630 do { 2631 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2632 out++; 2633 vl += vlInc; 2634 } while (--frameCount); 2635 2636 } else { 2637 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2638 int32_t vlInc = d / (int32_t)frameCount; 2639 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2640 int32_t vrInc = d / (int32_t)frameCount; 2641 int32_t vl = ((int32_t)mLeftVolShort << 16); 2642 int32_t vr = ((int32_t)mRightVolShort << 16); 2643 do { 2644 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2645 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2646 out += 2; 2647 vl += vlInc; 2648 vr += vrInc; 2649 } while (--frameCount); 2650 } 2651 } else { 2652 if (mChannelCount == 1) { 2653 do { 2654 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2655 out++; 2656 } while (--frameCount); 2657 } else { 2658 do { 2659 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2660 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2661 out += 2; 2662 } while (--frameCount); 2663 } 2664 } 2665 2666 // convert back to unsigned 8 bit after volume calculation 2667 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2668 size_t count = mFrameCount * mChannelCount; 2669 int16_t *src = mMixBuffer; 2670 uint8_t *dst = (uint8_t *)mMixBuffer; 2671 while(count--) { 2672 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2673 } 2674 } 2675 2676 mLeftVolShort = leftVol; 2677 mRightVolShort = rightVol; 2678} 2679 2680bool AudioFlinger::DirectOutputThread::threadLoop() 2681{ 2682 sp<Track> trackToRemove; 2683 sp<Track> activeTrack; 2684 nsecs_t standbyTime = systemTime(); 2685 size_t mixBufferSize = mFrameCount*mFrameSize; 2686 uint32_t activeSleepTime = activeSleepTimeUs(); 2687 uint32_t idleSleepTime = idleSleepTimeUs(); 2688 uint32_t sleepTime = idleSleepTime; 2689 // use shorter standby delay as on normal output to release 2690 // hardware resources as soon as possible 2691 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2692 2693 acquireWakeLock(); 2694 2695 while (!exitPending()) 2696 { 2697 bool rampVolume; 2698 uint16_t leftVol; 2699 uint16_t rightVol; 2700 Vector< sp<EffectChain> > effectChains; 2701 2702 processConfigEvents(); 2703 2704 mixer_state mixerStatus = MIXER_IDLE; 2705 { // scope for the mLock 2706 2707 Mutex::Autolock _l(mLock); 2708 2709 if (checkForNewParameters_l()) { 2710 mixBufferSize = mFrameCount*mFrameSize; 2711 activeSleepTime = activeSleepTimeUs(); 2712 idleSleepTime = idleSleepTimeUs(); 2713 standbyDelay = microseconds(activeSleepTime*2); 2714 } 2715 2716 // put audio hardware into standby after short delay 2717 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2718 mSuspended)) { 2719 // wait until we have something to do... 2720 if (!mStandby) { 2721 ALOGV("Audio hardware entering standby, mixer %p", this); 2722 mOutput->stream->common.standby(&mOutput->stream->common); 2723 mStandby = true; 2724 mBytesWritten = 0; 2725 } 2726 2727 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2728 // we're about to wait, flush the binder command buffer 2729 IPCThreadState::self()->flushCommands(); 2730 2731 if (exitPending()) break; 2732 2733 releaseWakeLock_l(); 2734 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2735 mWaitWorkCV.wait(mLock); 2736 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2737 acquireWakeLock_l(); 2738 2739 if (!mMasterMute) { 2740 char value[PROPERTY_VALUE_MAX]; 2741 property_get("ro.audio.silent", value, "0"); 2742 if (atoi(value)) { 2743 ALOGD("Silence is golden"); 2744 setMasterMute_l(true); 2745 } 2746 } 2747 2748 standbyTime = systemTime() + standbyDelay; 2749 sleepTime = idleSleepTime; 2750 continue; 2751 } 2752 } 2753 2754 effectChains = mEffectChains; 2755 2756 // find out which tracks need to be processed 2757 if (mActiveTracks.size() != 0) { 2758 sp<Track> t = mActiveTracks[0].promote(); 2759 if (t == 0) continue; 2760 2761 Track* const track = t.get(); 2762 audio_track_cblk_t* cblk = track->cblk(); 2763 2764 // The first time a track is added we wait 2765 // for all its buffers to be filled before processing it 2766 if (cblk->framesReady() && track->isReady() && 2767 !track->isPaused() && !track->isTerminated()) 2768 { 2769 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2770 2771 if (track->mFillingUpStatus == Track::FS_FILLED) { 2772 track->mFillingUpStatus = Track::FS_ACTIVE; 2773 mLeftVolFloat = mRightVolFloat = 0; 2774 mLeftVolShort = mRightVolShort = 0; 2775 if (track->mState == TrackBase::RESUMING) { 2776 track->mState = TrackBase::ACTIVE; 2777 rampVolume = true; 2778 } 2779 } else if (cblk->server != 0) { 2780 // If the track is stopped before the first frame was mixed, 2781 // do not apply ramp 2782 rampVolume = true; 2783 } 2784 // compute volume for this track 2785 float left, right; 2786 if (track->isMuted() || mMasterMute || track->isPausing() || 2787 mStreamTypes[track->streamType()].mute) { 2788 left = right = 0; 2789 if (track->isPausing()) { 2790 track->setPaused(); 2791 } 2792 } else { 2793 float typeVolume = mStreamTypes[track->streamType()].volume; 2794 float v = mMasterVolume * typeVolume; 2795 uint32_t vlr = cblk->getVolumeLR(); 2796 float v_clamped = v * (vlr & 0xFFFF); 2797 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2798 left = v_clamped/MAX_GAIN; 2799 v_clamped = v * (vlr >> 16); 2800 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2801 right = v_clamped/MAX_GAIN; 2802 } 2803 2804 if (left != mLeftVolFloat || right != mRightVolFloat) { 2805 mLeftVolFloat = left; 2806 mRightVolFloat = right; 2807 2808 // If audio HAL implements volume control, 2809 // force software volume to nominal value 2810 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2811 left = 1.0f; 2812 right = 1.0f; 2813 } 2814 2815 // Convert volumes from float to 8.24 2816 uint32_t vl = (uint32_t)(left * (1 << 24)); 2817 uint32_t vr = (uint32_t)(right * (1 << 24)); 2818 2819 // Delegate volume control to effect in track effect chain if needed 2820 // only one effect chain can be present on DirectOutputThread, so if 2821 // there is one, the track is connected to it 2822 if (!effectChains.isEmpty()) { 2823 // Do not ramp volume if volume is controlled by effect 2824 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2825 rampVolume = false; 2826 } 2827 } 2828 2829 // Convert volumes from 8.24 to 4.12 format 2830 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2831 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2832 leftVol = (uint16_t)v_clamped; 2833 v_clamped = (vr + (1 << 11)) >> 12; 2834 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2835 rightVol = (uint16_t)v_clamped; 2836 } else { 2837 leftVol = mLeftVolShort; 2838 rightVol = mRightVolShort; 2839 rampVolume = false; 2840 } 2841 2842 // reset retry count 2843 track->mRetryCount = kMaxTrackRetriesDirect; 2844 activeTrack = t; 2845 mixerStatus = MIXER_TRACKS_READY; 2846 } else { 2847 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2848 if (track->isStopped()) { 2849 track->reset(); 2850 } 2851 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2852 // We have consumed all the buffers of this track. 2853 // Remove it from the list of active tracks. 2854 trackToRemove = track; 2855 } else { 2856 // No buffers for this track. Give it a few chances to 2857 // fill a buffer, then remove it from active list. 2858 if (--(track->mRetryCount) <= 0) { 2859 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2860 trackToRemove = track; 2861 } else { 2862 mixerStatus = MIXER_TRACKS_ENABLED; 2863 } 2864 } 2865 } 2866 } 2867 2868 // remove all the tracks that need to be... 2869 if (CC_UNLIKELY(trackToRemove != 0)) { 2870 mActiveTracks.remove(trackToRemove); 2871 if (!effectChains.isEmpty()) { 2872 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2873 trackToRemove->sessionId()); 2874 effectChains[0]->decActiveTrackCnt(); 2875 } 2876 if (trackToRemove->isTerminated()) { 2877 removeTrack_l(trackToRemove); 2878 } 2879 } 2880 2881 lockEffectChains_l(effectChains); 2882 } 2883 2884 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2885 AudioBufferProvider::Buffer buffer; 2886 size_t frameCount = mFrameCount; 2887 int8_t *curBuf = (int8_t *)mMixBuffer; 2888 // output audio to hardware 2889 while (frameCount) { 2890 buffer.frameCount = frameCount; 2891 activeTrack->getNextBuffer(&buffer, 2892 AudioBufferProvider::kInvalidPTS); 2893 if (CC_UNLIKELY(buffer.raw == NULL)) { 2894 memset(curBuf, 0, frameCount * mFrameSize); 2895 break; 2896 } 2897 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2898 frameCount -= buffer.frameCount; 2899 curBuf += buffer.frameCount * mFrameSize; 2900 activeTrack->releaseBuffer(&buffer); 2901 } 2902 sleepTime = 0; 2903 standbyTime = systemTime() + standbyDelay; 2904 } else { 2905 if (sleepTime == 0) { 2906 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2907 sleepTime = activeSleepTime; 2908 } else { 2909 sleepTime = idleSleepTime; 2910 } 2911 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2912 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2913 sleepTime = 0; 2914 } 2915 } 2916 2917 if (mSuspended) { 2918 sleepTime = suspendSleepTimeUs(); 2919 } 2920 // sleepTime == 0 means we must write to audio hardware 2921 if (sleepTime == 0) { 2922 if (mixerStatus == MIXER_TRACKS_READY) { 2923 applyVolume(leftVol, rightVol, rampVolume); 2924 } 2925 for (size_t i = 0; i < effectChains.size(); i ++) { 2926 effectChains[i]->process_l(); 2927 } 2928 unlockEffectChains(effectChains); 2929 2930 mLastWriteTime = systemTime(); 2931 mInWrite = true; 2932 mBytesWritten += mixBufferSize; 2933 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2934 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2935 mNumWrites++; 2936 mInWrite = false; 2937 mStandby = false; 2938 } else { 2939 unlockEffectChains(effectChains); 2940 usleep(sleepTime); 2941 } 2942 2943 // finally let go of removed track, without the lock held 2944 // since we can't guarantee the destructors won't acquire that 2945 // same lock. 2946 trackToRemove.clear(); 2947 activeTrack.clear(); 2948 2949 // Effect chains will be actually deleted here if they were removed from 2950 // mEffectChains list during mixing or effects processing 2951 effectChains.clear(); 2952 } 2953 2954 if (!mStandby) { 2955 mOutput->stream->common.standby(&mOutput->stream->common); 2956 } 2957 2958 releaseWakeLock(); 2959 2960 ALOGV("DirectOutputThread %p exiting", this); 2961 return false; 2962} 2963 2964// getTrackName_l() must be called with ThreadBase::mLock held 2965int AudioFlinger::DirectOutputThread::getTrackName_l() 2966{ 2967 return 0; 2968} 2969 2970// deleteTrackName_l() must be called with ThreadBase::mLock held 2971void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2972{ 2973} 2974 2975// checkForNewParameters_l() must be called with ThreadBase::mLock held 2976bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2977{ 2978 bool reconfig = false; 2979 2980 while (!mNewParameters.isEmpty()) { 2981 status_t status = NO_ERROR; 2982 String8 keyValuePair = mNewParameters[0]; 2983 AudioParameter param = AudioParameter(keyValuePair); 2984 int value; 2985 2986 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2987 // do not accept frame count changes if tracks are open as the track buffer 2988 // size depends on frame count and correct behavior would not be garantied 2989 // if frame count is changed after track creation 2990 if (!mTracks.isEmpty()) { 2991 status = INVALID_OPERATION; 2992 } else { 2993 reconfig = true; 2994 } 2995 } 2996 if (status == NO_ERROR) { 2997 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2998 keyValuePair.string()); 2999 if (!mStandby && status == INVALID_OPERATION) { 3000 mOutput->stream->common.standby(&mOutput->stream->common); 3001 mStandby = true; 3002 mBytesWritten = 0; 3003 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3004 keyValuePair.string()); 3005 } 3006 if (status == NO_ERROR && reconfig) { 3007 readOutputParameters(); 3008 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3009 } 3010 } 3011 3012 mNewParameters.removeAt(0); 3013 3014 mParamStatus = status; 3015 mParamCond.signal(); 3016 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3017 // already timed out waiting for the status and will never signal the condition. 3018 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3019 } 3020 return reconfig; 3021} 3022 3023uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3024{ 3025 uint32_t time; 3026 if (audio_is_linear_pcm(mFormat)) { 3027 time = PlaybackThread::activeSleepTimeUs(); 3028 } else { 3029 time = 10000; 3030 } 3031 return time; 3032} 3033 3034uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3035{ 3036 uint32_t time; 3037 if (audio_is_linear_pcm(mFormat)) { 3038 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3039 } else { 3040 time = 10000; 3041 } 3042 return time; 3043} 3044 3045uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3046{ 3047 uint32_t time; 3048 if (audio_is_linear_pcm(mFormat)) { 3049 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3050 } else { 3051 time = 10000; 3052 } 3053 return time; 3054} 3055 3056 3057// ---------------------------------------------------------------------------- 3058 3059AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3060 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3061 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3062 mWaitTimeMs(UINT_MAX) 3063{ 3064 addOutputTrack(mainThread); 3065} 3066 3067AudioFlinger::DuplicatingThread::~DuplicatingThread() 3068{ 3069 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3070 mOutputTracks[i]->destroy(); 3071 } 3072} 3073 3074bool AudioFlinger::DuplicatingThread::threadLoop() 3075{ 3076 Vector< sp<Track> > tracksToRemove; 3077 nsecs_t standbyTime = systemTime(); 3078 size_t mixBufferSize = mFrameCount*mFrameSize; 3079 SortedVector< sp<OutputTrack> > outputTracks; 3080 uint32_t writeFrames = 0; 3081 uint32_t activeSleepTime = activeSleepTimeUs(); 3082 uint32_t idleSleepTime = idleSleepTimeUs(); 3083 uint32_t sleepTime = idleSleepTime; 3084 Vector< sp<EffectChain> > effectChains; 3085 3086 acquireWakeLock(); 3087 3088 while (!exitPending()) 3089 { 3090 processConfigEvents(); 3091 3092 mixer_state mixerStatus = MIXER_IDLE; 3093 { // scope for the mLock 3094 3095 Mutex::Autolock _l(mLock); 3096 3097 if (checkForNewParameters_l()) { 3098 mixBufferSize = mFrameCount*mFrameSize; 3099 updateWaitTime(); 3100 activeSleepTime = activeSleepTimeUs(); 3101 idleSleepTime = idleSleepTimeUs(); 3102 } 3103 3104 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3105 3106 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3107 outputTracks.add(mOutputTracks[i]); 3108 } 3109 3110 // put audio hardware into standby after short delay 3111 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3112 mSuspended)) { 3113 if (!mStandby) { 3114 for (size_t i = 0; i < outputTracks.size(); i++) { 3115 outputTracks[i]->stop(); 3116 } 3117 mStandby = true; 3118 mBytesWritten = 0; 3119 } 3120 3121 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3122 // we're about to wait, flush the binder command buffer 3123 IPCThreadState::self()->flushCommands(); 3124 outputTracks.clear(); 3125 3126 if (exitPending()) break; 3127 3128 releaseWakeLock_l(); 3129 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3130 mWaitWorkCV.wait(mLock); 3131 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3132 acquireWakeLock_l(); 3133 3134 mPrevMixerStatus = MIXER_IDLE; 3135 if (!mMasterMute) { 3136 char value[PROPERTY_VALUE_MAX]; 3137 property_get("ro.audio.silent", value, "0"); 3138 if (atoi(value)) { 3139 ALOGD("Silence is golden"); 3140 setMasterMute_l(true); 3141 } 3142 } 3143 3144 standbyTime = systemTime() + mStandbyTimeInNsecs; 3145 sleepTime = idleSleepTime; 3146 continue; 3147 } 3148 } 3149 3150 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3151 3152 // prevent any changes in effect chain list and in each effect chain 3153 // during mixing and effect process as the audio buffers could be deleted 3154 // or modified if an effect is created or deleted 3155 lockEffectChains_l(effectChains); 3156 } 3157 3158 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3159 // mix buffers... 3160 if (outputsReady(outputTracks)) { 3161 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3162 } else { 3163 memset(mMixBuffer, 0, mixBufferSize); 3164 } 3165 sleepTime = 0; 3166 writeFrames = mFrameCount; 3167 } else { 3168 if (sleepTime == 0) { 3169 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3170 sleepTime = activeSleepTime; 3171 } else { 3172 sleepTime = idleSleepTime; 3173 } 3174 } else if (mBytesWritten != 0) { 3175 // flush remaining overflow buffers in output tracks 3176 for (size_t i = 0; i < outputTracks.size(); i++) { 3177 if (outputTracks[i]->isActive()) { 3178 sleepTime = 0; 3179 writeFrames = 0; 3180 memset(mMixBuffer, 0, mixBufferSize); 3181 break; 3182 } 3183 } 3184 } 3185 } 3186 3187 if (mSuspended) { 3188 sleepTime = suspendSleepTimeUs(); 3189 } 3190 // sleepTime == 0 means we must write to audio hardware 3191 if (sleepTime == 0) { 3192 for (size_t i = 0; i < effectChains.size(); i ++) { 3193 effectChains[i]->process_l(); 3194 } 3195 // enable changes in effect chain 3196 unlockEffectChains(effectChains); 3197 3198 standbyTime = systemTime() + mStandbyTimeInNsecs; 3199 for (size_t i = 0; i < outputTracks.size(); i++) { 3200 outputTracks[i]->write(mMixBuffer, writeFrames); 3201 } 3202 mStandby = false; 3203 mBytesWritten += mixBufferSize; 3204 } else { 3205 // enable changes in effect chain 3206 unlockEffectChains(effectChains); 3207 usleep(sleepTime); 3208 } 3209 3210 // finally let go of all our tracks, without the lock held 3211 // since we can't guarantee the destructors won't acquire that 3212 // same lock. 3213 tracksToRemove.clear(); 3214 outputTracks.clear(); 3215 3216 // Effect chains will be actually deleted here if they were removed from 3217 // mEffectChains list during mixing or effects processing 3218 effectChains.clear(); 3219 } 3220 3221 releaseWakeLock(); 3222 3223 return false; 3224} 3225 3226void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3227{ 3228 // FIXME explain this formula 3229 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3230 OutputTrack *outputTrack = new OutputTrack(thread, 3231 this, 3232 mSampleRate, 3233 mFormat, 3234 mChannelMask, 3235 frameCount); 3236 if (outputTrack->cblk() != NULL) { 3237 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3238 mOutputTracks.add(outputTrack); 3239 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3240 updateWaitTime(); 3241 } 3242} 3243 3244void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3245{ 3246 Mutex::Autolock _l(mLock); 3247 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3248 if (mOutputTracks[i]->thread() == thread) { 3249 mOutputTracks[i]->destroy(); 3250 mOutputTracks.removeAt(i); 3251 updateWaitTime(); 3252 return; 3253 } 3254 } 3255 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3256} 3257 3258void AudioFlinger::DuplicatingThread::updateWaitTime() 3259{ 3260 mWaitTimeMs = UINT_MAX; 3261 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3262 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3263 if (strong != 0) { 3264 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3265 if (waitTimeMs < mWaitTimeMs) { 3266 mWaitTimeMs = waitTimeMs; 3267 } 3268 } 3269 } 3270} 3271 3272 3273bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3274{ 3275 for (size_t i = 0; i < outputTracks.size(); i++) { 3276 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3277 if (thread == 0) { 3278 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3279 return false; 3280 } 3281 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3282 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3283 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3284 return false; 3285 } 3286 } 3287 return true; 3288} 3289 3290uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3291{ 3292 return (mWaitTimeMs * 1000) / 2; 3293} 3294 3295// ---------------------------------------------------------------------------- 3296 3297// TrackBase constructor must be called with AudioFlinger::mLock held 3298AudioFlinger::ThreadBase::TrackBase::TrackBase( 3299 ThreadBase *thread, 3300 const sp<Client>& client, 3301 uint32_t sampleRate, 3302 audio_format_t format, 3303 uint32_t channelMask, 3304 int frameCount, 3305 uint32_t flags, 3306 const sp<IMemory>& sharedBuffer, 3307 int sessionId) 3308 : RefBase(), 3309 mThread(thread), 3310 mClient(client), 3311 mCblk(NULL), 3312 // mBuffer 3313 // mBufferEnd 3314 mFrameCount(0), 3315 mState(IDLE), 3316 mFormat(format), 3317 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3318 mSessionId(sessionId) 3319 // mChannelCount 3320 // mChannelMask 3321{ 3322 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3323 3324 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3325 size_t size = sizeof(audio_track_cblk_t); 3326 uint8_t channelCount = popcount(channelMask); 3327 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3328 if (sharedBuffer == 0) { 3329 size += bufferSize; 3330 } 3331 3332 if (client != NULL) { 3333 mCblkMemory = client->heap()->allocate(size); 3334 if (mCblkMemory != 0) { 3335 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3336 if (mCblk != NULL) { // construct the shared structure in-place. 3337 new(mCblk) audio_track_cblk_t(); 3338 // clear all buffers 3339 mCblk->frameCount = frameCount; 3340 mCblk->sampleRate = sampleRate; 3341 mChannelCount = channelCount; 3342 mChannelMask = channelMask; 3343 if (sharedBuffer == 0) { 3344 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3345 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3346 // Force underrun condition to avoid false underrun callback until first data is 3347 // written to buffer (other flags are cleared) 3348 mCblk->flags = CBLK_UNDERRUN_ON; 3349 } else { 3350 mBuffer = sharedBuffer->pointer(); 3351 } 3352 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3353 } 3354 } else { 3355 ALOGE("not enough memory for AudioTrack size=%u", size); 3356 client->heap()->dump("AudioTrack"); 3357 return; 3358 } 3359 } else { 3360 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3361 // construct the shared structure in-place. 3362 new(mCblk) audio_track_cblk_t(); 3363 // clear all buffers 3364 mCblk->frameCount = frameCount; 3365 mCblk->sampleRate = sampleRate; 3366 mChannelCount = channelCount; 3367 mChannelMask = channelMask; 3368 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3369 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3370 // Force underrun condition to avoid false underrun callback until first data is 3371 // written to buffer (other flags are cleared) 3372 mCblk->flags = CBLK_UNDERRUN_ON; 3373 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3374 } 3375} 3376 3377AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3378{ 3379 if (mCblk != NULL) { 3380 if (mClient == 0) { 3381 delete mCblk; 3382 } else { 3383 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3384 } 3385 } 3386 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3387 if (mClient != 0) { 3388 // Client destructor must run with AudioFlinger mutex locked 3389 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3390 // If the client's reference count drops to zero, the associated destructor 3391 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3392 // relying on the automatic clear() at end of scope. 3393 mClient.clear(); 3394 } 3395} 3396 3397void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3398{ 3399 buffer->raw = NULL; 3400 mFrameCount = buffer->frameCount; 3401 step(); 3402 buffer->frameCount = 0; 3403} 3404 3405bool AudioFlinger::ThreadBase::TrackBase::step() { 3406 bool result; 3407 audio_track_cblk_t* cblk = this->cblk(); 3408 3409 result = cblk->stepServer(mFrameCount); 3410 if (!result) { 3411 ALOGV("stepServer failed acquiring cblk mutex"); 3412 mFlags |= STEPSERVER_FAILED; 3413 } 3414 return result; 3415} 3416 3417void AudioFlinger::ThreadBase::TrackBase::reset() { 3418 audio_track_cblk_t* cblk = this->cblk(); 3419 3420 cblk->user = 0; 3421 cblk->server = 0; 3422 cblk->userBase = 0; 3423 cblk->serverBase = 0; 3424 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3425 ALOGV("TrackBase::reset"); 3426} 3427 3428int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3429 return (int)mCblk->sampleRate; 3430} 3431 3432void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3433 audio_track_cblk_t* cblk = this->cblk(); 3434 size_t frameSize = cblk->frameSize; 3435 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3436 int8_t *bufferEnd = bufferStart + frames * frameSize; 3437 3438 // Check validity of returned pointer in case the track control block would have been corrupted. 3439 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3440 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3441 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3442 server %d, serverBase %d, user %d, userBase %d", 3443 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3444 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3445 return NULL; 3446 } 3447 3448 return bufferStart; 3449} 3450 3451// ---------------------------------------------------------------------------- 3452 3453// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3454AudioFlinger::PlaybackThread::Track::Track( 3455 PlaybackThread *thread, 3456 const sp<Client>& client, 3457 audio_stream_type_t streamType, 3458 uint32_t sampleRate, 3459 audio_format_t format, 3460 uint32_t channelMask, 3461 int frameCount, 3462 const sp<IMemory>& sharedBuffer, 3463 int sessionId) 3464 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3465 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3466 mAuxEffectId(0), mHasVolumeController(false) 3467{ 3468 if (mCblk != NULL) { 3469 if (thread != NULL) { 3470 mName = thread->getTrackName_l(); 3471 mMainBuffer = thread->mixBuffer(); 3472 } 3473 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3474 if (mName < 0) { 3475 ALOGE("no more track names available"); 3476 } 3477 mStreamType = streamType; 3478 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3479 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3480 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3481 } 3482} 3483 3484AudioFlinger::PlaybackThread::Track::~Track() 3485{ 3486 ALOGV("PlaybackThread::Track destructor"); 3487 sp<ThreadBase> thread = mThread.promote(); 3488 if (thread != 0) { 3489 Mutex::Autolock _l(thread->mLock); 3490 mState = TERMINATED; 3491 } 3492} 3493 3494void AudioFlinger::PlaybackThread::Track::destroy() 3495{ 3496 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3497 // by removing it from mTracks vector, so there is a risk that this Tracks's 3498 // destructor is called. As the destructor needs to lock mLock, 3499 // we must acquire a strong reference on this Track before locking mLock 3500 // here so that the destructor is called only when exiting this function. 3501 // On the other hand, as long as Track::destroy() is only called by 3502 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3503 // this Track with its member mTrack. 3504 sp<Track> keep(this); 3505 { // scope for mLock 3506 sp<ThreadBase> thread = mThread.promote(); 3507 if (thread != 0) { 3508 if (!isOutputTrack()) { 3509 if (mState == ACTIVE || mState == RESUMING) { 3510 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3511 3512 // to track the speaker usage 3513 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3514 } 3515 AudioSystem::releaseOutput(thread->id()); 3516 } 3517 Mutex::Autolock _l(thread->mLock); 3518 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3519 playbackThread->destroyTrack_l(this); 3520 } 3521 } 3522} 3523 3524void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3525{ 3526 uint32_t vlr = mCblk->getVolumeLR(); 3527 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", 3528 mName - AudioMixer::TRACK0, 3529 (mClient == 0) ? getpid_cached : mClient->pid(), 3530 mStreamType, 3531 mFormat, 3532 mChannelMask, 3533 mSessionId, 3534 mFrameCount, 3535 mState, 3536 mMute, 3537 mFillingUpStatus, 3538 mCblk->sampleRate, 3539 vlr & 0xFFFF, 3540 vlr >> 16, 3541 mCblk->server, 3542 mCblk->user, 3543 (int)mMainBuffer, 3544 (int)mAuxBuffer); 3545} 3546 3547status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3548 AudioBufferProvider::Buffer* buffer, int64_t pts) 3549{ 3550 audio_track_cblk_t* cblk = this->cblk(); 3551 uint32_t framesReady; 3552 uint32_t framesReq = buffer->frameCount; 3553 3554 // Check if last stepServer failed, try to step now 3555 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3556 if (!step()) goto getNextBuffer_exit; 3557 ALOGV("stepServer recovered"); 3558 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3559 } 3560 3561 framesReady = cblk->framesReady(); 3562 3563 if (CC_LIKELY(framesReady)) { 3564 uint32_t s = cblk->server; 3565 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3566 3567 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3568 if (framesReq > framesReady) { 3569 framesReq = framesReady; 3570 } 3571 if (s + framesReq > bufferEnd) { 3572 framesReq = bufferEnd - s; 3573 } 3574 3575 buffer->raw = getBuffer(s, framesReq); 3576 if (buffer->raw == NULL) goto getNextBuffer_exit; 3577 3578 buffer->frameCount = framesReq; 3579 return NO_ERROR; 3580 } 3581 3582getNextBuffer_exit: 3583 buffer->raw = NULL; 3584 buffer->frameCount = 0; 3585 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3586 return NOT_ENOUGH_DATA; 3587} 3588 3589uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3590 return mCblk->framesReady(); 3591} 3592 3593bool AudioFlinger::PlaybackThread::Track::isReady() const { 3594 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3595 3596 if (framesReady() >= mCblk->frameCount || 3597 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3598 mFillingUpStatus = FS_FILLED; 3599 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3600 return true; 3601 } 3602 return false; 3603} 3604 3605status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3606{ 3607 status_t status = NO_ERROR; 3608 ALOGV("start(%d), calling pid %d session %d tid %d", 3609 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3610 sp<ThreadBase> thread = mThread.promote(); 3611 if (thread != 0) { 3612 Mutex::Autolock _l(thread->mLock); 3613 track_state state = mState; 3614 // here the track could be either new, or restarted 3615 // in both cases "unstop" the track 3616 if (mState == PAUSED) { 3617 mState = TrackBase::RESUMING; 3618 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3619 } else { 3620 mState = TrackBase::ACTIVE; 3621 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3622 } 3623 3624 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3625 thread->mLock.unlock(); 3626 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3627 thread->mLock.lock(); 3628 3629 // to track the speaker usage 3630 if (status == NO_ERROR) { 3631 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3632 } 3633 } 3634 if (status == NO_ERROR) { 3635 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3636 playbackThread->addTrack_l(this); 3637 } else { 3638 mState = state; 3639 } 3640 } else { 3641 status = BAD_VALUE; 3642 } 3643 return status; 3644} 3645 3646void AudioFlinger::PlaybackThread::Track::stop() 3647{ 3648 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3649 sp<ThreadBase> thread = mThread.promote(); 3650 if (thread != 0) { 3651 Mutex::Autolock _l(thread->mLock); 3652 track_state state = mState; 3653 if (mState > STOPPED) { 3654 mState = STOPPED; 3655 // If the track is not active (PAUSED and buffers full), flush buffers 3656 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3657 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3658 reset(); 3659 } 3660 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3661 } 3662 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3663 thread->mLock.unlock(); 3664 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3665 thread->mLock.lock(); 3666 3667 // to track the speaker usage 3668 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3669 } 3670 } 3671} 3672 3673void AudioFlinger::PlaybackThread::Track::pause() 3674{ 3675 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3676 sp<ThreadBase> thread = mThread.promote(); 3677 if (thread != 0) { 3678 Mutex::Autolock _l(thread->mLock); 3679 if (mState == ACTIVE || mState == RESUMING) { 3680 mState = PAUSING; 3681 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3682 if (!isOutputTrack()) { 3683 thread->mLock.unlock(); 3684 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3685 thread->mLock.lock(); 3686 3687 // to track the speaker usage 3688 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3689 } 3690 } 3691 } 3692} 3693 3694void AudioFlinger::PlaybackThread::Track::flush() 3695{ 3696 ALOGV("flush(%d)", mName); 3697 sp<ThreadBase> thread = mThread.promote(); 3698 if (thread != 0) { 3699 Mutex::Autolock _l(thread->mLock); 3700 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3701 return; 3702 } 3703 // No point remaining in PAUSED state after a flush => go to 3704 // STOPPED state 3705 mState = STOPPED; 3706 3707 // do not reset the track if it is still in the process of being stopped or paused. 3708 // this will be done by prepareTracks_l() when the track is stopped. 3709 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3710 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3711 reset(); 3712 } 3713 } 3714} 3715 3716void AudioFlinger::PlaybackThread::Track::reset() 3717{ 3718 // Do not reset twice to avoid discarding data written just after a flush and before 3719 // the audioflinger thread detects the track is stopped. 3720 if (!mResetDone) { 3721 TrackBase::reset(); 3722 // Force underrun condition to avoid false underrun callback until first data is 3723 // written to buffer 3724 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3725 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3726 mFillingUpStatus = FS_FILLING; 3727 mResetDone = true; 3728 } 3729} 3730 3731void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3732{ 3733 mMute = muted; 3734} 3735 3736status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3737{ 3738 status_t status = DEAD_OBJECT; 3739 sp<ThreadBase> thread = mThread.promote(); 3740 if (thread != 0) { 3741 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3742 status = playbackThread->attachAuxEffect(this, EffectId); 3743 } 3744 return status; 3745} 3746 3747void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3748{ 3749 mAuxEffectId = EffectId; 3750 mAuxBuffer = buffer; 3751} 3752 3753// timed audio tracks 3754 3755sp<AudioFlinger::PlaybackThread::TimedTrack> 3756AudioFlinger::PlaybackThread::TimedTrack::create( 3757 PlaybackThread *thread, 3758 const sp<Client>& client, 3759 audio_stream_type_t streamType, 3760 uint32_t sampleRate, 3761 audio_format_t format, 3762 uint32_t channelMask, 3763 int frameCount, 3764 const sp<IMemory>& sharedBuffer, 3765 int sessionId) { 3766 if (!client->reserveTimedTrack()) 3767 return NULL; 3768 3769 sp<TimedTrack> track = new TimedTrack( 3770 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3771 sharedBuffer, sessionId); 3772 3773 if (track == NULL) { 3774 client->releaseTimedTrack(); 3775 return NULL; 3776 } 3777 3778 return track; 3779} 3780 3781AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3782 PlaybackThread *thread, 3783 const sp<Client>& client, 3784 audio_stream_type_t streamType, 3785 uint32_t sampleRate, 3786 audio_format_t format, 3787 uint32_t channelMask, 3788 int frameCount, 3789 const sp<IMemory>& sharedBuffer, 3790 int sessionId) 3791 : Track(thread, client, streamType, sampleRate, format, channelMask, 3792 frameCount, sharedBuffer, sessionId), 3793 mTimedSilenceBuffer(NULL), 3794 mTimedSilenceBufferSize(0), 3795 mTimedAudioOutputOnTime(false), 3796 mMediaTimeTransformValid(false) 3797{ 3798 LocalClock lc; 3799 mLocalTimeFreq = lc.getLocalFreq(); 3800 3801 mLocalTimeToSampleTransform.a_zero = 0; 3802 mLocalTimeToSampleTransform.b_zero = 0; 3803 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3804 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3805 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3806 &mLocalTimeToSampleTransform.a_to_b_denom); 3807} 3808 3809AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3810 mClient->releaseTimedTrack(); 3811 delete [] mTimedSilenceBuffer; 3812} 3813 3814status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3815 size_t size, sp<IMemory>* buffer) { 3816 3817 Mutex::Autolock _l(mTimedBufferQueueLock); 3818 3819 trimTimedBufferQueue_l(); 3820 3821 // lazily initialize the shared memory heap for timed buffers 3822 if (mTimedMemoryDealer == NULL) { 3823 const int kTimedBufferHeapSize = 512 << 10; 3824 3825 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3826 "AudioFlingerTimed"); 3827 if (mTimedMemoryDealer == NULL) 3828 return NO_MEMORY; 3829 } 3830 3831 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3832 if (newBuffer == NULL) { 3833 newBuffer = mTimedMemoryDealer->allocate(size); 3834 if (newBuffer == NULL) 3835 return NO_MEMORY; 3836 } 3837 3838 *buffer = newBuffer; 3839 return NO_ERROR; 3840} 3841 3842// caller must hold mTimedBufferQueueLock 3843void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3844 int64_t mediaTimeNow; 3845 { 3846 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3847 if (!mMediaTimeTransformValid) 3848 return; 3849 3850 int64_t targetTimeNow; 3851 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3852 ? mCCHelper.getCommonTime(&targetTimeNow) 3853 : mCCHelper.getLocalTime(&targetTimeNow); 3854 3855 if (OK != res) 3856 return; 3857 3858 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3859 &mediaTimeNow)) { 3860 return; 3861 } 3862 } 3863 3864 size_t trimIndex; 3865 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3866 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3867 break; 3868 } 3869 3870 if (trimIndex) { 3871 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3872 } 3873} 3874 3875status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3876 const sp<IMemory>& buffer, int64_t pts) { 3877 3878 { 3879 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3880 if (!mMediaTimeTransformValid) 3881 return INVALID_OPERATION; 3882 } 3883 3884 Mutex::Autolock _l(mTimedBufferQueueLock); 3885 3886 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3887 3888 return NO_ERROR; 3889} 3890 3891status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3892 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3893 3894 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3895 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3896 target); 3897 3898 if (!(target == TimedAudioTrack::LOCAL_TIME || 3899 target == TimedAudioTrack::COMMON_TIME)) { 3900 return BAD_VALUE; 3901 } 3902 3903 Mutex::Autolock lock(mMediaTimeTransformLock); 3904 mMediaTimeTransform = xform; 3905 mMediaTimeTransformTarget = target; 3906 mMediaTimeTransformValid = true; 3907 3908 return NO_ERROR; 3909} 3910 3911#define min(a, b) ((a) < (b) ? (a) : (b)) 3912 3913// implementation of getNextBuffer for tracks whose buffers have timestamps 3914status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3915 AudioBufferProvider::Buffer* buffer, int64_t pts) 3916{ 3917 if (pts == AudioBufferProvider::kInvalidPTS) { 3918 buffer->raw = 0; 3919 buffer->frameCount = 0; 3920 return INVALID_OPERATION; 3921 } 3922 3923 Mutex::Autolock _l(mTimedBufferQueueLock); 3924 3925 while (true) { 3926 3927 // if we have no timed buffers, then fail 3928 if (mTimedBufferQueue.isEmpty()) { 3929 buffer->raw = 0; 3930 buffer->frameCount = 0; 3931 return NOT_ENOUGH_DATA; 3932 } 3933 3934 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3935 3936 // calculate the PTS of the head of the timed buffer queue expressed in 3937 // local time 3938 int64_t headLocalPTS; 3939 { 3940 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3941 3942 assert(mMediaTimeTransformValid); 3943 3944 if (mMediaTimeTransform.a_to_b_denom == 0) { 3945 // the transform represents a pause, so yield silence 3946 timedYieldSilence(buffer->frameCount, buffer); 3947 return NO_ERROR; 3948 } 3949 3950 int64_t transformedPTS; 3951 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3952 &transformedPTS)) { 3953 // the transform failed. this shouldn't happen, but if it does 3954 // then just drop this buffer 3955 ALOGW("timedGetNextBuffer transform failed"); 3956 buffer->raw = 0; 3957 buffer->frameCount = 0; 3958 mTimedBufferQueue.removeAt(0); 3959 return NO_ERROR; 3960 } 3961 3962 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3963 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3964 &headLocalPTS)) { 3965 buffer->raw = 0; 3966 buffer->frameCount = 0; 3967 return INVALID_OPERATION; 3968 } 3969 } else { 3970 headLocalPTS = transformedPTS; 3971 } 3972 } 3973 3974 // adjust the head buffer's PTS to reflect the portion of the head buffer 3975 // that has already been consumed 3976 int64_t effectivePTS = headLocalPTS + 3977 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3978 3979 // Calculate the delta in samples between the head of the input buffer 3980 // queue and the start of the next output buffer that will be written. 3981 // If the transformation fails because of over or underflow, it means 3982 // that the sample's position in the output stream is so far out of 3983 // whack that it should just be dropped. 3984 int64_t sampleDelta; 3985 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3986 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3987 mTimedBufferQueue.removeAt(0); 3988 continue; 3989 } 3990 if (!mLocalTimeToSampleTransform.doForwardTransform( 3991 (effectivePTS - pts) << 32, &sampleDelta)) { 3992 ALOGV("*** too late during sample rate transform: dropped buffer"); 3993 mTimedBufferQueue.removeAt(0); 3994 continue; 3995 } 3996 3997 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 3998 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 3999 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4000 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4001 4002 // if the delta between the ideal placement for the next input sample and 4003 // the current output position is within this threshold, then we will 4004 // concatenate the next input samples to the previous output 4005 const int64_t kSampleContinuityThreshold = 4006 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4007 4008 // if this is the first buffer of audio that we're emitting from this track 4009 // then it should be almost exactly on time. 4010 const int64_t kSampleStartupThreshold = 1LL << 32; 4011 4012 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4013 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4014 // the next input is close enough to being on time, so concatenate it 4015 // with the last output 4016 timedYieldSamples(buffer); 4017 4018 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4019 return NO_ERROR; 4020 } else if (sampleDelta > 0) { 4021 // the gap between the current output position and the proper start of 4022 // the next input sample is too big, so fill it with silence 4023 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4024 4025 timedYieldSilence(framesUntilNextInput, buffer); 4026 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4027 return NO_ERROR; 4028 } else { 4029 // the next input sample is late 4030 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4031 size_t onTimeSamplePosition = 4032 head.position() + lateFrames * mCblk->frameSize; 4033 4034 if (onTimeSamplePosition > head.buffer()->size()) { 4035 // all the remaining samples in the head are too late, so 4036 // drop it and move on 4037 ALOGV("*** too late: dropped buffer"); 4038 mTimedBufferQueue.removeAt(0); 4039 continue; 4040 } else { 4041 // skip over the late samples 4042 head.setPosition(onTimeSamplePosition); 4043 4044 // yield the available samples 4045 timedYieldSamples(buffer); 4046 4047 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4048 return NO_ERROR; 4049 } 4050 } 4051 } 4052} 4053 4054// Yield samples from the timed buffer queue head up to the given output 4055// buffer's capacity. 4056// 4057// Caller must hold mTimedBufferQueueLock 4058void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4059 AudioBufferProvider::Buffer* buffer) { 4060 4061 const TimedBuffer& head = mTimedBufferQueue[0]; 4062 4063 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4064 head.position()); 4065 4066 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4067 mCblk->frameSize); 4068 size_t framesRequested = buffer->frameCount; 4069 buffer->frameCount = min(framesLeftInHead, framesRequested); 4070 4071 mTimedAudioOutputOnTime = true; 4072} 4073 4074// Yield samples of silence up to the given output buffer's capacity 4075// 4076// Caller must hold mTimedBufferQueueLock 4077void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4078 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4079 4080 // lazily allocate a buffer filled with silence 4081 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4082 delete [] mTimedSilenceBuffer; 4083 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4084 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4085 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4086 } 4087 4088 buffer->raw = mTimedSilenceBuffer; 4089 size_t framesRequested = buffer->frameCount; 4090 buffer->frameCount = min(numFrames, framesRequested); 4091 4092 mTimedAudioOutputOnTime = false; 4093} 4094 4095void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4096 AudioBufferProvider::Buffer* buffer) { 4097 4098 Mutex::Autolock _l(mTimedBufferQueueLock); 4099 4100 // If the buffer which was just released is part of the buffer at the head 4101 // of the queue, be sure to update the amt of the buffer which has been 4102 // consumed. If the buffer being returned is not part of the head of the 4103 // queue, its either because the buffer is part of the silence buffer, or 4104 // because the head of the timed queue was trimmed after the mixer called 4105 // getNextBuffer but before the mixer called releaseBuffer. 4106 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4107 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4108 4109 void* start = head.buffer()->pointer(); 4110 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4111 4112 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4113 head.setPosition(head.position() + 4114 (buffer->frameCount * mCblk->frameSize)); 4115 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4116 mTimedBufferQueue.removeAt(0); 4117 } 4118 } 4119 } 4120 4121 buffer->raw = 0; 4122 buffer->frameCount = 0; 4123} 4124 4125uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4126 Mutex::Autolock _l(mTimedBufferQueueLock); 4127 4128 uint32_t frames = 0; 4129 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4130 const TimedBuffer& tb = mTimedBufferQueue[i]; 4131 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4132 } 4133 4134 return frames; 4135} 4136 4137AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4138 : mPTS(0), mPosition(0) {} 4139 4140AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4141 const sp<IMemory>& buffer, int64_t pts) 4142 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4143 4144// ---------------------------------------------------------------------------- 4145 4146// RecordTrack constructor must be called with AudioFlinger::mLock held 4147AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4148 RecordThread *thread, 4149 const sp<Client>& client, 4150 uint32_t sampleRate, 4151 audio_format_t format, 4152 uint32_t channelMask, 4153 int frameCount, 4154 uint32_t flags, 4155 int sessionId) 4156 : TrackBase(thread, client, sampleRate, format, 4157 channelMask, frameCount, flags, 0, sessionId), 4158 mOverflow(false) 4159{ 4160 if (mCblk != NULL) { 4161 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4162 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4163 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4164 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4165 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4166 } else { 4167 mCblk->frameSize = sizeof(int8_t); 4168 } 4169 } 4170} 4171 4172AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4173{ 4174 sp<ThreadBase> thread = mThread.promote(); 4175 if (thread != 0) { 4176 AudioSystem::releaseInput(thread->id()); 4177 } 4178} 4179 4180status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4181{ 4182 audio_track_cblk_t* cblk = this->cblk(); 4183 uint32_t framesAvail; 4184 uint32_t framesReq = buffer->frameCount; 4185 4186 // Check if last stepServer failed, try to step now 4187 if (mFlags & TrackBase::STEPSERVER_FAILED) { 4188 if (!step()) goto getNextBuffer_exit; 4189 ALOGV("stepServer recovered"); 4190 mFlags &= ~TrackBase::STEPSERVER_FAILED; 4191 } 4192 4193 framesAvail = cblk->framesAvailable_l(); 4194 4195 if (CC_LIKELY(framesAvail)) { 4196 uint32_t s = cblk->server; 4197 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4198 4199 if (framesReq > framesAvail) { 4200 framesReq = framesAvail; 4201 } 4202 if (s + framesReq > bufferEnd) { 4203 framesReq = bufferEnd - s; 4204 } 4205 4206 buffer->raw = getBuffer(s, framesReq); 4207 if (buffer->raw == NULL) goto getNextBuffer_exit; 4208 4209 buffer->frameCount = framesReq; 4210 return NO_ERROR; 4211 } 4212 4213getNextBuffer_exit: 4214 buffer->raw = NULL; 4215 buffer->frameCount = 0; 4216 return NOT_ENOUGH_DATA; 4217} 4218 4219status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4220{ 4221 sp<ThreadBase> thread = mThread.promote(); 4222 if (thread != 0) { 4223 RecordThread *recordThread = (RecordThread *)thread.get(); 4224 return recordThread->start(this, tid); 4225 } else { 4226 return BAD_VALUE; 4227 } 4228} 4229 4230void AudioFlinger::RecordThread::RecordTrack::stop() 4231{ 4232 sp<ThreadBase> thread = mThread.promote(); 4233 if (thread != 0) { 4234 RecordThread *recordThread = (RecordThread *)thread.get(); 4235 recordThread->stop(this); 4236 TrackBase::reset(); 4237 // Force overerrun condition to avoid false overrun callback until first data is 4238 // read from buffer 4239 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4240 } 4241} 4242 4243void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4244{ 4245 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4246 (mClient == 0) ? getpid_cached : mClient->pid(), 4247 mFormat, 4248 mChannelMask, 4249 mSessionId, 4250 mFrameCount, 4251 mState, 4252 mCblk->sampleRate, 4253 mCblk->server, 4254 mCblk->user); 4255} 4256 4257 4258// ---------------------------------------------------------------------------- 4259 4260AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4261 PlaybackThread *playbackThread, 4262 DuplicatingThread *sourceThread, 4263 uint32_t sampleRate, 4264 audio_format_t format, 4265 uint32_t channelMask, 4266 int frameCount) 4267 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4268 mActive(false), mSourceThread(sourceThread) 4269{ 4270 4271 if (mCblk != NULL) { 4272 mCblk->flags |= CBLK_DIRECTION_OUT; 4273 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4274 mOutBuffer.frameCount = 0; 4275 playbackThread->mTracks.add(this); 4276 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4277 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4278 mCblk, mBuffer, mCblk->buffers, 4279 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4280 } else { 4281 ALOGW("Error creating output track on thread %p", playbackThread); 4282 } 4283} 4284 4285AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4286{ 4287 clearBufferQueue(); 4288} 4289 4290status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4291{ 4292 status_t status = Track::start(tid); 4293 if (status != NO_ERROR) { 4294 return status; 4295 } 4296 4297 mActive = true; 4298 mRetryCount = 127; 4299 return status; 4300} 4301 4302void AudioFlinger::PlaybackThread::OutputTrack::stop() 4303{ 4304 Track::stop(); 4305 clearBufferQueue(); 4306 mOutBuffer.frameCount = 0; 4307 mActive = false; 4308} 4309 4310bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4311{ 4312 Buffer *pInBuffer; 4313 Buffer inBuffer; 4314 uint32_t channelCount = mChannelCount; 4315 bool outputBufferFull = false; 4316 inBuffer.frameCount = frames; 4317 inBuffer.i16 = data; 4318 4319 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4320 4321 if (!mActive && frames != 0) { 4322 start(0); 4323 sp<ThreadBase> thread = mThread.promote(); 4324 if (thread != 0) { 4325 MixerThread *mixerThread = (MixerThread *)thread.get(); 4326 if (mCblk->frameCount > frames){ 4327 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4328 uint32_t startFrames = (mCblk->frameCount - frames); 4329 pInBuffer = new Buffer; 4330 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4331 pInBuffer->frameCount = startFrames; 4332 pInBuffer->i16 = pInBuffer->mBuffer; 4333 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4334 mBufferQueue.add(pInBuffer); 4335 } else { 4336 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4337 } 4338 } 4339 } 4340 } 4341 4342 while (waitTimeLeftMs) { 4343 // First write pending buffers, then new data 4344 if (mBufferQueue.size()) { 4345 pInBuffer = mBufferQueue.itemAt(0); 4346 } else { 4347 pInBuffer = &inBuffer; 4348 } 4349 4350 if (pInBuffer->frameCount == 0) { 4351 break; 4352 } 4353 4354 if (mOutBuffer.frameCount == 0) { 4355 mOutBuffer.frameCount = pInBuffer->frameCount; 4356 nsecs_t startTime = systemTime(); 4357 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4358 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4359 outputBufferFull = true; 4360 break; 4361 } 4362 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4363 if (waitTimeLeftMs >= waitTimeMs) { 4364 waitTimeLeftMs -= waitTimeMs; 4365 } else { 4366 waitTimeLeftMs = 0; 4367 } 4368 } 4369 4370 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4371 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4372 mCblk->stepUser(outFrames); 4373 pInBuffer->frameCount -= outFrames; 4374 pInBuffer->i16 += outFrames * channelCount; 4375 mOutBuffer.frameCount -= outFrames; 4376 mOutBuffer.i16 += outFrames * channelCount; 4377 4378 if (pInBuffer->frameCount == 0) { 4379 if (mBufferQueue.size()) { 4380 mBufferQueue.removeAt(0); 4381 delete [] pInBuffer->mBuffer; 4382 delete pInBuffer; 4383 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4384 } else { 4385 break; 4386 } 4387 } 4388 } 4389 4390 // If we could not write all frames, allocate a buffer and queue it for next time. 4391 if (inBuffer.frameCount) { 4392 sp<ThreadBase> thread = mThread.promote(); 4393 if (thread != 0 && !thread->standby()) { 4394 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4395 pInBuffer = new Buffer; 4396 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4397 pInBuffer->frameCount = inBuffer.frameCount; 4398 pInBuffer->i16 = pInBuffer->mBuffer; 4399 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4400 mBufferQueue.add(pInBuffer); 4401 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4402 } else { 4403 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4404 } 4405 } 4406 } 4407 4408 // Calling write() with a 0 length buffer, means that no more data will be written: 4409 // If no more buffers are pending, fill output track buffer to make sure it is started 4410 // by output mixer. 4411 if (frames == 0 && mBufferQueue.size() == 0) { 4412 if (mCblk->user < mCblk->frameCount) { 4413 frames = mCblk->frameCount - mCblk->user; 4414 pInBuffer = new Buffer; 4415 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4416 pInBuffer->frameCount = frames; 4417 pInBuffer->i16 = pInBuffer->mBuffer; 4418 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4419 mBufferQueue.add(pInBuffer); 4420 } else if (mActive) { 4421 stop(); 4422 } 4423 } 4424 4425 return outputBufferFull; 4426} 4427 4428status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4429{ 4430 int active; 4431 status_t result; 4432 audio_track_cblk_t* cblk = mCblk; 4433 uint32_t framesReq = buffer->frameCount; 4434 4435// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4436 buffer->frameCount = 0; 4437 4438 uint32_t framesAvail = cblk->framesAvailable(); 4439 4440 4441 if (framesAvail == 0) { 4442 Mutex::Autolock _l(cblk->lock); 4443 goto start_loop_here; 4444 while (framesAvail == 0) { 4445 active = mActive; 4446 if (CC_UNLIKELY(!active)) { 4447 ALOGV("Not active and NO_MORE_BUFFERS"); 4448 return NO_MORE_BUFFERS; 4449 } 4450 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4451 if (result != NO_ERROR) { 4452 return NO_MORE_BUFFERS; 4453 } 4454 // read the server count again 4455 start_loop_here: 4456 framesAvail = cblk->framesAvailable_l(); 4457 } 4458 } 4459 4460// if (framesAvail < framesReq) { 4461// return NO_MORE_BUFFERS; 4462// } 4463 4464 if (framesReq > framesAvail) { 4465 framesReq = framesAvail; 4466 } 4467 4468 uint32_t u = cblk->user; 4469 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4470 4471 if (u + framesReq > bufferEnd) { 4472 framesReq = bufferEnd - u; 4473 } 4474 4475 buffer->frameCount = framesReq; 4476 buffer->raw = (void *)cblk->buffer(u); 4477 return NO_ERROR; 4478} 4479 4480 4481void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4482{ 4483 size_t size = mBufferQueue.size(); 4484 4485 for (size_t i = 0; i < size; i++) { 4486 Buffer *pBuffer = mBufferQueue.itemAt(i); 4487 delete [] pBuffer->mBuffer; 4488 delete pBuffer; 4489 } 4490 mBufferQueue.clear(); 4491} 4492 4493// ---------------------------------------------------------------------------- 4494 4495AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4496 : RefBase(), 4497 mAudioFlinger(audioFlinger), 4498 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4499 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4500 mPid(pid), 4501 mTimedTrackCount(0) 4502{ 4503 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4504} 4505 4506// Client destructor must be called with AudioFlinger::mLock held 4507AudioFlinger::Client::~Client() 4508{ 4509 mAudioFlinger->removeClient_l(mPid); 4510} 4511 4512sp<MemoryDealer> AudioFlinger::Client::heap() const 4513{ 4514 return mMemoryDealer; 4515} 4516 4517// Reserve one of the limited slots for a timed audio track associated 4518// with this client 4519bool AudioFlinger::Client::reserveTimedTrack() 4520{ 4521 const int kMaxTimedTracksPerClient = 4; 4522 4523 Mutex::Autolock _l(mTimedTrackLock); 4524 4525 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4526 ALOGW("can not create timed track - pid %d has exceeded the limit", 4527 mPid); 4528 return false; 4529 } 4530 4531 mTimedTrackCount++; 4532 return true; 4533} 4534 4535// Release a slot for a timed audio track 4536void AudioFlinger::Client::releaseTimedTrack() 4537{ 4538 Mutex::Autolock _l(mTimedTrackLock); 4539 mTimedTrackCount--; 4540} 4541 4542// ---------------------------------------------------------------------------- 4543 4544AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4545 const sp<IAudioFlingerClient>& client, 4546 pid_t pid) 4547 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4548{ 4549} 4550 4551AudioFlinger::NotificationClient::~NotificationClient() 4552{ 4553} 4554 4555void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4556{ 4557 sp<NotificationClient> keep(this); 4558 mAudioFlinger->removeNotificationClient(mPid); 4559} 4560 4561// ---------------------------------------------------------------------------- 4562 4563AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4564 : BnAudioTrack(), 4565 mTrack(track) 4566{ 4567} 4568 4569AudioFlinger::TrackHandle::~TrackHandle() { 4570 // just stop the track on deletion, associated resources 4571 // will be freed from the main thread once all pending buffers have 4572 // been played. Unless it's not in the active track list, in which 4573 // case we free everything now... 4574 mTrack->destroy(); 4575} 4576 4577sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4578 return mTrack->getCblk(); 4579} 4580 4581status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4582 return mTrack->start(tid); 4583} 4584 4585void AudioFlinger::TrackHandle::stop() { 4586 mTrack->stop(); 4587} 4588 4589void AudioFlinger::TrackHandle::flush() { 4590 mTrack->flush(); 4591} 4592 4593void AudioFlinger::TrackHandle::mute(bool e) { 4594 mTrack->mute(e); 4595} 4596 4597void AudioFlinger::TrackHandle::pause() { 4598 mTrack->pause(); 4599} 4600 4601status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4602{ 4603 return mTrack->attachAuxEffect(EffectId); 4604} 4605 4606status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4607 sp<IMemory>* buffer) { 4608 if (!mTrack->isTimedTrack()) 4609 return INVALID_OPERATION; 4610 4611 PlaybackThread::TimedTrack* tt = 4612 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4613 return tt->allocateTimedBuffer(size, buffer); 4614} 4615 4616status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4617 int64_t pts) { 4618 if (!mTrack->isTimedTrack()) 4619 return INVALID_OPERATION; 4620 4621 PlaybackThread::TimedTrack* tt = 4622 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4623 return tt->queueTimedBuffer(buffer, pts); 4624} 4625 4626status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4627 const LinearTransform& xform, int target) { 4628 4629 if (!mTrack->isTimedTrack()) 4630 return INVALID_OPERATION; 4631 4632 PlaybackThread::TimedTrack* tt = 4633 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4634 return tt->setMediaTimeTransform( 4635 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4636} 4637 4638status_t AudioFlinger::TrackHandle::onTransact( 4639 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4640{ 4641 return BnAudioTrack::onTransact(code, data, reply, flags); 4642} 4643 4644// ---------------------------------------------------------------------------- 4645 4646sp<IAudioRecord> AudioFlinger::openRecord( 4647 pid_t pid, 4648 audio_io_handle_t input, 4649 uint32_t sampleRate, 4650 audio_format_t format, 4651 uint32_t channelMask, 4652 int frameCount, 4653 uint32_t flags, 4654 int *sessionId, 4655 status_t *status) 4656{ 4657 sp<RecordThread::RecordTrack> recordTrack; 4658 sp<RecordHandle> recordHandle; 4659 sp<Client> client; 4660 status_t lStatus; 4661 RecordThread *thread; 4662 size_t inFrameCount; 4663 int lSessionId; 4664 4665 // check calling permissions 4666 if (!recordingAllowed()) { 4667 lStatus = PERMISSION_DENIED; 4668 goto Exit; 4669 } 4670 4671 // add client to list 4672 { // scope for mLock 4673 Mutex::Autolock _l(mLock); 4674 thread = checkRecordThread_l(input); 4675 if (thread == NULL) { 4676 lStatus = BAD_VALUE; 4677 goto Exit; 4678 } 4679 4680 client = registerPid_l(pid); 4681 4682 // If no audio session id is provided, create one here 4683 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4684 lSessionId = *sessionId; 4685 } else { 4686 lSessionId = nextUniqueId(); 4687 if (sessionId != NULL) { 4688 *sessionId = lSessionId; 4689 } 4690 } 4691 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4692 recordTrack = thread->createRecordTrack_l(client, 4693 sampleRate, 4694 format, 4695 channelMask, 4696 frameCount, 4697 flags, 4698 lSessionId, 4699 &lStatus); 4700 } 4701 if (lStatus != NO_ERROR) { 4702 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4703 // destructor is called by the TrackBase destructor with mLock held 4704 client.clear(); 4705 recordTrack.clear(); 4706 goto Exit; 4707 } 4708 4709 // return to handle to client 4710 recordHandle = new RecordHandle(recordTrack); 4711 lStatus = NO_ERROR; 4712 4713Exit: 4714 if (status) { 4715 *status = lStatus; 4716 } 4717 return recordHandle; 4718} 4719 4720// ---------------------------------------------------------------------------- 4721 4722AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4723 : BnAudioRecord(), 4724 mRecordTrack(recordTrack) 4725{ 4726} 4727 4728AudioFlinger::RecordHandle::~RecordHandle() { 4729 stop(); 4730} 4731 4732sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4733 return mRecordTrack->getCblk(); 4734} 4735 4736status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4737 ALOGV("RecordHandle::start()"); 4738 return mRecordTrack->start(tid); 4739} 4740 4741void AudioFlinger::RecordHandle::stop() { 4742 ALOGV("RecordHandle::stop()"); 4743 mRecordTrack->stop(); 4744} 4745 4746status_t AudioFlinger::RecordHandle::onTransact( 4747 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4748{ 4749 return BnAudioRecord::onTransact(code, data, reply, flags); 4750} 4751 4752// ---------------------------------------------------------------------------- 4753 4754AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4755 AudioStreamIn *input, 4756 uint32_t sampleRate, 4757 uint32_t channels, 4758 audio_io_handle_t id, 4759 uint32_t device) : 4760 ThreadBase(audioFlinger, id, device, RECORD), 4761 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4762 // mRsmpInIndex and mInputBytes set by readInputParameters() 4763 mReqChannelCount(popcount(channels)), 4764 mReqSampleRate(sampleRate) 4765 // mBytesRead is only meaningful while active, and so is cleared in start() 4766 // (but might be better to also clear here for dump?) 4767{ 4768 snprintf(mName, kNameLength, "AudioIn_%d", id); 4769 4770 readInputParameters(); 4771} 4772 4773 4774AudioFlinger::RecordThread::~RecordThread() 4775{ 4776 delete[] mRsmpInBuffer; 4777 delete mResampler; 4778 delete[] mRsmpOutBuffer; 4779} 4780 4781void AudioFlinger::RecordThread::onFirstRef() 4782{ 4783 run(mName, PRIORITY_URGENT_AUDIO); 4784} 4785 4786status_t AudioFlinger::RecordThread::readyToRun() 4787{ 4788 status_t status = initCheck(); 4789 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4790 return status; 4791} 4792 4793bool AudioFlinger::RecordThread::threadLoop() 4794{ 4795 AudioBufferProvider::Buffer buffer; 4796 sp<RecordTrack> activeTrack; 4797 Vector< sp<EffectChain> > effectChains; 4798 4799 nsecs_t lastWarning = 0; 4800 4801 acquireWakeLock(); 4802 4803 // start recording 4804 while (!exitPending()) { 4805 4806 processConfigEvents(); 4807 4808 { // scope for mLock 4809 Mutex::Autolock _l(mLock); 4810 checkForNewParameters_l(); 4811 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4812 if (!mStandby) { 4813 mInput->stream->common.standby(&mInput->stream->common); 4814 mStandby = true; 4815 } 4816 4817 if (exitPending()) break; 4818 4819 releaseWakeLock_l(); 4820 ALOGV("RecordThread: loop stopping"); 4821 // go to sleep 4822 mWaitWorkCV.wait(mLock); 4823 ALOGV("RecordThread: loop starting"); 4824 acquireWakeLock_l(); 4825 continue; 4826 } 4827 if (mActiveTrack != 0) { 4828 if (mActiveTrack->mState == TrackBase::PAUSING) { 4829 if (!mStandby) { 4830 mInput->stream->common.standby(&mInput->stream->common); 4831 mStandby = true; 4832 } 4833 mActiveTrack.clear(); 4834 mStartStopCond.broadcast(); 4835 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4836 if (mReqChannelCount != mActiveTrack->channelCount()) { 4837 mActiveTrack.clear(); 4838 mStartStopCond.broadcast(); 4839 } else if (mBytesRead != 0) { 4840 // record start succeeds only if first read from audio input 4841 // succeeds 4842 if (mBytesRead > 0) { 4843 mActiveTrack->mState = TrackBase::ACTIVE; 4844 } else { 4845 mActiveTrack.clear(); 4846 } 4847 mStartStopCond.broadcast(); 4848 } 4849 mStandby = false; 4850 } 4851 } 4852 lockEffectChains_l(effectChains); 4853 } 4854 4855 if (mActiveTrack != 0) { 4856 if (mActiveTrack->mState != TrackBase::ACTIVE && 4857 mActiveTrack->mState != TrackBase::RESUMING) { 4858 unlockEffectChains(effectChains); 4859 usleep(kRecordThreadSleepUs); 4860 continue; 4861 } 4862 for (size_t i = 0; i < effectChains.size(); i ++) { 4863 effectChains[i]->process_l(); 4864 } 4865 4866 buffer.frameCount = mFrameCount; 4867 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4868 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4869 size_t framesOut = buffer.frameCount; 4870 if (mResampler == NULL) { 4871 // no resampling 4872 while (framesOut) { 4873 size_t framesIn = mFrameCount - mRsmpInIndex; 4874 if (framesIn) { 4875 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4876 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4877 if (framesIn > framesOut) 4878 framesIn = framesOut; 4879 mRsmpInIndex += framesIn; 4880 framesOut -= framesIn; 4881 if ((int)mChannelCount == mReqChannelCount || 4882 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4883 memcpy(dst, src, framesIn * mFrameSize); 4884 } else { 4885 int16_t *src16 = (int16_t *)src; 4886 int16_t *dst16 = (int16_t *)dst; 4887 if (mChannelCount == 1) { 4888 while (framesIn--) { 4889 *dst16++ = *src16; 4890 *dst16++ = *src16++; 4891 } 4892 } else { 4893 while (framesIn--) { 4894 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4895 src16 += 2; 4896 } 4897 } 4898 } 4899 } 4900 if (framesOut && mFrameCount == mRsmpInIndex) { 4901 if (framesOut == mFrameCount && 4902 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4903 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4904 framesOut = 0; 4905 } else { 4906 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4907 mRsmpInIndex = 0; 4908 } 4909 if (mBytesRead < 0) { 4910 ALOGE("Error reading audio input"); 4911 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4912 // Force input into standby so that it tries to 4913 // recover at next read attempt 4914 mInput->stream->common.standby(&mInput->stream->common); 4915 usleep(kRecordThreadSleepUs); 4916 } 4917 mRsmpInIndex = mFrameCount; 4918 framesOut = 0; 4919 buffer.frameCount = 0; 4920 } 4921 } 4922 } 4923 } else { 4924 // resampling 4925 4926 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4927 // alter output frame count as if we were expecting stereo samples 4928 if (mChannelCount == 1 && mReqChannelCount == 1) { 4929 framesOut >>= 1; 4930 } 4931 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4932 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4933 // are 32 bit aligned which should be always true. 4934 if (mChannelCount == 2 && mReqChannelCount == 1) { 4935 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4936 // the resampler always outputs stereo samples: do post stereo to mono conversion 4937 int16_t *src = (int16_t *)mRsmpOutBuffer; 4938 int16_t *dst = buffer.i16; 4939 while (framesOut--) { 4940 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4941 src += 2; 4942 } 4943 } else { 4944 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4945 } 4946 4947 } 4948 mActiveTrack->releaseBuffer(&buffer); 4949 mActiveTrack->overflow(); 4950 } 4951 // client isn't retrieving buffers fast enough 4952 else { 4953 if (!mActiveTrack->setOverflow()) { 4954 nsecs_t now = systemTime(); 4955 if ((now - lastWarning) > kWarningThrottleNs) { 4956 ALOGW("RecordThread: buffer overflow"); 4957 lastWarning = now; 4958 } 4959 } 4960 // Release the processor for a while before asking for a new buffer. 4961 // This will give the application more chance to read from the buffer and 4962 // clear the overflow. 4963 usleep(kRecordThreadSleepUs); 4964 } 4965 } 4966 // enable changes in effect chain 4967 unlockEffectChains(effectChains); 4968 effectChains.clear(); 4969 } 4970 4971 if (!mStandby) { 4972 mInput->stream->common.standby(&mInput->stream->common); 4973 } 4974 mActiveTrack.clear(); 4975 4976 mStartStopCond.broadcast(); 4977 4978 releaseWakeLock(); 4979 4980 ALOGV("RecordThread %p exiting", this); 4981 return false; 4982} 4983 4984 4985sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4986 const sp<AudioFlinger::Client>& client, 4987 uint32_t sampleRate, 4988 audio_format_t format, 4989 int channelMask, 4990 int frameCount, 4991 uint32_t flags, 4992 int sessionId, 4993 status_t *status) 4994{ 4995 sp<RecordTrack> track; 4996 status_t lStatus; 4997 4998 lStatus = initCheck(); 4999 if (lStatus != NO_ERROR) { 5000 ALOGE("Audio driver not initialized."); 5001 goto Exit; 5002 } 5003 5004 { // scope for mLock 5005 Mutex::Autolock _l(mLock); 5006 5007 track = new RecordTrack(this, client, sampleRate, 5008 format, channelMask, frameCount, flags, sessionId); 5009 5010 if (track->getCblk() == 0) { 5011 lStatus = NO_MEMORY; 5012 goto Exit; 5013 } 5014 5015 mTrack = track.get(); 5016 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5017 bool suspend = audio_is_bluetooth_sco_device( 5018 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5019 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5020 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5021 } 5022 lStatus = NO_ERROR; 5023 5024Exit: 5025 if (status) { 5026 *status = lStatus; 5027 } 5028 return track; 5029} 5030 5031status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5032{ 5033 ALOGV("RecordThread::start tid=%d", tid); 5034 sp <ThreadBase> strongMe = this; 5035 status_t status = NO_ERROR; 5036 { 5037 AutoMutex lock(mLock); 5038 if (mActiveTrack != 0) { 5039 if (recordTrack != mActiveTrack.get()) { 5040 status = -EBUSY; 5041 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5042 mActiveTrack->mState = TrackBase::ACTIVE; 5043 } 5044 return status; 5045 } 5046 5047 recordTrack->mState = TrackBase::IDLE; 5048 mActiveTrack = recordTrack; 5049 mLock.unlock(); 5050 status_t status = AudioSystem::startInput(mId); 5051 mLock.lock(); 5052 if (status != NO_ERROR) { 5053 mActiveTrack.clear(); 5054 return status; 5055 } 5056 mRsmpInIndex = mFrameCount; 5057 mBytesRead = 0; 5058 if (mResampler != NULL) { 5059 mResampler->reset(); 5060 } 5061 mActiveTrack->mState = TrackBase::RESUMING; 5062 // signal thread to start 5063 ALOGV("Signal record thread"); 5064 mWaitWorkCV.signal(); 5065 // do not wait for mStartStopCond if exiting 5066 if (exitPending()) { 5067 mActiveTrack.clear(); 5068 status = INVALID_OPERATION; 5069 goto startError; 5070 } 5071 mStartStopCond.wait(mLock); 5072 if (mActiveTrack == 0) { 5073 ALOGV("Record failed to start"); 5074 status = BAD_VALUE; 5075 goto startError; 5076 } 5077 ALOGV("Record started OK"); 5078 return status; 5079 } 5080startError: 5081 AudioSystem::stopInput(mId); 5082 return status; 5083} 5084 5085void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5086 ALOGV("RecordThread::stop"); 5087 sp <ThreadBase> strongMe = this; 5088 { 5089 AutoMutex lock(mLock); 5090 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5091 mActiveTrack->mState = TrackBase::PAUSING; 5092 // do not wait for mStartStopCond if exiting 5093 if (exitPending()) { 5094 return; 5095 } 5096 mStartStopCond.wait(mLock); 5097 // if we have been restarted, recordTrack == mActiveTrack.get() here 5098 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5099 mLock.unlock(); 5100 AudioSystem::stopInput(mId); 5101 mLock.lock(); 5102 ALOGV("Record stopped OK"); 5103 } 5104 } 5105 } 5106} 5107 5108status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5109{ 5110 const size_t SIZE = 256; 5111 char buffer[SIZE]; 5112 String8 result; 5113 5114 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5115 result.append(buffer); 5116 5117 if (mActiveTrack != 0) { 5118 result.append("Active Track:\n"); 5119 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5120 mActiveTrack->dump(buffer, SIZE); 5121 result.append(buffer); 5122 5123 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5124 result.append(buffer); 5125 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5126 result.append(buffer); 5127 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5128 result.append(buffer); 5129 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5130 result.append(buffer); 5131 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5132 result.append(buffer); 5133 5134 5135 } else { 5136 result.append("No record client\n"); 5137 } 5138 write(fd, result.string(), result.size()); 5139 5140 dumpBase(fd, args); 5141 dumpEffectChains(fd, args); 5142 5143 return NO_ERROR; 5144} 5145 5146status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5147{ 5148 size_t framesReq = buffer->frameCount; 5149 size_t framesReady = mFrameCount - mRsmpInIndex; 5150 int channelCount; 5151 5152 if (framesReady == 0) { 5153 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5154 if (mBytesRead < 0) { 5155 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5156 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5157 // Force input into standby so that it tries to 5158 // recover at next read attempt 5159 mInput->stream->common.standby(&mInput->stream->common); 5160 usleep(kRecordThreadSleepUs); 5161 } 5162 buffer->raw = NULL; 5163 buffer->frameCount = 0; 5164 return NOT_ENOUGH_DATA; 5165 } 5166 mRsmpInIndex = 0; 5167 framesReady = mFrameCount; 5168 } 5169 5170 if (framesReq > framesReady) { 5171 framesReq = framesReady; 5172 } 5173 5174 if (mChannelCount == 1 && mReqChannelCount == 2) { 5175 channelCount = 1; 5176 } else { 5177 channelCount = 2; 5178 } 5179 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5180 buffer->frameCount = framesReq; 5181 return NO_ERROR; 5182} 5183 5184void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5185{ 5186 mRsmpInIndex += buffer->frameCount; 5187 buffer->frameCount = 0; 5188} 5189 5190bool AudioFlinger::RecordThread::checkForNewParameters_l() 5191{ 5192 bool reconfig = false; 5193 5194 while (!mNewParameters.isEmpty()) { 5195 status_t status = NO_ERROR; 5196 String8 keyValuePair = mNewParameters[0]; 5197 AudioParameter param = AudioParameter(keyValuePair); 5198 int value; 5199 audio_format_t reqFormat = mFormat; 5200 int reqSamplingRate = mReqSampleRate; 5201 int reqChannelCount = mReqChannelCount; 5202 5203 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5204 reqSamplingRate = value; 5205 reconfig = true; 5206 } 5207 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5208 reqFormat = (audio_format_t) value; 5209 reconfig = true; 5210 } 5211 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5212 reqChannelCount = popcount(value); 5213 reconfig = true; 5214 } 5215 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5216 // do not accept frame count changes if tracks are open as the track buffer 5217 // size depends on frame count and correct behavior would not be guaranteed 5218 // if frame count is changed after track creation 5219 if (mActiveTrack != 0) { 5220 status = INVALID_OPERATION; 5221 } else { 5222 reconfig = true; 5223 } 5224 } 5225 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5226 // forward device change to effects that have requested to be 5227 // aware of attached audio device. 5228 for (size_t i = 0; i < mEffectChains.size(); i++) { 5229 mEffectChains[i]->setDevice_l(value); 5230 } 5231 // store input device and output device but do not forward output device to audio HAL. 5232 // Note that status is ignored by the caller for output device 5233 // (see AudioFlinger::setParameters() 5234 if (value & AUDIO_DEVICE_OUT_ALL) { 5235 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5236 status = BAD_VALUE; 5237 } else { 5238 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5239 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5240 if (mTrack != NULL) { 5241 bool suspend = audio_is_bluetooth_sco_device( 5242 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5243 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5244 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5245 } 5246 } 5247 mDevice |= (uint32_t)value; 5248 } 5249 if (status == NO_ERROR) { 5250 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5251 if (status == INVALID_OPERATION) { 5252 mInput->stream->common.standby(&mInput->stream->common); 5253 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5254 } 5255 if (reconfig) { 5256 if (status == BAD_VALUE && 5257 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5258 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5259 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5260 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5261 (reqChannelCount < 3)) { 5262 status = NO_ERROR; 5263 } 5264 if (status == NO_ERROR) { 5265 readInputParameters(); 5266 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5267 } 5268 } 5269 } 5270 5271 mNewParameters.removeAt(0); 5272 5273 mParamStatus = status; 5274 mParamCond.signal(); 5275 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5276 // already timed out waiting for the status and will never signal the condition. 5277 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5278 } 5279 return reconfig; 5280} 5281 5282String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5283{ 5284 char *s; 5285 String8 out_s8 = String8(); 5286 5287 Mutex::Autolock _l(mLock); 5288 if (initCheck() != NO_ERROR) { 5289 return out_s8; 5290 } 5291 5292 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5293 out_s8 = String8(s); 5294 free(s); 5295 return out_s8; 5296} 5297 5298void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5299 AudioSystem::OutputDescriptor desc; 5300 void *param2 = NULL; 5301 5302 switch (event) { 5303 case AudioSystem::INPUT_OPENED: 5304 case AudioSystem::INPUT_CONFIG_CHANGED: 5305 desc.channels = mChannelMask; 5306 desc.samplingRate = mSampleRate; 5307 desc.format = mFormat; 5308 desc.frameCount = mFrameCount; 5309 desc.latency = 0; 5310 param2 = &desc; 5311 break; 5312 5313 case AudioSystem::INPUT_CLOSED: 5314 default: 5315 break; 5316 } 5317 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5318} 5319 5320void AudioFlinger::RecordThread::readInputParameters() 5321{ 5322 delete mRsmpInBuffer; 5323 // mRsmpInBuffer is always assigned a new[] below 5324 delete mRsmpOutBuffer; 5325 mRsmpOutBuffer = NULL; 5326 delete mResampler; 5327 mResampler = NULL; 5328 5329 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5330 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5331 mChannelCount = (uint16_t)popcount(mChannelMask); 5332 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5333 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5334 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5335 mFrameCount = mInputBytes / mFrameSize; 5336 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5337 5338 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5339 { 5340 int channelCount; 5341 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5342 // stereo to mono post process as the resampler always outputs stereo. 5343 if (mChannelCount == 1 && mReqChannelCount == 2) { 5344 channelCount = 1; 5345 } else { 5346 channelCount = 2; 5347 } 5348 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5349 mResampler->setSampleRate(mSampleRate); 5350 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5351 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5352 5353 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5354 if (mChannelCount == 1 && mReqChannelCount == 1) { 5355 mFrameCount >>= 1; 5356 } 5357 5358 } 5359 mRsmpInIndex = mFrameCount; 5360} 5361 5362unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5363{ 5364 Mutex::Autolock _l(mLock); 5365 if (initCheck() != NO_ERROR) { 5366 return 0; 5367 } 5368 5369 return mInput->stream->get_input_frames_lost(mInput->stream); 5370} 5371 5372uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5373{ 5374 Mutex::Autolock _l(mLock); 5375 uint32_t result = 0; 5376 if (getEffectChain_l(sessionId) != 0) { 5377 result = EFFECT_SESSION; 5378 } 5379 5380 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5381 result |= TRACK_SESSION; 5382 } 5383 5384 return result; 5385} 5386 5387AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5388{ 5389 Mutex::Autolock _l(mLock); 5390 return mTrack; 5391} 5392 5393AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5394{ 5395 Mutex::Autolock _l(mLock); 5396 return mInput; 5397} 5398 5399AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5400{ 5401 Mutex::Autolock _l(mLock); 5402 AudioStreamIn *input = mInput; 5403 mInput = NULL; 5404 return input; 5405} 5406 5407// this method must always be called either with ThreadBase mLock held or inside the thread loop 5408audio_stream_t* AudioFlinger::RecordThread::stream() 5409{ 5410 if (mInput == NULL) { 5411 return NULL; 5412 } 5413 return &mInput->stream->common; 5414} 5415 5416 5417// ---------------------------------------------------------------------------- 5418 5419audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5420 uint32_t *pSamplingRate, 5421 audio_format_t *pFormat, 5422 uint32_t *pChannels, 5423 uint32_t *pLatencyMs, 5424 uint32_t flags) 5425{ 5426 status_t status; 5427 PlaybackThread *thread = NULL; 5428 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5429 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5430 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5431 uint32_t channels = pChannels ? *pChannels : 0; 5432 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5433 audio_stream_out_t *outStream; 5434 audio_hw_device_t *outHwDev; 5435 5436 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5437 pDevices ? *pDevices : 0, 5438 samplingRate, 5439 format, 5440 channels, 5441 flags); 5442 5443 if (pDevices == NULL || *pDevices == 0) { 5444 return 0; 5445 } 5446 5447 Mutex::Autolock _l(mLock); 5448 5449 outHwDev = findSuitableHwDev_l(*pDevices); 5450 if (outHwDev == NULL) 5451 return 0; 5452 5453 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5454 &channels, &samplingRate, &outStream); 5455 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5456 outStream, 5457 samplingRate, 5458 format, 5459 channels, 5460 status); 5461 5462 mHardwareStatus = AUDIO_HW_IDLE; 5463 if (outStream != NULL) { 5464 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5465 audio_io_handle_t id = nextUniqueId(); 5466 5467 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5468 (format != AUDIO_FORMAT_PCM_16_BIT) || 5469 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5470 thread = new DirectOutputThread(this, output, id, *pDevices); 5471 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5472 } else { 5473 thread = new MixerThread(this, output, id, *pDevices); 5474 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5475 } 5476 mPlaybackThreads.add(id, thread); 5477 5478 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5479 if (pFormat != NULL) *pFormat = format; 5480 if (pChannels != NULL) *pChannels = channels; 5481 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5482 5483 // notify client processes of the new output creation 5484 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5485 return id; 5486 } 5487 5488 return 0; 5489} 5490 5491audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5492 audio_io_handle_t output2) 5493{ 5494 Mutex::Autolock _l(mLock); 5495 MixerThread *thread1 = checkMixerThread_l(output1); 5496 MixerThread *thread2 = checkMixerThread_l(output2); 5497 5498 if (thread1 == NULL || thread2 == NULL) { 5499 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5500 return 0; 5501 } 5502 5503 audio_io_handle_t id = nextUniqueId(); 5504 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5505 thread->addOutputTrack(thread2); 5506 mPlaybackThreads.add(id, thread); 5507 // notify client processes of the new output creation 5508 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5509 return id; 5510} 5511 5512status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5513{ 5514 // keep strong reference on the playback thread so that 5515 // it is not destroyed while exit() is executed 5516 sp <PlaybackThread> thread; 5517 { 5518 Mutex::Autolock _l(mLock); 5519 thread = checkPlaybackThread_l(output); 5520 if (thread == NULL) { 5521 return BAD_VALUE; 5522 } 5523 5524 ALOGV("closeOutput() %d", output); 5525 5526 if (thread->type() == ThreadBase::MIXER) { 5527 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5528 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5529 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5530 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5531 } 5532 } 5533 } 5534 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5535 mPlaybackThreads.removeItem(output); 5536 } 5537 thread->exit(); 5538 // The thread entity (active unit of execution) is no longer running here, 5539 // but the ThreadBase container still exists. 5540 5541 if (thread->type() != ThreadBase::DUPLICATING) { 5542 AudioStreamOut *out = thread->clearOutput(); 5543 assert(out != NULL); 5544 // from now on thread->mOutput is NULL 5545 out->hwDev->close_output_stream(out->hwDev, out->stream); 5546 delete out; 5547 } 5548 return NO_ERROR; 5549} 5550 5551status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5552{ 5553 Mutex::Autolock _l(mLock); 5554 PlaybackThread *thread = checkPlaybackThread_l(output); 5555 5556 if (thread == NULL) { 5557 return BAD_VALUE; 5558 } 5559 5560 ALOGV("suspendOutput() %d", output); 5561 thread->suspend(); 5562 5563 return NO_ERROR; 5564} 5565 5566status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5567{ 5568 Mutex::Autolock _l(mLock); 5569 PlaybackThread *thread = checkPlaybackThread_l(output); 5570 5571 if (thread == NULL) { 5572 return BAD_VALUE; 5573 } 5574 5575 ALOGV("restoreOutput() %d", output); 5576 5577 thread->restore(); 5578 5579 return NO_ERROR; 5580} 5581 5582audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5583 uint32_t *pSamplingRate, 5584 audio_format_t *pFormat, 5585 uint32_t *pChannels, 5586 audio_in_acoustics_t acoustics) 5587{ 5588 status_t status; 5589 RecordThread *thread = NULL; 5590 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5591 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5592 uint32_t channels = pChannels ? *pChannels : 0; 5593 uint32_t reqSamplingRate = samplingRate; 5594 audio_format_t reqFormat = format; 5595 uint32_t reqChannels = channels; 5596 audio_stream_in_t *inStream; 5597 audio_hw_device_t *inHwDev; 5598 5599 if (pDevices == NULL || *pDevices == 0) { 5600 return 0; 5601 } 5602 5603 Mutex::Autolock _l(mLock); 5604 5605 inHwDev = findSuitableHwDev_l(*pDevices); 5606 if (inHwDev == NULL) 5607 return 0; 5608 5609 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5610 &channels, &samplingRate, 5611 acoustics, 5612 &inStream); 5613 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5614 inStream, 5615 samplingRate, 5616 format, 5617 channels, 5618 acoustics, 5619 status); 5620 5621 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5622 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5623 // or stereo to mono conversions on 16 bit PCM inputs. 5624 if (inStream == NULL && status == BAD_VALUE && 5625 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5626 (samplingRate <= 2 * reqSamplingRate) && 5627 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5628 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5629 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5630 &channels, &samplingRate, 5631 acoustics, 5632 &inStream); 5633 } 5634 5635 if (inStream != NULL) { 5636 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5637 5638 audio_io_handle_t id = nextUniqueId(); 5639 // Start record thread 5640 // RecorThread require both input and output device indication to forward to audio 5641 // pre processing modules 5642 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5643 thread = new RecordThread(this, 5644 input, 5645 reqSamplingRate, 5646 reqChannels, 5647 id, 5648 device); 5649 mRecordThreads.add(id, thread); 5650 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5651 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5652 if (pFormat != NULL) *pFormat = format; 5653 if (pChannels != NULL) *pChannels = reqChannels; 5654 5655 input->stream->common.standby(&input->stream->common); 5656 5657 // notify client processes of the new input creation 5658 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5659 return id; 5660 } 5661 5662 return 0; 5663} 5664 5665status_t AudioFlinger::closeInput(audio_io_handle_t input) 5666{ 5667 // keep strong reference on the record thread so that 5668 // it is not destroyed while exit() is executed 5669 sp <RecordThread> thread; 5670 { 5671 Mutex::Autolock _l(mLock); 5672 thread = checkRecordThread_l(input); 5673 if (thread == NULL) { 5674 return BAD_VALUE; 5675 } 5676 5677 ALOGV("closeInput() %d", input); 5678 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5679 mRecordThreads.removeItem(input); 5680 } 5681 thread->exit(); 5682 // The thread entity (active unit of execution) is no longer running here, 5683 // but the ThreadBase container still exists. 5684 5685 AudioStreamIn *in = thread->clearInput(); 5686 assert(in != NULL); 5687 // from now on thread->mInput is NULL 5688 in->hwDev->close_input_stream(in->hwDev, in->stream); 5689 delete in; 5690 5691 return NO_ERROR; 5692} 5693 5694status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5695{ 5696 Mutex::Autolock _l(mLock); 5697 MixerThread *dstThread = checkMixerThread_l(output); 5698 if (dstThread == NULL) { 5699 ALOGW("setStreamOutput() bad output id %d", output); 5700 return BAD_VALUE; 5701 } 5702 5703 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5704 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5705 5706 dstThread->setStreamValid(stream, true); 5707 5708 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5709 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5710 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5711 MixerThread *srcThread = (MixerThread *)thread; 5712 srcThread->setStreamValid(stream, false); 5713 srcThread->invalidateTracks(stream); 5714 } 5715 } 5716 5717 return NO_ERROR; 5718} 5719 5720 5721int AudioFlinger::newAudioSessionId() 5722{ 5723 return nextUniqueId(); 5724} 5725 5726void AudioFlinger::acquireAudioSessionId(int audioSession) 5727{ 5728 Mutex::Autolock _l(mLock); 5729 pid_t caller = IPCThreadState::self()->getCallingPid(); 5730 ALOGV("acquiring %d from %d", audioSession, caller); 5731 size_t num = mAudioSessionRefs.size(); 5732 for (size_t i = 0; i< num; i++) { 5733 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5734 if (ref->sessionid == audioSession && ref->pid == caller) { 5735 ref->cnt++; 5736 ALOGV(" incremented refcount to %d", ref->cnt); 5737 return; 5738 } 5739 } 5740 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5741 ALOGV(" added new entry for %d", audioSession); 5742} 5743 5744void AudioFlinger::releaseAudioSessionId(int audioSession) 5745{ 5746 Mutex::Autolock _l(mLock); 5747 pid_t caller = IPCThreadState::self()->getCallingPid(); 5748 ALOGV("releasing %d from %d", audioSession, caller); 5749 size_t num = mAudioSessionRefs.size(); 5750 for (size_t i = 0; i< num; i++) { 5751 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5752 if (ref->sessionid == audioSession && ref->pid == caller) { 5753 ref->cnt--; 5754 ALOGV(" decremented refcount to %d", ref->cnt); 5755 if (ref->cnt == 0) { 5756 mAudioSessionRefs.removeAt(i); 5757 delete ref; 5758 purgeStaleEffects_l(); 5759 } 5760 return; 5761 } 5762 } 5763 ALOGW("session id %d not found for pid %d", audioSession, caller); 5764} 5765 5766void AudioFlinger::purgeStaleEffects_l() { 5767 5768 ALOGV("purging stale effects"); 5769 5770 Vector< sp<EffectChain> > chains; 5771 5772 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5773 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5774 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5775 sp<EffectChain> ec = t->mEffectChains[j]; 5776 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5777 chains.push(ec); 5778 } 5779 } 5780 } 5781 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5782 sp<RecordThread> t = mRecordThreads.valueAt(i); 5783 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5784 sp<EffectChain> ec = t->mEffectChains[j]; 5785 chains.push(ec); 5786 } 5787 } 5788 5789 for (size_t i = 0; i < chains.size(); i++) { 5790 sp<EffectChain> ec = chains[i]; 5791 int sessionid = ec->sessionId(); 5792 sp<ThreadBase> t = ec->mThread.promote(); 5793 if (t == 0) { 5794 continue; 5795 } 5796 size_t numsessionrefs = mAudioSessionRefs.size(); 5797 bool found = false; 5798 for (size_t k = 0; k < numsessionrefs; k++) { 5799 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5800 if (ref->sessionid == sessionid) { 5801 ALOGV(" session %d still exists for %d with %d refs", 5802 sessionid, ref->pid, ref->cnt); 5803 found = true; 5804 break; 5805 } 5806 } 5807 if (!found) { 5808 // remove all effects from the chain 5809 while (ec->mEffects.size()) { 5810 sp<EffectModule> effect = ec->mEffects[0]; 5811 effect->unPin(); 5812 Mutex::Autolock _l (t->mLock); 5813 t->removeEffect_l(effect); 5814 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5815 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5816 if (handle != 0) { 5817 handle->mEffect.clear(); 5818 if (handle->mHasControl && handle->mEnabled) { 5819 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5820 } 5821 } 5822 } 5823 AudioSystem::unregisterEffect(effect->id()); 5824 } 5825 } 5826 } 5827 return; 5828} 5829 5830// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5831AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5832{ 5833 return mPlaybackThreads.valueFor(output).get(); 5834} 5835 5836// checkMixerThread_l() must be called with AudioFlinger::mLock held 5837AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5838{ 5839 PlaybackThread *thread = checkPlaybackThread_l(output); 5840 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5841} 5842 5843// checkRecordThread_l() must be called with AudioFlinger::mLock held 5844AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5845{ 5846 return mRecordThreads.valueFor(input).get(); 5847} 5848 5849uint32_t AudioFlinger::nextUniqueId() 5850{ 5851 return android_atomic_inc(&mNextUniqueId); 5852} 5853 5854AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5855{ 5856 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5857 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5858 AudioStreamOut *output = thread->getOutput(); 5859 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5860 return thread; 5861 } 5862 } 5863 return NULL; 5864} 5865 5866uint32_t AudioFlinger::primaryOutputDevice_l() 5867{ 5868 PlaybackThread *thread = primaryPlaybackThread_l(); 5869 5870 if (thread == NULL) { 5871 return 0; 5872 } 5873 5874 return thread->device(); 5875} 5876 5877 5878// ---------------------------------------------------------------------------- 5879// Effect management 5880// ---------------------------------------------------------------------------- 5881 5882 5883status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5884{ 5885 Mutex::Autolock _l(mLock); 5886 return EffectQueryNumberEffects(numEffects); 5887} 5888 5889status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5890{ 5891 Mutex::Autolock _l(mLock); 5892 return EffectQueryEffect(index, descriptor); 5893} 5894 5895status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5896 effect_descriptor_t *descriptor) const 5897{ 5898 Mutex::Autolock _l(mLock); 5899 return EffectGetDescriptor(pUuid, descriptor); 5900} 5901 5902 5903sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5904 effect_descriptor_t *pDesc, 5905 const sp<IEffectClient>& effectClient, 5906 int32_t priority, 5907 audio_io_handle_t io, 5908 int sessionId, 5909 status_t *status, 5910 int *id, 5911 int *enabled) 5912{ 5913 status_t lStatus = NO_ERROR; 5914 sp<EffectHandle> handle; 5915 effect_descriptor_t desc; 5916 5917 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5918 pid, effectClient.get(), priority, sessionId, io); 5919 5920 if (pDesc == NULL) { 5921 lStatus = BAD_VALUE; 5922 goto Exit; 5923 } 5924 5925 // check audio settings permission for global effects 5926 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5927 lStatus = PERMISSION_DENIED; 5928 goto Exit; 5929 } 5930 5931 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5932 // that can only be created by audio policy manager (running in same process) 5933 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5934 lStatus = PERMISSION_DENIED; 5935 goto Exit; 5936 } 5937 5938 if (io == 0) { 5939 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5940 // output must be specified by AudioPolicyManager when using session 5941 // AUDIO_SESSION_OUTPUT_STAGE 5942 lStatus = BAD_VALUE; 5943 goto Exit; 5944 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5945 // if the output returned by getOutputForEffect() is removed before we lock the 5946 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5947 // and we will exit safely 5948 io = AudioSystem::getOutputForEffect(&desc); 5949 } 5950 } 5951 5952 { 5953 Mutex::Autolock _l(mLock); 5954 5955 5956 if (!EffectIsNullUuid(&pDesc->uuid)) { 5957 // if uuid is specified, request effect descriptor 5958 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5959 if (lStatus < 0) { 5960 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5961 goto Exit; 5962 } 5963 } else { 5964 // if uuid is not specified, look for an available implementation 5965 // of the required type in effect factory 5966 if (EffectIsNullUuid(&pDesc->type)) { 5967 ALOGW("createEffect() no effect type"); 5968 lStatus = BAD_VALUE; 5969 goto Exit; 5970 } 5971 uint32_t numEffects = 0; 5972 effect_descriptor_t d; 5973 d.flags = 0; // prevent compiler warning 5974 bool found = false; 5975 5976 lStatus = EffectQueryNumberEffects(&numEffects); 5977 if (lStatus < 0) { 5978 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5979 goto Exit; 5980 } 5981 for (uint32_t i = 0; i < numEffects; i++) { 5982 lStatus = EffectQueryEffect(i, &desc); 5983 if (lStatus < 0) { 5984 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5985 continue; 5986 } 5987 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5988 // If matching type found save effect descriptor. If the session is 5989 // 0 and the effect is not auxiliary, continue enumeration in case 5990 // an auxiliary version of this effect type is available 5991 found = true; 5992 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5993 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5994 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5995 break; 5996 } 5997 } 5998 } 5999 if (!found) { 6000 lStatus = BAD_VALUE; 6001 ALOGW("createEffect() effect not found"); 6002 goto Exit; 6003 } 6004 // For same effect type, chose auxiliary version over insert version if 6005 // connect to output mix (Compliance to OpenSL ES) 6006 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6007 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6008 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6009 } 6010 } 6011 6012 // Do not allow auxiliary effects on a session different from 0 (output mix) 6013 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6014 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6015 lStatus = INVALID_OPERATION; 6016 goto Exit; 6017 } 6018 6019 // check recording permission for visualizer 6020 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6021 !recordingAllowed()) { 6022 lStatus = PERMISSION_DENIED; 6023 goto Exit; 6024 } 6025 6026 // return effect descriptor 6027 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6028 6029 // If output is not specified try to find a matching audio session ID in one of the 6030 // output threads. 6031 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6032 // because of code checking output when entering the function. 6033 // Note: io is never 0 when creating an effect on an input 6034 if (io == 0) { 6035 // look for the thread where the specified audio session is present 6036 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6037 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6038 io = mPlaybackThreads.keyAt(i); 6039 break; 6040 } 6041 } 6042 if (io == 0) { 6043 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6044 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6045 io = mRecordThreads.keyAt(i); 6046 break; 6047 } 6048 } 6049 } 6050 // If no output thread contains the requested session ID, default to 6051 // first output. The effect chain will be moved to the correct output 6052 // thread when a track with the same session ID is created 6053 if (io == 0 && mPlaybackThreads.size()) { 6054 io = mPlaybackThreads.keyAt(0); 6055 } 6056 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6057 } 6058 ThreadBase *thread = checkRecordThread_l(io); 6059 if (thread == NULL) { 6060 thread = checkPlaybackThread_l(io); 6061 if (thread == NULL) { 6062 ALOGE("createEffect() unknown output thread"); 6063 lStatus = BAD_VALUE; 6064 goto Exit; 6065 } 6066 } 6067 6068 sp<Client> client = registerPid_l(pid); 6069 6070 // create effect on selected output thread 6071 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6072 &desc, enabled, &lStatus); 6073 if (handle != 0 && id != NULL) { 6074 *id = handle->id(); 6075 } 6076 } 6077 6078Exit: 6079 if(status) { 6080 *status = lStatus; 6081 } 6082 return handle; 6083} 6084 6085status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6086 audio_io_handle_t dstOutput) 6087{ 6088 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6089 sessionId, srcOutput, dstOutput); 6090 Mutex::Autolock _l(mLock); 6091 if (srcOutput == dstOutput) { 6092 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6093 return NO_ERROR; 6094 } 6095 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6096 if (srcThread == NULL) { 6097 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6098 return BAD_VALUE; 6099 } 6100 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6101 if (dstThread == NULL) { 6102 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6103 return BAD_VALUE; 6104 } 6105 6106 Mutex::Autolock _dl(dstThread->mLock); 6107 Mutex::Autolock _sl(srcThread->mLock); 6108 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6109 6110 return NO_ERROR; 6111} 6112 6113// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6114status_t AudioFlinger::moveEffectChain_l(int sessionId, 6115 AudioFlinger::PlaybackThread *srcThread, 6116 AudioFlinger::PlaybackThread *dstThread, 6117 bool reRegister) 6118{ 6119 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6120 sessionId, srcThread, dstThread); 6121 6122 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6123 if (chain == 0) { 6124 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6125 sessionId, srcThread); 6126 return INVALID_OPERATION; 6127 } 6128 6129 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6130 // so that a new chain is created with correct parameters when first effect is added. This is 6131 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6132 // removed. 6133 srcThread->removeEffectChain_l(chain); 6134 6135 // transfer all effects one by one so that new effect chain is created on new thread with 6136 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6137 audio_io_handle_t dstOutput = dstThread->id(); 6138 sp<EffectChain> dstChain; 6139 uint32_t strategy = 0; // prevent compiler warning 6140 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6141 while (effect != 0) { 6142 srcThread->removeEffect_l(effect); 6143 dstThread->addEffect_l(effect); 6144 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6145 if (effect->state() == EffectModule::ACTIVE || 6146 effect->state() == EffectModule::STOPPING) { 6147 effect->start(); 6148 } 6149 // if the move request is not received from audio policy manager, the effect must be 6150 // re-registered with the new strategy and output 6151 if (dstChain == 0) { 6152 dstChain = effect->chain().promote(); 6153 if (dstChain == 0) { 6154 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6155 srcThread->addEffect_l(effect); 6156 return NO_INIT; 6157 } 6158 strategy = dstChain->strategy(); 6159 } 6160 if (reRegister) { 6161 AudioSystem::unregisterEffect(effect->id()); 6162 AudioSystem::registerEffect(&effect->desc(), 6163 dstOutput, 6164 strategy, 6165 sessionId, 6166 effect->id()); 6167 } 6168 effect = chain->getEffectFromId_l(0); 6169 } 6170 6171 return NO_ERROR; 6172} 6173 6174 6175// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6176sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6177 const sp<AudioFlinger::Client>& client, 6178 const sp<IEffectClient>& effectClient, 6179 int32_t priority, 6180 int sessionId, 6181 effect_descriptor_t *desc, 6182 int *enabled, 6183 status_t *status 6184 ) 6185{ 6186 sp<EffectModule> effect; 6187 sp<EffectHandle> handle; 6188 status_t lStatus; 6189 sp<EffectChain> chain; 6190 bool chainCreated = false; 6191 bool effectCreated = false; 6192 bool effectRegistered = false; 6193 6194 lStatus = initCheck(); 6195 if (lStatus != NO_ERROR) { 6196 ALOGW("createEffect_l() Audio driver not initialized."); 6197 goto Exit; 6198 } 6199 6200 // Do not allow effects with session ID 0 on direct output or duplicating threads 6201 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6202 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6203 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6204 desc->name, sessionId); 6205 lStatus = BAD_VALUE; 6206 goto Exit; 6207 } 6208 // Only Pre processor effects are allowed on input threads and only on input threads 6209 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6210 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6211 desc->name, desc->flags, mType); 6212 lStatus = BAD_VALUE; 6213 goto Exit; 6214 } 6215 6216 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6217 6218 { // scope for mLock 6219 Mutex::Autolock _l(mLock); 6220 6221 // check for existing effect chain with the requested audio session 6222 chain = getEffectChain_l(sessionId); 6223 if (chain == 0) { 6224 // create a new chain for this session 6225 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6226 chain = new EffectChain(this, sessionId); 6227 addEffectChain_l(chain); 6228 chain->setStrategy(getStrategyForSession_l(sessionId)); 6229 chainCreated = true; 6230 } else { 6231 effect = chain->getEffectFromDesc_l(desc); 6232 } 6233 6234 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6235 6236 if (effect == 0) { 6237 int id = mAudioFlinger->nextUniqueId(); 6238 // Check CPU and memory usage 6239 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6240 if (lStatus != NO_ERROR) { 6241 goto Exit; 6242 } 6243 effectRegistered = true; 6244 // create a new effect module if none present in the chain 6245 effect = new EffectModule(this, chain, desc, id, sessionId); 6246 lStatus = effect->status(); 6247 if (lStatus != NO_ERROR) { 6248 goto Exit; 6249 } 6250 lStatus = chain->addEffect_l(effect); 6251 if (lStatus != NO_ERROR) { 6252 goto Exit; 6253 } 6254 effectCreated = true; 6255 6256 effect->setDevice(mDevice); 6257 effect->setMode(mAudioFlinger->getMode()); 6258 } 6259 // create effect handle and connect it to effect module 6260 handle = new EffectHandle(effect, client, effectClient, priority); 6261 lStatus = effect->addHandle(handle); 6262 if (enabled != NULL) { 6263 *enabled = (int)effect->isEnabled(); 6264 } 6265 } 6266 6267Exit: 6268 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6269 Mutex::Autolock _l(mLock); 6270 if (effectCreated) { 6271 chain->removeEffect_l(effect); 6272 } 6273 if (effectRegistered) { 6274 AudioSystem::unregisterEffect(effect->id()); 6275 } 6276 if (chainCreated) { 6277 removeEffectChain_l(chain); 6278 } 6279 handle.clear(); 6280 } 6281 6282 if(status) { 6283 *status = lStatus; 6284 } 6285 return handle; 6286} 6287 6288sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6289{ 6290 sp<EffectChain> chain = getEffectChain_l(sessionId); 6291 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6292} 6293 6294// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6295// PlaybackThread::mLock held 6296status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6297{ 6298 // check for existing effect chain with the requested audio session 6299 int sessionId = effect->sessionId(); 6300 sp<EffectChain> chain = getEffectChain_l(sessionId); 6301 bool chainCreated = false; 6302 6303 if (chain == 0) { 6304 // create a new chain for this session 6305 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6306 chain = new EffectChain(this, sessionId); 6307 addEffectChain_l(chain); 6308 chain->setStrategy(getStrategyForSession_l(sessionId)); 6309 chainCreated = true; 6310 } 6311 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6312 6313 if (chain->getEffectFromId_l(effect->id()) != 0) { 6314 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6315 this, effect->desc().name, chain.get()); 6316 return BAD_VALUE; 6317 } 6318 6319 status_t status = chain->addEffect_l(effect); 6320 if (status != NO_ERROR) { 6321 if (chainCreated) { 6322 removeEffectChain_l(chain); 6323 } 6324 return status; 6325 } 6326 6327 effect->setDevice(mDevice); 6328 effect->setMode(mAudioFlinger->getMode()); 6329 return NO_ERROR; 6330} 6331 6332void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6333 6334 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6335 effect_descriptor_t desc = effect->desc(); 6336 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6337 detachAuxEffect_l(effect->id()); 6338 } 6339 6340 sp<EffectChain> chain = effect->chain().promote(); 6341 if (chain != 0) { 6342 // remove effect chain if removing last effect 6343 if (chain->removeEffect_l(effect) == 0) { 6344 removeEffectChain_l(chain); 6345 } 6346 } else { 6347 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6348 } 6349} 6350 6351void AudioFlinger::ThreadBase::lockEffectChains_l( 6352 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6353{ 6354 effectChains = mEffectChains; 6355 for (size_t i = 0; i < mEffectChains.size(); i++) { 6356 mEffectChains[i]->lock(); 6357 } 6358} 6359 6360void AudioFlinger::ThreadBase::unlockEffectChains( 6361 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6362{ 6363 for (size_t i = 0; i < effectChains.size(); i++) { 6364 effectChains[i]->unlock(); 6365 } 6366} 6367 6368sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6369{ 6370 Mutex::Autolock _l(mLock); 6371 return getEffectChain_l(sessionId); 6372} 6373 6374sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6375{ 6376 size_t size = mEffectChains.size(); 6377 for (size_t i = 0; i < size; i++) { 6378 if (mEffectChains[i]->sessionId() == sessionId) { 6379 return mEffectChains[i]; 6380 } 6381 } 6382 return 0; 6383} 6384 6385void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6386{ 6387 Mutex::Autolock _l(mLock); 6388 size_t size = mEffectChains.size(); 6389 for (size_t i = 0; i < size; i++) { 6390 mEffectChains[i]->setMode_l(mode); 6391 } 6392} 6393 6394void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6395 const wp<EffectHandle>& handle, 6396 bool unpinIfLast) { 6397 6398 Mutex::Autolock _l(mLock); 6399 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6400 // delete the effect module if removing last handle on it 6401 if (effect->removeHandle(handle) == 0) { 6402 if (!effect->isPinned() || unpinIfLast) { 6403 removeEffect_l(effect); 6404 AudioSystem::unregisterEffect(effect->id()); 6405 } 6406 } 6407} 6408 6409status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6410{ 6411 int session = chain->sessionId(); 6412 int16_t *buffer = mMixBuffer; 6413 bool ownsBuffer = false; 6414 6415 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6416 if (session > 0) { 6417 // Only one effect chain can be present in direct output thread and it uses 6418 // the mix buffer as input 6419 if (mType != DIRECT) { 6420 size_t numSamples = mFrameCount * mChannelCount; 6421 buffer = new int16_t[numSamples]; 6422 memset(buffer, 0, numSamples * sizeof(int16_t)); 6423 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6424 ownsBuffer = true; 6425 } 6426 6427 // Attach all tracks with same session ID to this chain. 6428 for (size_t i = 0; i < mTracks.size(); ++i) { 6429 sp<Track> track = mTracks[i]; 6430 if (session == track->sessionId()) { 6431 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6432 track->setMainBuffer(buffer); 6433 chain->incTrackCnt(); 6434 } 6435 } 6436 6437 // indicate all active tracks in the chain 6438 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6439 sp<Track> track = mActiveTracks[i].promote(); 6440 if (track == 0) continue; 6441 if (session == track->sessionId()) { 6442 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6443 chain->incActiveTrackCnt(); 6444 } 6445 } 6446 } 6447 6448 chain->setInBuffer(buffer, ownsBuffer); 6449 chain->setOutBuffer(mMixBuffer); 6450 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6451 // chains list in order to be processed last as it contains output stage effects 6452 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6453 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6454 // after track specific effects and before output stage 6455 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6456 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6457 // Effect chain for other sessions are inserted at beginning of effect 6458 // chains list to be processed before output mix effects. Relative order between other 6459 // sessions is not important 6460 size_t size = mEffectChains.size(); 6461 size_t i = 0; 6462 for (i = 0; i < size; i++) { 6463 if (mEffectChains[i]->sessionId() < session) break; 6464 } 6465 mEffectChains.insertAt(chain, i); 6466 checkSuspendOnAddEffectChain_l(chain); 6467 6468 return NO_ERROR; 6469} 6470 6471size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6472{ 6473 int session = chain->sessionId(); 6474 6475 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6476 6477 for (size_t i = 0; i < mEffectChains.size(); i++) { 6478 if (chain == mEffectChains[i]) { 6479 mEffectChains.removeAt(i); 6480 // detach all active tracks from the chain 6481 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6482 sp<Track> track = mActiveTracks[i].promote(); 6483 if (track == 0) continue; 6484 if (session == track->sessionId()) { 6485 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6486 chain.get(), session); 6487 chain->decActiveTrackCnt(); 6488 } 6489 } 6490 6491 // detach all tracks with same session ID from this chain 6492 for (size_t i = 0; i < mTracks.size(); ++i) { 6493 sp<Track> track = mTracks[i]; 6494 if (session == track->sessionId()) { 6495 track->setMainBuffer(mMixBuffer); 6496 chain->decTrackCnt(); 6497 } 6498 } 6499 break; 6500 } 6501 } 6502 return mEffectChains.size(); 6503} 6504 6505status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6506 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6507{ 6508 Mutex::Autolock _l(mLock); 6509 return attachAuxEffect_l(track, EffectId); 6510} 6511 6512status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6513 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6514{ 6515 status_t status = NO_ERROR; 6516 6517 if (EffectId == 0) { 6518 track->setAuxBuffer(0, NULL); 6519 } else { 6520 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6521 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6522 if (effect != 0) { 6523 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6524 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6525 } else { 6526 status = INVALID_OPERATION; 6527 } 6528 } else { 6529 status = BAD_VALUE; 6530 } 6531 } 6532 return status; 6533} 6534 6535void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6536{ 6537 for (size_t i = 0; i < mTracks.size(); ++i) { 6538 sp<Track> track = mTracks[i]; 6539 if (track->auxEffectId() == effectId) { 6540 attachAuxEffect_l(track, 0); 6541 } 6542 } 6543} 6544 6545status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6546{ 6547 // only one chain per input thread 6548 if (mEffectChains.size() != 0) { 6549 return INVALID_OPERATION; 6550 } 6551 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6552 6553 chain->setInBuffer(NULL); 6554 chain->setOutBuffer(NULL); 6555 6556 checkSuspendOnAddEffectChain_l(chain); 6557 6558 mEffectChains.add(chain); 6559 6560 return NO_ERROR; 6561} 6562 6563size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6564{ 6565 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6566 ALOGW_IF(mEffectChains.size() != 1, 6567 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6568 chain.get(), mEffectChains.size(), this); 6569 if (mEffectChains.size() == 1) { 6570 mEffectChains.removeAt(0); 6571 } 6572 return 0; 6573} 6574 6575// ---------------------------------------------------------------------------- 6576// EffectModule implementation 6577// ---------------------------------------------------------------------------- 6578 6579#undef LOG_TAG 6580#define LOG_TAG "AudioFlinger::EffectModule" 6581 6582AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6583 const wp<AudioFlinger::EffectChain>& chain, 6584 effect_descriptor_t *desc, 6585 int id, 6586 int sessionId) 6587 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6588 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6589{ 6590 ALOGV("Constructor %p", this); 6591 int lStatus; 6592 if (thread == NULL) { 6593 return; 6594 } 6595 6596 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6597 6598 // create effect engine from effect factory 6599 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6600 6601 if (mStatus != NO_ERROR) { 6602 return; 6603 } 6604 lStatus = init(); 6605 if (lStatus < 0) { 6606 mStatus = lStatus; 6607 goto Error; 6608 } 6609 6610 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6611 mPinned = true; 6612 } 6613 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6614 return; 6615Error: 6616 EffectRelease(mEffectInterface); 6617 mEffectInterface = NULL; 6618 ALOGV("Constructor Error %d", mStatus); 6619} 6620 6621AudioFlinger::EffectModule::~EffectModule() 6622{ 6623 ALOGV("Destructor %p", this); 6624 if (mEffectInterface != NULL) { 6625 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6626 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6627 sp<ThreadBase> thread = mThread.promote(); 6628 if (thread != 0) { 6629 audio_stream_t *stream = thread->stream(); 6630 if (stream != NULL) { 6631 stream->remove_audio_effect(stream, mEffectInterface); 6632 } 6633 } 6634 } 6635 // release effect engine 6636 EffectRelease(mEffectInterface); 6637 } 6638} 6639 6640status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6641{ 6642 status_t status; 6643 6644 Mutex::Autolock _l(mLock); 6645 int priority = handle->priority(); 6646 size_t size = mHandles.size(); 6647 sp<EffectHandle> h; 6648 size_t i; 6649 for (i = 0; i < size; i++) { 6650 h = mHandles[i].promote(); 6651 if (h == 0) continue; 6652 if (h->priority() <= priority) break; 6653 } 6654 // if inserted in first place, move effect control from previous owner to this handle 6655 if (i == 0) { 6656 bool enabled = false; 6657 if (h != 0) { 6658 enabled = h->enabled(); 6659 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6660 } 6661 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6662 status = NO_ERROR; 6663 } else { 6664 status = ALREADY_EXISTS; 6665 } 6666 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6667 mHandles.insertAt(handle, i); 6668 return status; 6669} 6670 6671size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6672{ 6673 Mutex::Autolock _l(mLock); 6674 size_t size = mHandles.size(); 6675 size_t i; 6676 for (i = 0; i < size; i++) { 6677 if (mHandles[i] == handle) break; 6678 } 6679 if (i == size) { 6680 return size; 6681 } 6682 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6683 6684 bool enabled = false; 6685 EffectHandle *hdl = handle.unsafe_get(); 6686 if (hdl != NULL) { 6687 ALOGV("removeHandle() unsafe_get OK"); 6688 enabled = hdl->enabled(); 6689 } 6690 mHandles.removeAt(i); 6691 size = mHandles.size(); 6692 // if removed from first place, move effect control from this handle to next in line 6693 if (i == 0 && size != 0) { 6694 sp<EffectHandle> h = mHandles[0].promote(); 6695 if (h != 0) { 6696 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6697 } 6698 } 6699 6700 // Prevent calls to process() and other functions on effect interface from now on. 6701 // The effect engine will be released by the destructor when the last strong reference on 6702 // this object is released which can happen after next process is called. 6703 if (size == 0 && !mPinned) { 6704 mState = DESTROYED; 6705 } 6706 6707 return size; 6708} 6709 6710sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6711{ 6712 Mutex::Autolock _l(mLock); 6713 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6714} 6715 6716void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6717{ 6718 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6719 // keep a strong reference on this EffectModule to avoid calling the 6720 // destructor before we exit 6721 sp<EffectModule> keep(this); 6722 { 6723 sp<ThreadBase> thread = mThread.promote(); 6724 if (thread != 0) { 6725 thread->disconnectEffect(keep, handle, unpinIfLast); 6726 } 6727 } 6728} 6729 6730void AudioFlinger::EffectModule::updateState() { 6731 Mutex::Autolock _l(mLock); 6732 6733 switch (mState) { 6734 case RESTART: 6735 reset_l(); 6736 // FALL THROUGH 6737 6738 case STARTING: 6739 // clear auxiliary effect input buffer for next accumulation 6740 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6741 memset(mConfig.inputCfg.buffer.raw, 6742 0, 6743 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6744 } 6745 start_l(); 6746 mState = ACTIVE; 6747 break; 6748 case STOPPING: 6749 stop_l(); 6750 mDisableWaitCnt = mMaxDisableWaitCnt; 6751 mState = STOPPED; 6752 break; 6753 case STOPPED: 6754 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6755 // turn off sequence. 6756 if (--mDisableWaitCnt == 0) { 6757 reset_l(); 6758 mState = IDLE; 6759 } 6760 break; 6761 default: //IDLE , ACTIVE, DESTROYED 6762 break; 6763 } 6764} 6765 6766void AudioFlinger::EffectModule::process() 6767{ 6768 Mutex::Autolock _l(mLock); 6769 6770 if (mState == DESTROYED || mEffectInterface == NULL || 6771 mConfig.inputCfg.buffer.raw == NULL || 6772 mConfig.outputCfg.buffer.raw == NULL) { 6773 return; 6774 } 6775 6776 if (isProcessEnabled()) { 6777 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6778 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6779 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6780 mConfig.inputCfg.buffer.s32, 6781 mConfig.inputCfg.buffer.frameCount/2); 6782 } 6783 6784 // do the actual processing in the effect engine 6785 int ret = (*mEffectInterface)->process(mEffectInterface, 6786 &mConfig.inputCfg.buffer, 6787 &mConfig.outputCfg.buffer); 6788 6789 // force transition to IDLE state when engine is ready 6790 if (mState == STOPPED && ret == -ENODATA) { 6791 mDisableWaitCnt = 1; 6792 } 6793 6794 // clear auxiliary effect input buffer for next accumulation 6795 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6796 memset(mConfig.inputCfg.buffer.raw, 0, 6797 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6798 } 6799 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6800 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6801 // If an insert effect is idle and input buffer is different from output buffer, 6802 // accumulate input onto output 6803 sp<EffectChain> chain = mChain.promote(); 6804 if (chain != 0 && chain->activeTrackCnt() != 0) { 6805 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6806 int16_t *in = mConfig.inputCfg.buffer.s16; 6807 int16_t *out = mConfig.outputCfg.buffer.s16; 6808 for (size_t i = 0; i < frameCnt; i++) { 6809 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6810 } 6811 } 6812 } 6813} 6814 6815void AudioFlinger::EffectModule::reset_l() 6816{ 6817 if (mEffectInterface == NULL) { 6818 return; 6819 } 6820 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6821} 6822 6823status_t AudioFlinger::EffectModule::configure() 6824{ 6825 uint32_t channels; 6826 if (mEffectInterface == NULL) { 6827 return NO_INIT; 6828 } 6829 6830 sp<ThreadBase> thread = mThread.promote(); 6831 if (thread == 0) { 6832 return DEAD_OBJECT; 6833 } 6834 6835 // TODO: handle configuration of effects replacing track process 6836 if (thread->channelCount() == 1) { 6837 channels = AUDIO_CHANNEL_OUT_MONO; 6838 } else { 6839 channels = AUDIO_CHANNEL_OUT_STEREO; 6840 } 6841 6842 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6843 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6844 } else { 6845 mConfig.inputCfg.channels = channels; 6846 } 6847 mConfig.outputCfg.channels = channels; 6848 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6849 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6850 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6851 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6852 mConfig.inputCfg.bufferProvider.cookie = NULL; 6853 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6854 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6855 mConfig.outputCfg.bufferProvider.cookie = NULL; 6856 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6857 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6858 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6859 // Insert effect: 6860 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6861 // always overwrites output buffer: input buffer == output buffer 6862 // - in other sessions: 6863 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6864 // other effect: overwrites output buffer: input buffer == output buffer 6865 // Auxiliary effect: 6866 // accumulates in output buffer: input buffer != output buffer 6867 // Therefore: accumulate <=> input buffer != output buffer 6868 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6869 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6870 } else { 6871 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6872 } 6873 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6874 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6875 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6876 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6877 6878 ALOGV("configure() %p thread %p buffer %p framecount %d", 6879 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6880 6881 status_t cmdStatus; 6882 uint32_t size = sizeof(int); 6883 status_t status = (*mEffectInterface)->command(mEffectInterface, 6884 EFFECT_CMD_SET_CONFIG, 6885 sizeof(effect_config_t), 6886 &mConfig, 6887 &size, 6888 &cmdStatus); 6889 if (status == 0) { 6890 status = cmdStatus; 6891 } 6892 6893 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6894 (1000 * mConfig.outputCfg.buffer.frameCount); 6895 6896 return status; 6897} 6898 6899status_t AudioFlinger::EffectModule::init() 6900{ 6901 Mutex::Autolock _l(mLock); 6902 if (mEffectInterface == NULL) { 6903 return NO_INIT; 6904 } 6905 status_t cmdStatus; 6906 uint32_t size = sizeof(status_t); 6907 status_t status = (*mEffectInterface)->command(mEffectInterface, 6908 EFFECT_CMD_INIT, 6909 0, 6910 NULL, 6911 &size, 6912 &cmdStatus); 6913 if (status == 0) { 6914 status = cmdStatus; 6915 } 6916 return status; 6917} 6918 6919status_t AudioFlinger::EffectModule::start() 6920{ 6921 Mutex::Autolock _l(mLock); 6922 return start_l(); 6923} 6924 6925status_t AudioFlinger::EffectModule::start_l() 6926{ 6927 if (mEffectInterface == NULL) { 6928 return NO_INIT; 6929 } 6930 status_t cmdStatus; 6931 uint32_t size = sizeof(status_t); 6932 status_t status = (*mEffectInterface)->command(mEffectInterface, 6933 EFFECT_CMD_ENABLE, 6934 0, 6935 NULL, 6936 &size, 6937 &cmdStatus); 6938 if (status == 0) { 6939 status = cmdStatus; 6940 } 6941 if (status == 0 && 6942 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6943 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6944 sp<ThreadBase> thread = mThread.promote(); 6945 if (thread != 0) { 6946 audio_stream_t *stream = thread->stream(); 6947 if (stream != NULL) { 6948 stream->add_audio_effect(stream, mEffectInterface); 6949 } 6950 } 6951 } 6952 return status; 6953} 6954 6955status_t AudioFlinger::EffectModule::stop() 6956{ 6957 Mutex::Autolock _l(mLock); 6958 return stop_l(); 6959} 6960 6961status_t AudioFlinger::EffectModule::stop_l() 6962{ 6963 if (mEffectInterface == NULL) { 6964 return NO_INIT; 6965 } 6966 status_t cmdStatus; 6967 uint32_t size = sizeof(status_t); 6968 status_t status = (*mEffectInterface)->command(mEffectInterface, 6969 EFFECT_CMD_DISABLE, 6970 0, 6971 NULL, 6972 &size, 6973 &cmdStatus); 6974 if (status == 0) { 6975 status = cmdStatus; 6976 } 6977 if (status == 0 && 6978 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6979 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6980 sp<ThreadBase> thread = mThread.promote(); 6981 if (thread != 0) { 6982 audio_stream_t *stream = thread->stream(); 6983 if (stream != NULL) { 6984 stream->remove_audio_effect(stream, mEffectInterface); 6985 } 6986 } 6987 } 6988 return status; 6989} 6990 6991status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6992 uint32_t cmdSize, 6993 void *pCmdData, 6994 uint32_t *replySize, 6995 void *pReplyData) 6996{ 6997 Mutex::Autolock _l(mLock); 6998// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6999 7000 if (mState == DESTROYED || mEffectInterface == NULL) { 7001 return NO_INIT; 7002 } 7003 status_t status = (*mEffectInterface)->command(mEffectInterface, 7004 cmdCode, 7005 cmdSize, 7006 pCmdData, 7007 replySize, 7008 pReplyData); 7009 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7010 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7011 for (size_t i = 1; i < mHandles.size(); i++) { 7012 sp<EffectHandle> h = mHandles[i].promote(); 7013 if (h != 0) { 7014 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7015 } 7016 } 7017 } 7018 return status; 7019} 7020 7021status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7022{ 7023 7024 Mutex::Autolock _l(mLock); 7025 ALOGV("setEnabled %p enabled %d", this, enabled); 7026 7027 if (enabled != isEnabled()) { 7028 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7029 if (enabled && status != NO_ERROR) { 7030 return status; 7031 } 7032 7033 switch (mState) { 7034 // going from disabled to enabled 7035 case IDLE: 7036 mState = STARTING; 7037 break; 7038 case STOPPED: 7039 mState = RESTART; 7040 break; 7041 case STOPPING: 7042 mState = ACTIVE; 7043 break; 7044 7045 // going from enabled to disabled 7046 case RESTART: 7047 mState = STOPPED; 7048 break; 7049 case STARTING: 7050 mState = IDLE; 7051 break; 7052 case ACTIVE: 7053 mState = STOPPING; 7054 break; 7055 case DESTROYED: 7056 return NO_ERROR; // simply ignore as we are being destroyed 7057 } 7058 for (size_t i = 1; i < mHandles.size(); i++) { 7059 sp<EffectHandle> h = mHandles[i].promote(); 7060 if (h != 0) { 7061 h->setEnabled(enabled); 7062 } 7063 } 7064 } 7065 return NO_ERROR; 7066} 7067 7068bool AudioFlinger::EffectModule::isEnabled() const 7069{ 7070 switch (mState) { 7071 case RESTART: 7072 case STARTING: 7073 case ACTIVE: 7074 return true; 7075 case IDLE: 7076 case STOPPING: 7077 case STOPPED: 7078 case DESTROYED: 7079 default: 7080 return false; 7081 } 7082} 7083 7084bool AudioFlinger::EffectModule::isProcessEnabled() const 7085{ 7086 switch (mState) { 7087 case RESTART: 7088 case ACTIVE: 7089 case STOPPING: 7090 case STOPPED: 7091 return true; 7092 case IDLE: 7093 case STARTING: 7094 case DESTROYED: 7095 default: 7096 return false; 7097 } 7098} 7099 7100status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7101{ 7102 Mutex::Autolock _l(mLock); 7103 status_t status = NO_ERROR; 7104 7105 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7106 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7107 if (isProcessEnabled() && 7108 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7109 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7110 status_t cmdStatus; 7111 uint32_t volume[2]; 7112 uint32_t *pVolume = NULL; 7113 uint32_t size = sizeof(volume); 7114 volume[0] = *left; 7115 volume[1] = *right; 7116 if (controller) { 7117 pVolume = volume; 7118 } 7119 status = (*mEffectInterface)->command(mEffectInterface, 7120 EFFECT_CMD_SET_VOLUME, 7121 size, 7122 volume, 7123 &size, 7124 pVolume); 7125 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7126 *left = volume[0]; 7127 *right = volume[1]; 7128 } 7129 } 7130 return status; 7131} 7132 7133status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7134{ 7135 Mutex::Autolock _l(mLock); 7136 status_t status = NO_ERROR; 7137 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7138 // audio pre processing modules on RecordThread can receive both output and 7139 // input device indication in the same call 7140 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7141 if (dev) { 7142 status_t cmdStatus; 7143 uint32_t size = sizeof(status_t); 7144 7145 status = (*mEffectInterface)->command(mEffectInterface, 7146 EFFECT_CMD_SET_DEVICE, 7147 sizeof(uint32_t), 7148 &dev, 7149 &size, 7150 &cmdStatus); 7151 if (status == NO_ERROR) { 7152 status = cmdStatus; 7153 } 7154 } 7155 dev = device & AUDIO_DEVICE_IN_ALL; 7156 if (dev) { 7157 status_t cmdStatus; 7158 uint32_t size = sizeof(status_t); 7159 7160 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7161 EFFECT_CMD_SET_INPUT_DEVICE, 7162 sizeof(uint32_t), 7163 &dev, 7164 &size, 7165 &cmdStatus); 7166 if (status2 == NO_ERROR) { 7167 status2 = cmdStatus; 7168 } 7169 if (status == NO_ERROR) { 7170 status = status2; 7171 } 7172 } 7173 } 7174 return status; 7175} 7176 7177status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7178{ 7179 Mutex::Autolock _l(mLock); 7180 status_t status = NO_ERROR; 7181 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7182 status_t cmdStatus; 7183 uint32_t size = sizeof(status_t); 7184 status = (*mEffectInterface)->command(mEffectInterface, 7185 EFFECT_CMD_SET_AUDIO_MODE, 7186 sizeof(audio_mode_t), 7187 &mode, 7188 &size, 7189 &cmdStatus); 7190 if (status == NO_ERROR) { 7191 status = cmdStatus; 7192 } 7193 } 7194 return status; 7195} 7196 7197void AudioFlinger::EffectModule::setSuspended(bool suspended) 7198{ 7199 Mutex::Autolock _l(mLock); 7200 mSuspended = suspended; 7201} 7202 7203bool AudioFlinger::EffectModule::suspended() const 7204{ 7205 Mutex::Autolock _l(mLock); 7206 return mSuspended; 7207} 7208 7209status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7210{ 7211 const size_t SIZE = 256; 7212 char buffer[SIZE]; 7213 String8 result; 7214 7215 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7216 result.append(buffer); 7217 7218 bool locked = tryLock(mLock); 7219 // failed to lock - AudioFlinger is probably deadlocked 7220 if (!locked) { 7221 result.append("\t\tCould not lock Fx mutex:\n"); 7222 } 7223 7224 result.append("\t\tSession Status State Engine:\n"); 7225 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7226 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7227 result.append(buffer); 7228 7229 result.append("\t\tDescriptor:\n"); 7230 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7231 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7232 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7233 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7234 result.append(buffer); 7235 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7236 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7237 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7238 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7239 result.append(buffer); 7240 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7241 mDescriptor.apiVersion, 7242 mDescriptor.flags); 7243 result.append(buffer); 7244 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7245 mDescriptor.name); 7246 result.append(buffer); 7247 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7248 mDescriptor.implementor); 7249 result.append(buffer); 7250 7251 result.append("\t\t- Input configuration:\n"); 7252 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7253 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7254 (uint32_t)mConfig.inputCfg.buffer.raw, 7255 mConfig.inputCfg.buffer.frameCount, 7256 mConfig.inputCfg.samplingRate, 7257 mConfig.inputCfg.channels, 7258 mConfig.inputCfg.format); 7259 result.append(buffer); 7260 7261 result.append("\t\t- Output 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.outputCfg.buffer.raw, 7265 mConfig.outputCfg.buffer.frameCount, 7266 mConfig.outputCfg.samplingRate, 7267 mConfig.outputCfg.channels, 7268 mConfig.outputCfg.format); 7269 result.append(buffer); 7270 7271 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7272 result.append(buffer); 7273 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7274 for (size_t i = 0; i < mHandles.size(); ++i) { 7275 sp<EffectHandle> handle = mHandles[i].promote(); 7276 if (handle != 0) { 7277 handle->dump(buffer, SIZE); 7278 result.append(buffer); 7279 } 7280 } 7281 7282 result.append("\n"); 7283 7284 write(fd, result.string(), result.length()); 7285 7286 if (locked) { 7287 mLock.unlock(); 7288 } 7289 7290 return NO_ERROR; 7291} 7292 7293// ---------------------------------------------------------------------------- 7294// EffectHandle implementation 7295// ---------------------------------------------------------------------------- 7296 7297#undef LOG_TAG 7298#define LOG_TAG "AudioFlinger::EffectHandle" 7299 7300AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7301 const sp<AudioFlinger::Client>& client, 7302 const sp<IEffectClient>& effectClient, 7303 int32_t priority) 7304 : BnEffect(), 7305 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7306 mPriority(priority), mHasControl(false), mEnabled(false) 7307{ 7308 ALOGV("constructor %p", this); 7309 7310 if (client == 0) { 7311 return; 7312 } 7313 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7314 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7315 if (mCblkMemory != 0) { 7316 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7317 7318 if (mCblk != NULL) { 7319 new(mCblk) effect_param_cblk_t(); 7320 mBuffer = (uint8_t *)mCblk + bufOffset; 7321 } 7322 } else { 7323 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7324 return; 7325 } 7326} 7327 7328AudioFlinger::EffectHandle::~EffectHandle() 7329{ 7330 ALOGV("Destructor %p", this); 7331 disconnect(false); 7332 ALOGV("Destructor DONE %p", this); 7333} 7334 7335status_t AudioFlinger::EffectHandle::enable() 7336{ 7337 ALOGV("enable %p", this); 7338 if (!mHasControl) return INVALID_OPERATION; 7339 if (mEffect == 0) return DEAD_OBJECT; 7340 7341 if (mEnabled) { 7342 return NO_ERROR; 7343 } 7344 7345 mEnabled = true; 7346 7347 sp<ThreadBase> thread = mEffect->thread().promote(); 7348 if (thread != 0) { 7349 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7350 } 7351 7352 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7353 if (mEffect->suspended()) { 7354 return NO_ERROR; 7355 } 7356 7357 status_t status = mEffect->setEnabled(true); 7358 if (status != NO_ERROR) { 7359 if (thread != 0) { 7360 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7361 } 7362 mEnabled = false; 7363 } 7364 return status; 7365} 7366 7367status_t AudioFlinger::EffectHandle::disable() 7368{ 7369 ALOGV("disable %p", this); 7370 if (!mHasControl) return INVALID_OPERATION; 7371 if (mEffect == 0) return DEAD_OBJECT; 7372 7373 if (!mEnabled) { 7374 return NO_ERROR; 7375 } 7376 mEnabled = false; 7377 7378 if (mEffect->suspended()) { 7379 return NO_ERROR; 7380 } 7381 7382 status_t status = mEffect->setEnabled(false); 7383 7384 sp<ThreadBase> thread = mEffect->thread().promote(); 7385 if (thread != 0) { 7386 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7387 } 7388 7389 return status; 7390} 7391 7392void AudioFlinger::EffectHandle::disconnect() 7393{ 7394 disconnect(true); 7395} 7396 7397void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7398{ 7399 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7400 if (mEffect == 0) { 7401 return; 7402 } 7403 mEffect->disconnect(this, unpinIfLast); 7404 7405 if (mHasControl && mEnabled) { 7406 sp<ThreadBase> thread = mEffect->thread().promote(); 7407 if (thread != 0) { 7408 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7409 } 7410 } 7411 7412 // release sp on module => module destructor can be called now 7413 mEffect.clear(); 7414 if (mClient != 0) { 7415 if (mCblk != NULL) { 7416 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7417 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7418 } 7419 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7420 // Client destructor must run with AudioFlinger mutex locked 7421 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7422 mClient.clear(); 7423 } 7424} 7425 7426status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7427 uint32_t cmdSize, 7428 void *pCmdData, 7429 uint32_t *replySize, 7430 void *pReplyData) 7431{ 7432// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7433// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7434 7435 // only get parameter command is permitted for applications not controlling the effect 7436 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7437 return INVALID_OPERATION; 7438 } 7439 if (mEffect == 0) return DEAD_OBJECT; 7440 if (mClient == 0) return INVALID_OPERATION; 7441 7442 // handle commands that are not forwarded transparently to effect engine 7443 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7444 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7445 // no risk to block the whole media server process or mixer threads is we are stuck here 7446 Mutex::Autolock _l(mCblk->lock); 7447 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7448 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7449 mCblk->serverIndex = 0; 7450 mCblk->clientIndex = 0; 7451 return BAD_VALUE; 7452 } 7453 status_t status = NO_ERROR; 7454 while (mCblk->serverIndex < mCblk->clientIndex) { 7455 int reply; 7456 uint32_t rsize = sizeof(int); 7457 int *p = (int *)(mBuffer + mCblk->serverIndex); 7458 int size = *p++; 7459 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7460 ALOGW("command(): invalid parameter block size"); 7461 break; 7462 } 7463 effect_param_t *param = (effect_param_t *)p; 7464 if (param->psize == 0 || param->vsize == 0) { 7465 ALOGW("command(): null parameter or value size"); 7466 mCblk->serverIndex += size; 7467 continue; 7468 } 7469 uint32_t psize = sizeof(effect_param_t) + 7470 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7471 param->vsize; 7472 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7473 psize, 7474 p, 7475 &rsize, 7476 &reply); 7477 // stop at first error encountered 7478 if (ret != NO_ERROR) { 7479 status = ret; 7480 *(int *)pReplyData = reply; 7481 break; 7482 } else if (reply != NO_ERROR) { 7483 *(int *)pReplyData = reply; 7484 break; 7485 } 7486 mCblk->serverIndex += size; 7487 } 7488 mCblk->serverIndex = 0; 7489 mCblk->clientIndex = 0; 7490 return status; 7491 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7492 *(int *)pReplyData = NO_ERROR; 7493 return enable(); 7494 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7495 *(int *)pReplyData = NO_ERROR; 7496 return disable(); 7497 } 7498 7499 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7500} 7501 7502void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7503{ 7504 ALOGV("setControl %p control %d", this, hasControl); 7505 7506 mHasControl = hasControl; 7507 mEnabled = enabled; 7508 7509 if (signal && mEffectClient != 0) { 7510 mEffectClient->controlStatusChanged(hasControl); 7511 } 7512} 7513 7514void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7515 uint32_t cmdSize, 7516 void *pCmdData, 7517 uint32_t replySize, 7518 void *pReplyData) 7519{ 7520 if (mEffectClient != 0) { 7521 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7522 } 7523} 7524 7525 7526 7527void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7528{ 7529 if (mEffectClient != 0) { 7530 mEffectClient->enableStatusChanged(enabled); 7531 } 7532} 7533 7534status_t AudioFlinger::EffectHandle::onTransact( 7535 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7536{ 7537 return BnEffect::onTransact(code, data, reply, flags); 7538} 7539 7540 7541void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7542{ 7543 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7544 7545 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7546 (mClient == 0) ? getpid_cached : mClient->pid(), 7547 mPriority, 7548 mHasControl, 7549 !locked, 7550 mCblk ? mCblk->clientIndex : 0, 7551 mCblk ? mCblk->serverIndex : 0 7552 ); 7553 7554 if (locked) { 7555 mCblk->lock.unlock(); 7556 } 7557} 7558 7559#undef LOG_TAG 7560#define LOG_TAG "AudioFlinger::EffectChain" 7561 7562AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7563 int sessionId) 7564 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7565 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7566 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7567{ 7568 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7569 if (thread == NULL) { 7570 return; 7571 } 7572 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7573 thread->frameCount(); 7574} 7575 7576AudioFlinger::EffectChain::~EffectChain() 7577{ 7578 if (mOwnInBuffer) { 7579 delete mInBuffer; 7580 } 7581 7582} 7583 7584// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7585sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7586{ 7587 size_t size = mEffects.size(); 7588 7589 for (size_t i = 0; i < size; i++) { 7590 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7591 return mEffects[i]; 7592 } 7593 } 7594 return 0; 7595} 7596 7597// getEffectFromId_l() must be called with ThreadBase::mLock held 7598sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7599{ 7600 size_t size = mEffects.size(); 7601 7602 for (size_t i = 0; i < size; i++) { 7603 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7604 if (id == 0 || mEffects[i]->id() == id) { 7605 return mEffects[i]; 7606 } 7607 } 7608 return 0; 7609} 7610 7611// getEffectFromType_l() must be called with ThreadBase::mLock held 7612sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7613 const effect_uuid_t *type) 7614{ 7615 size_t size = mEffects.size(); 7616 7617 for (size_t i = 0; i < size; i++) { 7618 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7619 return mEffects[i]; 7620 } 7621 } 7622 return 0; 7623} 7624 7625// Must be called with EffectChain::mLock locked 7626void AudioFlinger::EffectChain::process_l() 7627{ 7628 sp<ThreadBase> thread = mThread.promote(); 7629 if (thread == 0) { 7630 ALOGW("process_l(): cannot promote mixer thread"); 7631 return; 7632 } 7633 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7634 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7635 // always process effects unless no more tracks are on the session and the effect tail 7636 // has been rendered 7637 bool doProcess = true; 7638 if (!isGlobalSession) { 7639 bool tracksOnSession = (trackCnt() != 0); 7640 7641 if (!tracksOnSession && mTailBufferCount == 0) { 7642 doProcess = false; 7643 } 7644 7645 if (activeTrackCnt() == 0) { 7646 // if no track is active and the effect tail has not been rendered, 7647 // the input buffer must be cleared here as the mixer process will not do it 7648 if (tracksOnSession || mTailBufferCount > 0) { 7649 size_t numSamples = thread->frameCount() * thread->channelCount(); 7650 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7651 if (mTailBufferCount > 0) { 7652 mTailBufferCount--; 7653 } 7654 } 7655 } 7656 } 7657 7658 size_t size = mEffects.size(); 7659 if (doProcess) { 7660 for (size_t i = 0; i < size; i++) { 7661 mEffects[i]->process(); 7662 } 7663 } 7664 for (size_t i = 0; i < size; i++) { 7665 mEffects[i]->updateState(); 7666 } 7667} 7668 7669// addEffect_l() must be called with PlaybackThread::mLock held 7670status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7671{ 7672 effect_descriptor_t desc = effect->desc(); 7673 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7674 7675 Mutex::Autolock _l(mLock); 7676 effect->setChain(this); 7677 sp<ThreadBase> thread = mThread.promote(); 7678 if (thread == 0) { 7679 return NO_INIT; 7680 } 7681 effect->setThread(thread); 7682 7683 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7684 // Auxiliary effects are inserted at the beginning of mEffects vector as 7685 // they are processed first and accumulated in chain input buffer 7686 mEffects.insertAt(effect, 0); 7687 7688 // the input buffer for auxiliary effect contains mono samples in 7689 // 32 bit format. This is to avoid saturation in AudoMixer 7690 // accumulation stage. Saturation is done in EffectModule::process() before 7691 // calling the process in effect engine 7692 size_t numSamples = thread->frameCount(); 7693 int32_t *buffer = new int32_t[numSamples]; 7694 memset(buffer, 0, numSamples * sizeof(int32_t)); 7695 effect->setInBuffer((int16_t *)buffer); 7696 // auxiliary effects output samples to chain input buffer for further processing 7697 // by insert effects 7698 effect->setOutBuffer(mInBuffer); 7699 } else { 7700 // Insert effects are inserted at the end of mEffects vector as they are processed 7701 // after track and auxiliary effects. 7702 // Insert effect order as a function of indicated preference: 7703 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7704 // another effect is present 7705 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7706 // last effect claiming first position 7707 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7708 // first effect claiming last position 7709 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7710 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7711 // already present 7712 7713 size_t size = mEffects.size(); 7714 size_t idx_insert = size; 7715 ssize_t idx_insert_first = -1; 7716 ssize_t idx_insert_last = -1; 7717 7718 for (size_t i = 0; i < size; i++) { 7719 effect_descriptor_t d = mEffects[i]->desc(); 7720 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7721 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7722 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7723 // check invalid effect chaining combinations 7724 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7725 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7726 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7727 return INVALID_OPERATION; 7728 } 7729 // remember position of first insert effect and by default 7730 // select this as insert position for new effect 7731 if (idx_insert == size) { 7732 idx_insert = i; 7733 } 7734 // remember position of last insert effect claiming 7735 // first position 7736 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7737 idx_insert_first = i; 7738 } 7739 // remember position of first insert effect claiming 7740 // last position 7741 if (iPref == EFFECT_FLAG_INSERT_LAST && 7742 idx_insert_last == -1) { 7743 idx_insert_last = i; 7744 } 7745 } 7746 } 7747 7748 // modify idx_insert from first position if needed 7749 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7750 if (idx_insert_last != -1) { 7751 idx_insert = idx_insert_last; 7752 } else { 7753 idx_insert = size; 7754 } 7755 } else { 7756 if (idx_insert_first != -1) { 7757 idx_insert = idx_insert_first + 1; 7758 } 7759 } 7760 7761 // always read samples from chain input buffer 7762 effect->setInBuffer(mInBuffer); 7763 7764 // if last effect in the chain, output samples to chain 7765 // output buffer, otherwise to chain input buffer 7766 if (idx_insert == size) { 7767 if (idx_insert != 0) { 7768 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7769 mEffects[idx_insert-1]->configure(); 7770 } 7771 effect->setOutBuffer(mOutBuffer); 7772 } else { 7773 effect->setOutBuffer(mInBuffer); 7774 } 7775 mEffects.insertAt(effect, idx_insert); 7776 7777 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7778 } 7779 effect->configure(); 7780 return NO_ERROR; 7781} 7782 7783// removeEffect_l() must be called with PlaybackThread::mLock held 7784size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7785{ 7786 Mutex::Autolock _l(mLock); 7787 size_t size = mEffects.size(); 7788 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7789 7790 for (size_t i = 0; i < size; i++) { 7791 if (effect == mEffects[i]) { 7792 // calling stop here will remove pre-processing effect from the audio HAL. 7793 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7794 // the middle of a read from audio HAL 7795 if (mEffects[i]->state() == EffectModule::ACTIVE || 7796 mEffects[i]->state() == EffectModule::STOPPING) { 7797 mEffects[i]->stop(); 7798 } 7799 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7800 delete[] effect->inBuffer(); 7801 } else { 7802 if (i == size - 1 && i != 0) { 7803 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7804 mEffects[i - 1]->configure(); 7805 } 7806 } 7807 mEffects.removeAt(i); 7808 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7809 break; 7810 } 7811 } 7812 7813 return mEffects.size(); 7814} 7815 7816// setDevice_l() must be called with PlaybackThread::mLock held 7817void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7818{ 7819 size_t size = mEffects.size(); 7820 for (size_t i = 0; i < size; i++) { 7821 mEffects[i]->setDevice(device); 7822 } 7823} 7824 7825// setMode_l() must be called with PlaybackThread::mLock held 7826void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7827{ 7828 size_t size = mEffects.size(); 7829 for (size_t i = 0; i < size; i++) { 7830 mEffects[i]->setMode(mode); 7831 } 7832} 7833 7834// setVolume_l() must be called with PlaybackThread::mLock held 7835bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7836{ 7837 uint32_t newLeft = *left; 7838 uint32_t newRight = *right; 7839 bool hasControl = false; 7840 int ctrlIdx = -1; 7841 size_t size = mEffects.size(); 7842 7843 // first update volume controller 7844 for (size_t i = size; i > 0; i--) { 7845 if (mEffects[i - 1]->isProcessEnabled() && 7846 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7847 ctrlIdx = i - 1; 7848 hasControl = true; 7849 break; 7850 } 7851 } 7852 7853 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7854 if (hasControl) { 7855 *left = mNewLeftVolume; 7856 *right = mNewRightVolume; 7857 } 7858 return hasControl; 7859 } 7860 7861 mVolumeCtrlIdx = ctrlIdx; 7862 mLeftVolume = newLeft; 7863 mRightVolume = newRight; 7864 7865 // second get volume update from volume controller 7866 if (ctrlIdx >= 0) { 7867 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7868 mNewLeftVolume = newLeft; 7869 mNewRightVolume = newRight; 7870 } 7871 // then indicate volume to all other effects in chain. 7872 // Pass altered volume to effects before volume controller 7873 // and requested volume to effects after controller 7874 uint32_t lVol = newLeft; 7875 uint32_t rVol = newRight; 7876 7877 for (size_t i = 0; i < size; i++) { 7878 if ((int)i == ctrlIdx) continue; 7879 // this also works for ctrlIdx == -1 when there is no volume controller 7880 if ((int)i > ctrlIdx) { 7881 lVol = *left; 7882 rVol = *right; 7883 } 7884 mEffects[i]->setVolume(&lVol, &rVol, false); 7885 } 7886 *left = newLeft; 7887 *right = newRight; 7888 7889 return hasControl; 7890} 7891 7892status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7893{ 7894 const size_t SIZE = 256; 7895 char buffer[SIZE]; 7896 String8 result; 7897 7898 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7899 result.append(buffer); 7900 7901 bool locked = tryLock(mLock); 7902 // failed to lock - AudioFlinger is probably deadlocked 7903 if (!locked) { 7904 result.append("\tCould not lock mutex:\n"); 7905 } 7906 7907 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7908 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7909 mEffects.size(), 7910 (uint32_t)mInBuffer, 7911 (uint32_t)mOutBuffer, 7912 mActiveTrackCnt); 7913 result.append(buffer); 7914 write(fd, result.string(), result.size()); 7915 7916 for (size_t i = 0; i < mEffects.size(); ++i) { 7917 sp<EffectModule> effect = mEffects[i]; 7918 if (effect != 0) { 7919 effect->dump(fd, args); 7920 } 7921 } 7922 7923 if (locked) { 7924 mLock.unlock(); 7925 } 7926 7927 return NO_ERROR; 7928} 7929 7930// must be called with ThreadBase::mLock held 7931void AudioFlinger::EffectChain::setEffectSuspended_l( 7932 const effect_uuid_t *type, bool suspend) 7933{ 7934 sp<SuspendedEffectDesc> desc; 7935 // use effect type UUID timelow as key as there is no real risk of identical 7936 // timeLow fields among effect type UUIDs. 7937 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7938 if (suspend) { 7939 if (index >= 0) { 7940 desc = mSuspendedEffects.valueAt(index); 7941 } else { 7942 desc = new SuspendedEffectDesc(); 7943 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7944 mSuspendedEffects.add(type->timeLow, desc); 7945 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7946 } 7947 if (desc->mRefCount++ == 0) { 7948 sp<EffectModule> effect = getEffectIfEnabled(type); 7949 if (effect != 0) { 7950 desc->mEffect = effect; 7951 effect->setSuspended(true); 7952 effect->setEnabled(false); 7953 } 7954 } 7955 } else { 7956 if (index < 0) { 7957 return; 7958 } 7959 desc = mSuspendedEffects.valueAt(index); 7960 if (desc->mRefCount <= 0) { 7961 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7962 desc->mRefCount = 1; 7963 } 7964 if (--desc->mRefCount == 0) { 7965 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7966 if (desc->mEffect != 0) { 7967 sp<EffectModule> effect = desc->mEffect.promote(); 7968 if (effect != 0) { 7969 effect->setSuspended(false); 7970 sp<EffectHandle> handle = effect->controlHandle(); 7971 if (handle != 0) { 7972 effect->setEnabled(handle->enabled()); 7973 } 7974 } 7975 desc->mEffect.clear(); 7976 } 7977 mSuspendedEffects.removeItemsAt(index); 7978 } 7979 } 7980} 7981 7982// must be called with ThreadBase::mLock held 7983void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7984{ 7985 sp<SuspendedEffectDesc> desc; 7986 7987 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7988 if (suspend) { 7989 if (index >= 0) { 7990 desc = mSuspendedEffects.valueAt(index); 7991 } else { 7992 desc = new SuspendedEffectDesc(); 7993 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7994 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7995 } 7996 if (desc->mRefCount++ == 0) { 7997 Vector< sp<EffectModule> > effects; 7998 getSuspendEligibleEffects(effects); 7999 for (size_t i = 0; i < effects.size(); i++) { 8000 setEffectSuspended_l(&effects[i]->desc().type, true); 8001 } 8002 } 8003 } else { 8004 if (index < 0) { 8005 return; 8006 } 8007 desc = mSuspendedEffects.valueAt(index); 8008 if (desc->mRefCount <= 0) { 8009 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8010 desc->mRefCount = 1; 8011 } 8012 if (--desc->mRefCount == 0) { 8013 Vector<const effect_uuid_t *> types; 8014 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8015 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8016 continue; 8017 } 8018 types.add(&mSuspendedEffects.valueAt(i)->mType); 8019 } 8020 for (size_t i = 0; i < types.size(); i++) { 8021 setEffectSuspended_l(types[i], false); 8022 } 8023 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8024 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8025 } 8026 } 8027} 8028 8029 8030// The volume effect is used for automated tests only 8031#ifndef OPENSL_ES_H_ 8032static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8033 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8034const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8035#endif //OPENSL_ES_H_ 8036 8037bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8038{ 8039 // auxiliary effects and visualizer are never suspended on output mix 8040 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8041 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8042 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8043 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8044 return false; 8045 } 8046 return true; 8047} 8048 8049void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8050{ 8051 effects.clear(); 8052 for (size_t i = 0; i < mEffects.size(); i++) { 8053 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8054 effects.add(mEffects[i]); 8055 } 8056 } 8057} 8058 8059sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8060 const effect_uuid_t *type) 8061{ 8062 sp<EffectModule> effect = getEffectFromType_l(type); 8063 return effect != 0 && effect->isEnabled() ? effect : 0; 8064} 8065 8066void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8067 bool enabled) 8068{ 8069 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8070 if (enabled) { 8071 if (index < 0) { 8072 // if the effect is not suspend check if all effects are suspended 8073 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8074 if (index < 0) { 8075 return; 8076 } 8077 if (!isEffectEligibleForSuspend(effect->desc())) { 8078 return; 8079 } 8080 setEffectSuspended_l(&effect->desc().type, enabled); 8081 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8082 if (index < 0) { 8083 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8084 return; 8085 } 8086 } 8087 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8088 effect->desc().type.timeLow); 8089 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8090 // if effect is requested to suspended but was not yet enabled, supend it now. 8091 if (desc->mEffect == 0) { 8092 desc->mEffect = effect; 8093 effect->setEnabled(false); 8094 effect->setSuspended(true); 8095 } 8096 } else { 8097 if (index < 0) { 8098 return; 8099 } 8100 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8101 effect->desc().type.timeLow); 8102 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8103 desc->mEffect.clear(); 8104 effect->setSuspended(false); 8105 } 8106} 8107 8108#undef LOG_TAG 8109#define LOG_TAG "AudioFlinger" 8110 8111// ---------------------------------------------------------------------------- 8112 8113status_t AudioFlinger::onTransact( 8114 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8115{ 8116 return BnAudioFlinger::onTransact(code, data, reply, flags); 8117} 8118 8119}; // namespace android 8120