AudioFlinger.cpp revision 9eaa55756c5b245970447019250ce852f5189525
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_INIT; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_INIT; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 uint32_t flags, 436 const sp<IMemory>& sharedBuffer, 437 audio_io_handle_t output, 438 bool isTimed, 439 int *sessionId, 440 status_t *status) 441{ 442 sp<PlaybackThread::Track> track; 443 sp<TrackHandle> trackHandle; 444 sp<Client> client; 445 status_t lStatus; 446 int lSessionId; 447 448 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 449 // but if someone uses binder directly they could bypass that and cause us to crash 450 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 451 ALOGE("createTrack() invalid stream type %d", streamType); 452 lStatus = BAD_VALUE; 453 goto Exit; 454 } 455 456 { 457 Mutex::Autolock _l(mLock); 458 PlaybackThread *thread = checkPlaybackThread_l(output); 459 PlaybackThread *effectThread = NULL; 460 if (thread == NULL) { 461 ALOGE("unknown output thread"); 462 lStatus = BAD_VALUE; 463 goto Exit; 464 } 465 466 client = registerPid_l(pid); 467 468 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 469 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 470 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 471 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 472 if (mPlaybackThreads.keyAt(i) != output) { 473 // prevent same audio session on different output threads 474 uint32_t sessions = t->hasAudioSession(*sessionId); 475 if (sessions & PlaybackThread::TRACK_SESSION) { 476 ALOGE("createTrack() session ID %d already in use", *sessionId); 477 lStatus = BAD_VALUE; 478 goto Exit; 479 } 480 // check if an effect with same session ID is waiting for a track to be created 481 if (sessions & PlaybackThread::EFFECT_SESSION) { 482 effectThread = t.get(); 483 } 484 } 485 } 486 lSessionId = *sessionId; 487 } else { 488 // if no audio session id is provided, create one here 489 lSessionId = nextUniqueId(); 490 if (sessionId != NULL) { 491 *sessionId = lSessionId; 492 } 493 } 494 ALOGV("createTrack() lSessionId: %d", lSessionId); 495 496 track = thread->createTrack_l(client, streamType, sampleRate, format, 497 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 498 499 // move effect chain to this output thread if an effect on same session was waiting 500 // for a track to be created 501 if (lStatus == NO_ERROR && effectThread != NULL) { 502 Mutex::Autolock _dl(thread->mLock); 503 Mutex::Autolock _sl(effectThread->mLock); 504 moveEffectChain_l(lSessionId, effectThread, thread, true); 505 } 506 } 507 if (lStatus == NO_ERROR) { 508 trackHandle = new TrackHandle(track); 509 } else { 510 // remove local strong reference to Client before deleting the Track so that the Client 511 // destructor is called by the TrackBase destructor with mLock held 512 client.clear(); 513 track.clear(); 514 } 515 516Exit: 517 if(status) { 518 *status = lStatus; 519 } 520 return trackHandle; 521} 522 523uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 524{ 525 Mutex::Autolock _l(mLock); 526 PlaybackThread *thread = checkPlaybackThread_l(output); 527 if (thread == NULL) { 528 ALOGW("sampleRate() unknown thread %d", output); 529 return 0; 530 } 531 return thread->sampleRate(); 532} 533 534int AudioFlinger::channelCount(audio_io_handle_t output) const 535{ 536 Mutex::Autolock _l(mLock); 537 PlaybackThread *thread = checkPlaybackThread_l(output); 538 if (thread == NULL) { 539 ALOGW("channelCount() unknown thread %d", output); 540 return 0; 541 } 542 return thread->channelCount(); 543} 544 545audio_format_t AudioFlinger::format(audio_io_handle_t output) const 546{ 547 Mutex::Autolock _l(mLock); 548 PlaybackThread *thread = checkPlaybackThread_l(output); 549 if (thread == NULL) { 550 ALOGW("format() unknown thread %d", output); 551 return AUDIO_FORMAT_INVALID; 552 } 553 return thread->format(); 554} 555 556size_t AudioFlinger::frameCount(audio_io_handle_t output) const 557{ 558 Mutex::Autolock _l(mLock); 559 PlaybackThread *thread = checkPlaybackThread_l(output); 560 if (thread == NULL) { 561 ALOGW("frameCount() unknown thread %d", output); 562 return 0; 563 } 564 return thread->frameCount(); 565} 566 567uint32_t AudioFlinger::latency(audio_io_handle_t output) const 568{ 569 Mutex::Autolock _l(mLock); 570 PlaybackThread *thread = checkPlaybackThread_l(output); 571 if (thread == NULL) { 572 ALOGW("latency() unknown thread %d", output); 573 return 0; 574 } 575 return thread->latency(); 576} 577 578status_t AudioFlinger::setMasterVolume(float value) 579{ 580 status_t ret = initCheck(); 581 if (ret != NO_ERROR) { 582 return ret; 583 } 584 585 // check calling permissions 586 if (!settingsAllowed()) { 587 return PERMISSION_DENIED; 588 } 589 590 float swmv = value; 591 592 // when hw supports master volume, don't scale in sw mixer 593 if (MVS_NONE != mMasterVolumeSupportLvl) { 594 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 595 AutoMutex lock(mHardwareLock); 596 audio_hw_device_t *dev = mAudioHwDevs[i]; 597 598 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 599 if (NULL != dev->set_master_volume) { 600 dev->set_master_volume(dev, value); 601 } 602 mHardwareStatus = AUDIO_HW_IDLE; 603 } 604 605 swmv = 1.0; 606 } 607 608 Mutex::Autolock _l(mLock); 609 mMasterVolume = value; 610 mMasterVolumeSW = swmv; 611 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 612 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 613 614 return NO_ERROR; 615} 616 617status_t AudioFlinger::setMode(audio_mode_t mode) 618{ 619 status_t ret = initCheck(); 620 if (ret != NO_ERROR) { 621 return ret; 622 } 623 624 // check calling permissions 625 if (!settingsAllowed()) { 626 return PERMISSION_DENIED; 627 } 628 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 629 ALOGW("Illegal value: setMode(%d)", mode); 630 return BAD_VALUE; 631 } 632 633 { // scope for the lock 634 AutoMutex lock(mHardwareLock); 635 mHardwareStatus = AUDIO_HW_SET_MODE; 636 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 637 mHardwareStatus = AUDIO_HW_IDLE; 638 } 639 640 if (NO_ERROR == ret) { 641 Mutex::Autolock _l(mLock); 642 mMode = mode; 643 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 644 mPlaybackThreads.valueAt(i)->setMode(mode); 645 } 646 647 return ret; 648} 649 650status_t AudioFlinger::setMicMute(bool state) 651{ 652 status_t ret = initCheck(); 653 if (ret != NO_ERROR) { 654 return ret; 655 } 656 657 // check calling permissions 658 if (!settingsAllowed()) { 659 return PERMISSION_DENIED; 660 } 661 662 AutoMutex lock(mHardwareLock); 663 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 664 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 665 mHardwareStatus = AUDIO_HW_IDLE; 666 return ret; 667} 668 669bool AudioFlinger::getMicMute() const 670{ 671 status_t ret = initCheck(); 672 if (ret != NO_ERROR) { 673 return false; 674 } 675 676 bool state = AUDIO_MODE_INVALID; 677 AutoMutex lock(mHardwareLock); 678 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 679 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 680 mHardwareStatus = AUDIO_HW_IDLE; 681 return state; 682} 683 684status_t AudioFlinger::setMasterMute(bool muted) 685{ 686 // check calling permissions 687 if (!settingsAllowed()) { 688 return PERMISSION_DENIED; 689 } 690 691 Mutex::Autolock _l(mLock); 692 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 693 mMasterMute = muted; 694 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 695 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 696 697 return NO_ERROR; 698} 699 700float AudioFlinger::masterVolume() const 701{ 702 Mutex::Autolock _l(mLock); 703 return masterVolume_l(); 704} 705 706float AudioFlinger::masterVolumeSW() const 707{ 708 Mutex::Autolock _l(mLock); 709 return masterVolumeSW_l(); 710} 711 712bool AudioFlinger::masterMute() const 713{ 714 Mutex::Autolock _l(mLock); 715 return masterMute_l(); 716} 717 718float AudioFlinger::masterVolume_l() const 719{ 720 if (MVS_FULL == mMasterVolumeSupportLvl) { 721 float ret_val; 722 AutoMutex lock(mHardwareLock); 723 724 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 725 assert(NULL != mPrimaryHardwareDev); 726 assert(NULL != mPrimaryHardwareDev->get_master_volume); 727 728 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 729 mHardwareStatus = AUDIO_HW_IDLE; 730 return ret_val; 731 } 732 733 return mMasterVolume; 734} 735 736status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 737 audio_io_handle_t output) 738{ 739 // check calling permissions 740 if (!settingsAllowed()) { 741 return PERMISSION_DENIED; 742 } 743 744 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 745 ALOGE("setStreamVolume() invalid stream %d", stream); 746 return BAD_VALUE; 747 } 748 749 AutoMutex lock(mLock); 750 PlaybackThread *thread = NULL; 751 if (output) { 752 thread = checkPlaybackThread_l(output); 753 if (thread == NULL) { 754 return BAD_VALUE; 755 } 756 } 757 758 mStreamTypes[stream].volume = value; 759 760 if (thread == NULL) { 761 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 762 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 763 } 764 } else { 765 thread->setStreamVolume(stream, value); 766 } 767 768 return NO_ERROR; 769} 770 771status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 772{ 773 // check calling permissions 774 if (!settingsAllowed()) { 775 return PERMISSION_DENIED; 776 } 777 778 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 779 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 780 ALOGE("setStreamMute() invalid stream %d", stream); 781 return BAD_VALUE; 782 } 783 784 AutoMutex lock(mLock); 785 mStreamTypes[stream].mute = muted; 786 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 787 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 788 789 return NO_ERROR; 790} 791 792float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 793{ 794 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 795 return 0.0f; 796 } 797 798 AutoMutex lock(mLock); 799 float volume; 800 if (output) { 801 PlaybackThread *thread = checkPlaybackThread_l(output); 802 if (thread == NULL) { 803 return 0.0f; 804 } 805 volume = thread->streamVolume(stream); 806 } else { 807 volume = streamVolume_l(stream); 808 } 809 810 return volume; 811} 812 813bool AudioFlinger::streamMute(audio_stream_type_t stream) const 814{ 815 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 816 return true; 817 } 818 819 AutoMutex lock(mLock); 820 return streamMute_l(stream); 821} 822 823status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 824{ 825 status_t result; 826 827 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 828 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 829 // check calling permissions 830 if (!settingsAllowed()) { 831 return PERMISSION_DENIED; 832 } 833 834 // ioHandle == 0 means the parameters are global to the audio hardware interface 835 if (ioHandle == 0) { 836 AutoMutex lock(mHardwareLock); 837 mHardwareStatus = AUDIO_SET_PARAMETER; 838 status_t final_result = NO_ERROR; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs[i]; 841 result = dev->set_parameters(dev, keyValuePairs.string()); 842 final_result = result ?: final_result; 843 } 844 mHardwareStatus = AUDIO_HW_IDLE; 845 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 846 AudioParameter param = AudioParameter(keyValuePairs); 847 String8 value; 848 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 849 Mutex::Autolock _l(mLock); 850 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 851 if (mBtNrecIsOff != btNrecIsOff) { 852 for (size_t i = 0; i < mRecordThreads.size(); i++) { 853 sp<RecordThread> thread = mRecordThreads.valueAt(i); 854 RecordThread::RecordTrack *track = thread->track(); 855 if (track != NULL) { 856 audio_devices_t device = (audio_devices_t)( 857 thread->device() & AUDIO_DEVICE_IN_ALL); 858 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 859 thread->setEffectSuspended(FX_IID_AEC, 860 suspend, 861 track->sessionId()); 862 thread->setEffectSuspended(FX_IID_NS, 863 suspend, 864 track->sessionId()); 865 } 866 } 867 mBtNrecIsOff = btNrecIsOff; 868 } 869 } 870 return final_result; 871 } 872 873 // hold a strong ref on thread in case closeOutput() or closeInput() is called 874 // and the thread is exited once the lock is released 875 sp<ThreadBase> thread; 876 { 877 Mutex::Autolock _l(mLock); 878 thread = checkPlaybackThread_l(ioHandle); 879 if (thread == NULL) { 880 thread = checkRecordThread_l(ioHandle); 881 } else if (thread == primaryPlaybackThread_l()) { 882 // indicate output device change to all input threads for pre processing 883 AudioParameter param = AudioParameter(keyValuePairs); 884 int value; 885 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 886 for (size_t i = 0; i < mRecordThreads.size(); i++) { 887 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 888 } 889 } 890 } 891 } 892 if (thread != 0) { 893 return thread->setParameters(keyValuePairs); 894 } 895 return BAD_VALUE; 896} 897 898String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 899{ 900// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 901// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 902 903 if (ioHandle == 0) { 904 String8 out_s8; 905 906 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 907 audio_hw_device_t *dev = mAudioHwDevs[i]; 908 char *s = dev->get_parameters(dev, keys.string()); 909 out_s8 += String8(s ? s : ""); 910 free(s); 911 } 912 return out_s8; 913 } 914 915 Mutex::Autolock _l(mLock); 916 917 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 918 if (playbackThread != NULL) { 919 return playbackThread->getParameters(keys); 920 } 921 RecordThread *recordThread = checkRecordThread_l(ioHandle); 922 if (recordThread != NULL) { 923 return recordThread->getParameters(keys); 924 } 925 return String8(""); 926} 927 928size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 929{ 930 status_t ret = initCheck(); 931 if (ret != NO_ERROR) { 932 return 0; 933 } 934 935 AutoMutex lock(mHardwareLock); 936 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 937 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 938 mHardwareStatus = AUDIO_HW_IDLE; 939 return size; 940} 941 942unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 943{ 944 if (ioHandle == 0) { 945 return 0; 946 } 947 948 Mutex::Autolock _l(mLock); 949 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getInputFramesLost(); 953 } 954 return 0; 955} 956 957status_t AudioFlinger::setVoiceVolume(float value) 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return ret; 962 } 963 964 // check calling permissions 965 if (!settingsAllowed()) { 966 return PERMISSION_DENIED; 967 } 968 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 971 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 972 mHardwareStatus = AUDIO_HW_IDLE; 973 974 return ret; 975} 976 977status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 978 audio_io_handle_t output) const 979{ 980 status_t status; 981 982 Mutex::Autolock _l(mLock); 983 984 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 985 if (playbackThread != NULL) { 986 return playbackThread->getRenderPosition(halFrames, dspFrames); 987 } 988 989 return BAD_VALUE; 990} 991 992void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 993{ 994 995 Mutex::Autolock _l(mLock); 996 997 pid_t pid = IPCThreadState::self()->getCallingPid(); 998 if (mNotificationClients.indexOfKey(pid) < 0) { 999 sp<NotificationClient> notificationClient = new NotificationClient(this, 1000 client, 1001 pid); 1002 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1003 1004 mNotificationClients.add(pid, notificationClient); 1005 1006 sp<IBinder> binder = client->asBinder(); 1007 binder->linkToDeath(notificationClient); 1008 1009 // the config change is always sent from playback or record threads to avoid deadlock 1010 // with AudioSystem::gLock 1011 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1012 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1013 } 1014 1015 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1016 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1017 } 1018 } 1019} 1020 1021void AudioFlinger::removeNotificationClient(pid_t pid) 1022{ 1023 Mutex::Autolock _l(mLock); 1024 1025 ssize_t index = mNotificationClients.indexOfKey(pid); 1026 if (index >= 0) { 1027 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 1028 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 1029 mNotificationClients.removeItem(pid); 1030 } 1031 1032 ALOGV("%d died, releasing its sessions", pid); 1033 size_t num = mAudioSessionRefs.size(); 1034 bool removed = false; 1035 for (size_t i = 0; i< num; ) { 1036 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1037 ALOGV(" pid %d @ %d", ref->pid, i); 1038 if (ref->pid == pid) { 1039 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1040 mAudioSessionRefs.removeAt(i); 1041 delete ref; 1042 removed = true; 1043 num--; 1044 } else { 1045 i++; 1046 } 1047 } 1048 if (removed) { 1049 purgeStaleEffects_l(); 1050 } 1051} 1052 1053// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1054void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1055{ 1056 size_t size = mNotificationClients.size(); 1057 for (size_t i = 0; i < size; i++) { 1058 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1059 param2); 1060 } 1061} 1062 1063// removeClient_l() must be called with AudioFlinger::mLock held 1064void AudioFlinger::removeClient_l(pid_t pid) 1065{ 1066 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1067 mClients.removeItem(pid); 1068} 1069 1070 1071// ---------------------------------------------------------------------------- 1072 1073AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1074 uint32_t device, type_t type) 1075 : Thread(false), 1076 mType(type), 1077 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1078 // mChannelMask 1079 mChannelCount(0), 1080 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1081 mParamStatus(NO_ERROR), 1082 mStandby(false), mId(id), 1083 mDevice(device), 1084 mDeathRecipient(new PMDeathRecipient(this)) 1085{ 1086} 1087 1088AudioFlinger::ThreadBase::~ThreadBase() 1089{ 1090 mParamCond.broadcast(); 1091 // do not lock the mutex in destructor 1092 releaseWakeLock_l(); 1093 if (mPowerManager != 0) { 1094 sp<IBinder> binder = mPowerManager->asBinder(); 1095 binder->unlinkToDeath(mDeathRecipient); 1096 } 1097} 1098 1099void AudioFlinger::ThreadBase::exit() 1100{ 1101 ALOGV("ThreadBase::exit"); 1102 { 1103 // This lock prevents the following race in thread (uniprocessor for illustration): 1104 // if (!exitPending()) { 1105 // // context switch from here to exit() 1106 // // exit() calls requestExit(), what exitPending() observes 1107 // // exit() calls signal(), which is dropped since no waiters 1108 // // context switch back from exit() to here 1109 // mWaitWorkCV.wait(...); 1110 // // now thread is hung 1111 // } 1112 AutoMutex lock(mLock); 1113 requestExit(); 1114 mWaitWorkCV.signal(); 1115 } 1116 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1117 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1118 requestExitAndWait(); 1119} 1120 1121status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1122{ 1123 status_t status; 1124 1125 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1126 Mutex::Autolock _l(mLock); 1127 1128 mNewParameters.add(keyValuePairs); 1129 mWaitWorkCV.signal(); 1130 // wait condition with timeout in case the thread loop has exited 1131 // before the request could be processed 1132 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1133 status = mParamStatus; 1134 mWaitWorkCV.signal(); 1135 } else { 1136 status = TIMED_OUT; 1137 } 1138 return status; 1139} 1140 1141void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1142{ 1143 Mutex::Autolock _l(mLock); 1144 sendConfigEvent_l(event, param); 1145} 1146 1147// sendConfigEvent_l() must be called with ThreadBase::mLock held 1148void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1149{ 1150 ConfigEvent configEvent; 1151 configEvent.mEvent = event; 1152 configEvent.mParam = param; 1153 mConfigEvents.add(configEvent); 1154 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1155 mWaitWorkCV.signal(); 1156} 1157 1158void AudioFlinger::ThreadBase::processConfigEvents() 1159{ 1160 mLock.lock(); 1161 while(!mConfigEvents.isEmpty()) { 1162 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1163 ConfigEvent configEvent = mConfigEvents[0]; 1164 mConfigEvents.removeAt(0); 1165 // release mLock before locking AudioFlinger mLock: lock order is always 1166 // AudioFlinger then ThreadBase to avoid cross deadlock 1167 mLock.unlock(); 1168 mAudioFlinger->mLock.lock(); 1169 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1170 mAudioFlinger->mLock.unlock(); 1171 mLock.lock(); 1172 } 1173 mLock.unlock(); 1174} 1175 1176status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1177{ 1178 const size_t SIZE = 256; 1179 char buffer[SIZE]; 1180 String8 result; 1181 1182 bool locked = tryLock(mLock); 1183 if (!locked) { 1184 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1185 write(fd, buffer, strlen(buffer)); 1186 } 1187 1188 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1189 result.append(buffer); 1190 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1191 result.append(buffer); 1192 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1193 result.append(buffer); 1194 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1197 result.append(buffer); 1198 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1199 result.append(buffer); 1200 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1201 result.append(buffer); 1202 1203 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1204 result.append(buffer); 1205 result.append(" Index Command"); 1206 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1207 snprintf(buffer, SIZE, "\n %02d ", i); 1208 result.append(buffer); 1209 result.append(mNewParameters[i]); 1210 } 1211 1212 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, " Index event param\n"); 1215 result.append(buffer); 1216 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1217 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1218 result.append(buffer); 1219 } 1220 result.append("\n"); 1221 1222 write(fd, result.string(), result.size()); 1223 1224 if (locked) { 1225 mLock.unlock(); 1226 } 1227 return NO_ERROR; 1228} 1229 1230status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1231{ 1232 const size_t SIZE = 256; 1233 char buffer[SIZE]; 1234 String8 result; 1235 1236 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1237 write(fd, buffer, strlen(buffer)); 1238 1239 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1240 sp<EffectChain> chain = mEffectChains[i]; 1241 if (chain != 0) { 1242 chain->dump(fd, args); 1243 } 1244 } 1245 return NO_ERROR; 1246} 1247 1248void AudioFlinger::ThreadBase::acquireWakeLock() 1249{ 1250 Mutex::Autolock _l(mLock); 1251 acquireWakeLock_l(); 1252} 1253 1254void AudioFlinger::ThreadBase::acquireWakeLock_l() 1255{ 1256 if (mPowerManager == 0) { 1257 // use checkService() to avoid blocking if power service is not up yet 1258 sp<IBinder> binder = 1259 defaultServiceManager()->checkService(String16("power")); 1260 if (binder == 0) { 1261 ALOGW("Thread %s cannot connect to the power manager service", mName); 1262 } else { 1263 mPowerManager = interface_cast<IPowerManager>(binder); 1264 binder->linkToDeath(mDeathRecipient); 1265 } 1266 } 1267 if (mPowerManager != 0) { 1268 sp<IBinder> binder = new BBinder(); 1269 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1270 binder, 1271 String16(mName)); 1272 if (status == NO_ERROR) { 1273 mWakeLockToken = binder; 1274 } 1275 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1276 } 1277} 1278 1279void AudioFlinger::ThreadBase::releaseWakeLock() 1280{ 1281 Mutex::Autolock _l(mLock); 1282 releaseWakeLock_l(); 1283} 1284 1285void AudioFlinger::ThreadBase::releaseWakeLock_l() 1286{ 1287 if (mWakeLockToken != 0) { 1288 ALOGV("releaseWakeLock_l() %s", mName); 1289 if (mPowerManager != 0) { 1290 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1291 } 1292 mWakeLockToken.clear(); 1293 } 1294} 1295 1296void AudioFlinger::ThreadBase::clearPowerManager() 1297{ 1298 Mutex::Autolock _l(mLock); 1299 releaseWakeLock_l(); 1300 mPowerManager.clear(); 1301} 1302 1303void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1304{ 1305 sp<ThreadBase> thread = mThread.promote(); 1306 if (thread != 0) { 1307 thread->clearPowerManager(); 1308 } 1309 ALOGW("power manager service died !!!"); 1310} 1311 1312void AudioFlinger::ThreadBase::setEffectSuspended( 1313 const effect_uuid_t *type, bool suspend, int sessionId) 1314{ 1315 Mutex::Autolock _l(mLock); 1316 setEffectSuspended_l(type, suspend, sessionId); 1317} 1318 1319void AudioFlinger::ThreadBase::setEffectSuspended_l( 1320 const effect_uuid_t *type, bool suspend, int sessionId) 1321{ 1322 sp<EffectChain> chain = getEffectChain_l(sessionId); 1323 if (chain != 0) { 1324 if (type != NULL) { 1325 chain->setEffectSuspended_l(type, suspend); 1326 } else { 1327 chain->setEffectSuspendedAll_l(suspend); 1328 } 1329 } 1330 1331 updateSuspendedSessions_l(type, suspend, sessionId); 1332} 1333 1334void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1335{ 1336 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1337 if (index < 0) { 1338 return; 1339 } 1340 1341 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1342 mSuspendedSessions.editValueAt(index); 1343 1344 for (size_t i = 0; i < sessionEffects.size(); i++) { 1345 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1346 for (int j = 0; j < desc->mRefCount; j++) { 1347 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1348 chain->setEffectSuspendedAll_l(true); 1349 } else { 1350 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1351 desc->mType.timeLow); 1352 chain->setEffectSuspended_l(&desc->mType, true); 1353 } 1354 } 1355 } 1356} 1357 1358void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1359 bool suspend, 1360 int sessionId) 1361{ 1362 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1363 1364 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1365 1366 if (suspend) { 1367 if (index >= 0) { 1368 sessionEffects = mSuspendedSessions.editValueAt(index); 1369 } else { 1370 mSuspendedSessions.add(sessionId, sessionEffects); 1371 } 1372 } else { 1373 if (index < 0) { 1374 return; 1375 } 1376 sessionEffects = mSuspendedSessions.editValueAt(index); 1377 } 1378 1379 1380 int key = EffectChain::kKeyForSuspendAll; 1381 if (type != NULL) { 1382 key = type->timeLow; 1383 } 1384 index = sessionEffects.indexOfKey(key); 1385 1386 sp <SuspendedSessionDesc> desc; 1387 if (suspend) { 1388 if (index >= 0) { 1389 desc = sessionEffects.valueAt(index); 1390 } else { 1391 desc = new SuspendedSessionDesc(); 1392 if (type != NULL) { 1393 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1394 } 1395 sessionEffects.add(key, desc); 1396 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1397 } 1398 desc->mRefCount++; 1399 } else { 1400 if (index < 0) { 1401 return; 1402 } 1403 desc = sessionEffects.valueAt(index); 1404 if (--desc->mRefCount == 0) { 1405 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1406 sessionEffects.removeItemsAt(index); 1407 if (sessionEffects.isEmpty()) { 1408 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1409 sessionId); 1410 mSuspendedSessions.removeItem(sessionId); 1411 } 1412 } 1413 } 1414 if (!sessionEffects.isEmpty()) { 1415 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1416 } 1417} 1418 1419void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1420 bool enabled, 1421 int sessionId) 1422{ 1423 Mutex::Autolock _l(mLock); 1424 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1425} 1426 1427void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1428 bool enabled, 1429 int sessionId) 1430{ 1431 if (mType != RECORD) { 1432 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1433 // another session. This gives the priority to well behaved effect control panels 1434 // and applications not using global effects. 1435 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1436 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1437 } 1438 } 1439 1440 sp<EffectChain> chain = getEffectChain_l(sessionId); 1441 if (chain != 0) { 1442 chain->checkSuspendOnEffectEnabled(effect, enabled); 1443 } 1444} 1445 1446// ---------------------------------------------------------------------------- 1447 1448AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1449 AudioStreamOut* output, 1450 audio_io_handle_t id, 1451 uint32_t device, 1452 type_t type) 1453 : ThreadBase(audioFlinger, id, device, type), 1454 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1455 // Assumes constructor is called by AudioFlinger with it's mLock held, 1456 // but it would be safer to explicitly pass initial masterMute as parameter 1457 mMasterMute(audioFlinger->masterMute_l()), 1458 // mStreamTypes[] initialized in constructor body 1459 mOutput(output), 1460 // Assumes constructor is called by AudioFlinger with it's mLock held, 1461 // but it would be safer to explicitly pass initial masterVolume as parameter 1462 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1463 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1464{ 1465 snprintf(mName, kNameLength, "AudioOut_%d", id); 1466 1467 readOutputParameters(); 1468 1469 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1470 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1471 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1472 stream = (audio_stream_type_t) (stream + 1)) { 1473 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1474 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1475 // initialized by stream_type_t default constructor 1476 // mStreamTypes[stream].valid = true; 1477 } 1478 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1479 // because mAudioFlinger doesn't have one to copy from 1480} 1481 1482AudioFlinger::PlaybackThread::~PlaybackThread() 1483{ 1484 delete [] mMixBuffer; 1485} 1486 1487status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1488{ 1489 dumpInternals(fd, args); 1490 dumpTracks(fd, args); 1491 dumpEffectChains(fd, args); 1492 return NO_ERROR; 1493} 1494 1495status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1496{ 1497 const size_t SIZE = 256; 1498 char buffer[SIZE]; 1499 String8 result; 1500 1501 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1502 result.append(buffer); 1503 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1504 for (size_t i = 0; i < mTracks.size(); ++i) { 1505 sp<Track> track = mTracks[i]; 1506 if (track != 0) { 1507 track->dump(buffer, SIZE); 1508 result.append(buffer); 1509 } 1510 } 1511 1512 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1513 result.append(buffer); 1514 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1515 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1516 sp<Track> track = mActiveTracks[i].promote(); 1517 if (track != 0) { 1518 track->dump(buffer, SIZE); 1519 result.append(buffer); 1520 } 1521 } 1522 write(fd, result.string(), result.size()); 1523 return NO_ERROR; 1524} 1525 1526status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1527{ 1528 const size_t SIZE = 256; 1529 char buffer[SIZE]; 1530 String8 result; 1531 1532 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1533 result.append(buffer); 1534 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1535 result.append(buffer); 1536 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1537 result.append(buffer); 1538 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1541 result.append(buffer); 1542 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1545 result.append(buffer); 1546 write(fd, result.string(), result.size()); 1547 1548 dumpBase(fd, args); 1549 1550 return NO_ERROR; 1551} 1552 1553// Thread virtuals 1554status_t AudioFlinger::PlaybackThread::readyToRun() 1555{ 1556 status_t status = initCheck(); 1557 if (status == NO_ERROR) { 1558 ALOGI("AudioFlinger's thread %p ready to run", this); 1559 } else { 1560 ALOGE("No working audio driver found."); 1561 } 1562 return status; 1563} 1564 1565void AudioFlinger::PlaybackThread::onFirstRef() 1566{ 1567 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1568} 1569 1570// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1571sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1572 const sp<AudioFlinger::Client>& client, 1573 audio_stream_type_t streamType, 1574 uint32_t sampleRate, 1575 audio_format_t format, 1576 uint32_t channelMask, 1577 int frameCount, 1578 const sp<IMemory>& sharedBuffer, 1579 int sessionId, 1580 bool isTimed, 1581 status_t *status) 1582{ 1583 sp<Track> track; 1584 status_t lStatus; 1585 1586 if (mType == DIRECT) { 1587 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1588 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1589 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1590 "for output %p with format %d", 1591 sampleRate, format, channelMask, mOutput, mFormat); 1592 lStatus = BAD_VALUE; 1593 goto Exit; 1594 } 1595 } 1596 } else { 1597 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1598 if (sampleRate > mSampleRate*2) { 1599 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1600 lStatus = BAD_VALUE; 1601 goto Exit; 1602 } 1603 } 1604 1605 lStatus = initCheck(); 1606 if (lStatus != NO_ERROR) { 1607 ALOGE("Audio driver not initialized."); 1608 goto Exit; 1609 } 1610 1611 { // scope for mLock 1612 Mutex::Autolock _l(mLock); 1613 1614 // all tracks in same audio session must share the same routing strategy otherwise 1615 // conflicts will happen when tracks are moved from one output to another by audio policy 1616 // manager 1617 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1618 for (size_t i = 0; i < mTracks.size(); ++i) { 1619 sp<Track> t = mTracks[i]; 1620 if (t != 0) { 1621 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1622 if (sessionId == t->sessionId() && strategy != actual) { 1623 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1624 strategy, actual); 1625 lStatus = BAD_VALUE; 1626 goto Exit; 1627 } 1628 } 1629 } 1630 1631 if (!isTimed) { 1632 track = new Track(this, client, streamType, sampleRate, format, 1633 channelMask, frameCount, sharedBuffer, sessionId); 1634 } else { 1635 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1636 channelMask, frameCount, sharedBuffer, sessionId); 1637 } 1638 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1639 lStatus = NO_MEMORY; 1640 goto Exit; 1641 } 1642 mTracks.add(track); 1643 1644 sp<EffectChain> chain = getEffectChain_l(sessionId); 1645 if (chain != 0) { 1646 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1647 track->setMainBuffer(chain->inBuffer()); 1648 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1649 chain->incTrackCnt(); 1650 } 1651 1652 // invalidate track immediately if the stream type was moved to another thread since 1653 // createTrack() was called by the client process. 1654 if (!mStreamTypes[streamType].valid) { 1655 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1656 this, streamType); 1657 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1658 } 1659 } 1660 lStatus = NO_ERROR; 1661 1662Exit: 1663 if(status) { 1664 *status = lStatus; 1665 } 1666 return track; 1667} 1668 1669uint32_t AudioFlinger::PlaybackThread::latency() const 1670{ 1671 Mutex::Autolock _l(mLock); 1672 if (initCheck() == NO_ERROR) { 1673 return mOutput->stream->get_latency(mOutput->stream); 1674 } else { 1675 return 0; 1676 } 1677} 1678 1679void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1680{ 1681 Mutex::Autolock _l(mLock); 1682 mMasterVolume = value; 1683} 1684 1685void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1686{ 1687 Mutex::Autolock _l(mLock); 1688 setMasterMute_l(muted); 1689} 1690 1691void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1692{ 1693 Mutex::Autolock _l(mLock); 1694 mStreamTypes[stream].volume = value; 1695} 1696 1697void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1698{ 1699 Mutex::Autolock _l(mLock); 1700 mStreamTypes[stream].mute = muted; 1701} 1702 1703float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1704{ 1705 Mutex::Autolock _l(mLock); 1706 return mStreamTypes[stream].volume; 1707} 1708 1709// addTrack_l() must be called with ThreadBase::mLock held 1710status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1711{ 1712 status_t status = ALREADY_EXISTS; 1713 1714 // set retry count for buffer fill 1715 track->mRetryCount = kMaxTrackStartupRetries; 1716 if (mActiveTracks.indexOf(track) < 0) { 1717 // the track is newly added, make sure it fills up all its 1718 // buffers before playing. This is to ensure the client will 1719 // effectively get the latency it requested. 1720 track->mFillingUpStatus = Track::FS_FILLING; 1721 track->mResetDone = false; 1722 mActiveTracks.add(track); 1723 if (track->mainBuffer() != mMixBuffer) { 1724 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1725 if (chain != 0) { 1726 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1727 chain->incActiveTrackCnt(); 1728 } 1729 } 1730 1731 status = NO_ERROR; 1732 } 1733 1734 ALOGV("mWaitWorkCV.broadcast"); 1735 mWaitWorkCV.broadcast(); 1736 1737 return status; 1738} 1739 1740// destroyTrack_l() must be called with ThreadBase::mLock held 1741void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1742{ 1743 track->mState = TrackBase::TERMINATED; 1744 if (mActiveTracks.indexOf(track) < 0) { 1745 removeTrack_l(track); 1746 } 1747} 1748 1749void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1750{ 1751 mTracks.remove(track); 1752 deleteTrackName_l(track->name()); 1753 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1754 if (chain != 0) { 1755 chain->decTrackCnt(); 1756 } 1757} 1758 1759String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1760{ 1761 String8 out_s8 = String8(""); 1762 char *s; 1763 1764 Mutex::Autolock _l(mLock); 1765 if (initCheck() != NO_ERROR) { 1766 return out_s8; 1767 } 1768 1769 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1770 out_s8 = String8(s); 1771 free(s); 1772 return out_s8; 1773} 1774 1775// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1776void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1777 AudioSystem::OutputDescriptor desc; 1778 void *param2 = NULL; 1779 1780 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1781 1782 switch (event) { 1783 case AudioSystem::OUTPUT_OPENED: 1784 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1785 desc.channels = mChannelMask; 1786 desc.samplingRate = mSampleRate; 1787 desc.format = mFormat; 1788 desc.frameCount = mFrameCount; 1789 desc.latency = latency(); 1790 param2 = &desc; 1791 break; 1792 1793 case AudioSystem::STREAM_CONFIG_CHANGED: 1794 param2 = ¶m; 1795 case AudioSystem::OUTPUT_CLOSED: 1796 default: 1797 break; 1798 } 1799 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1800} 1801 1802void AudioFlinger::PlaybackThread::readOutputParameters() 1803{ 1804 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1805 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1806 mChannelCount = (uint16_t)popcount(mChannelMask); 1807 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1808 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1809 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1810 1811 // FIXME - Current mixer implementation only supports stereo output: Always 1812 // Allocate a stereo buffer even if HW output is mono. 1813 delete[] mMixBuffer; 1814 mMixBuffer = new int16_t[mFrameCount * 2]; 1815 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1816 1817 // force reconfiguration of effect chains and engines to take new buffer size and audio 1818 // parameters into account 1819 // Note that mLock is not held when readOutputParameters() is called from the constructor 1820 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1821 // matter. 1822 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1823 Vector< sp<EffectChain> > effectChains = mEffectChains; 1824 for (size_t i = 0; i < effectChains.size(); i ++) { 1825 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1826 } 1827} 1828 1829status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1830{ 1831 if (halFrames == NULL || dspFrames == NULL) { 1832 return BAD_VALUE; 1833 } 1834 Mutex::Autolock _l(mLock); 1835 if (initCheck() != NO_ERROR) { 1836 return INVALID_OPERATION; 1837 } 1838 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1839 1840 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1841} 1842 1843uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1844{ 1845 Mutex::Autolock _l(mLock); 1846 uint32_t result = 0; 1847 if (getEffectChain_l(sessionId) != 0) { 1848 result = EFFECT_SESSION; 1849 } 1850 1851 for (size_t i = 0; i < mTracks.size(); ++i) { 1852 sp<Track> track = mTracks[i]; 1853 if (sessionId == track->sessionId() && 1854 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1855 result |= TRACK_SESSION; 1856 break; 1857 } 1858 } 1859 1860 return result; 1861} 1862 1863uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1864{ 1865 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1866 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1867 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1868 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1869 } 1870 for (size_t i = 0; i < mTracks.size(); i++) { 1871 sp<Track> track = mTracks[i]; 1872 if (sessionId == track->sessionId() && 1873 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1874 return AudioSystem::getStrategyForStream(track->streamType()); 1875 } 1876 } 1877 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1878} 1879 1880 1881AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1882{ 1883 Mutex::Autolock _l(mLock); 1884 return mOutput; 1885} 1886 1887AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1888{ 1889 Mutex::Autolock _l(mLock); 1890 AudioStreamOut *output = mOutput; 1891 mOutput = NULL; 1892 return output; 1893} 1894 1895// this method must always be called either with ThreadBase mLock held or inside the thread loop 1896audio_stream_t* AudioFlinger::PlaybackThread::stream() 1897{ 1898 if (mOutput == NULL) { 1899 return NULL; 1900 } 1901 return &mOutput->stream->common; 1902} 1903 1904uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1905{ 1906 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1907 // decoding and transfer time. So sleeping for half of the latency would likely cause 1908 // underruns 1909 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1910 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1911 } else { 1912 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1913 } 1914} 1915 1916// ---------------------------------------------------------------------------- 1917 1918AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1919 audio_io_handle_t id, uint32_t device, type_t type) 1920 : PlaybackThread(audioFlinger, output, id, device, type), 1921 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1922 mPrevMixerStatus(MIXER_IDLE) 1923{ 1924 // FIXME - Current mixer implementation only supports stereo output 1925 if (mChannelCount == 1) { 1926 ALOGE("Invalid audio hardware channel count"); 1927 } 1928} 1929 1930AudioFlinger::MixerThread::~MixerThread() 1931{ 1932 delete mAudioMixer; 1933} 1934 1935bool AudioFlinger::MixerThread::threadLoop() 1936{ 1937 Vector< sp<Track> > tracksToRemove; 1938 mixer_state mixerStatus = MIXER_IDLE; 1939 nsecs_t standbyTime = systemTime(); 1940 size_t mixBufferSize = mFrameCount * mFrameSize; 1941 // FIXME: Relaxed timing because of a certain device that can't meet latency 1942 // Should be reduced to 2x after the vendor fixes the driver issue 1943 // increase threshold again due to low power audio mode. The way this warning threshold is 1944 // calculated and its usefulness should be reconsidered anyway. 1945 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1946 nsecs_t lastWarning = 0; 1947 bool longStandbyExit = false; 1948 uint32_t activeSleepTime = activeSleepTimeUs(); 1949 uint32_t idleSleepTime = idleSleepTimeUs(); 1950 uint32_t sleepTime = idleSleepTime; 1951 uint32_t sleepTimeShift = 0; 1952 Vector< sp<EffectChain> > effectChains; 1953#ifdef DEBUG_CPU_USAGE 1954 ThreadCpuUsage cpu; 1955 const CentralTendencyStatistics& stats = cpu.statistics(); 1956#endif 1957 1958 acquireWakeLock(); 1959 1960 while (!exitPending()) 1961 { 1962#ifdef DEBUG_CPU_USAGE 1963 cpu.sampleAndEnable(); 1964 unsigned n = stats.n(); 1965 // cpu.elapsed() is expensive, so don't call it every loop 1966 if ((n & 127) == 1) { 1967 long long elapsed = cpu.elapsed(); 1968 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1969 double perLoop = elapsed / (double) n; 1970 double perLoop100 = perLoop * 0.01; 1971 double mean = stats.mean(); 1972 double stddev = stats.stddev(); 1973 double minimum = stats.minimum(); 1974 double maximum = stats.maximum(); 1975 cpu.resetStatistics(); 1976 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1977 elapsed * .000000001, n, perLoop * .000001, 1978 mean * .001, 1979 stddev * .001, 1980 minimum * .001, 1981 maximum * .001, 1982 mean / perLoop100, 1983 stddev / perLoop100, 1984 minimum / perLoop100, 1985 maximum / perLoop100); 1986 } 1987 } 1988#endif 1989 processConfigEvents(); 1990 1991 mixerStatus = MIXER_IDLE; 1992 { // scope for mLock 1993 1994 Mutex::Autolock _l(mLock); 1995 1996 if (checkForNewParameters_l()) { 1997 mixBufferSize = mFrameCount * mFrameSize; 1998 // FIXME: Relaxed timing because of a certain device that can't meet latency 1999 // Should be reduced to 2x after the vendor fixes the driver issue 2000 // increase threshold again due to low power audio mode. The way this warning 2001 // threshold is calculated and its usefulness should be reconsidered anyway. 2002 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2003 activeSleepTime = activeSleepTimeUs(); 2004 idleSleepTime = idleSleepTimeUs(); 2005 } 2006 2007 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2008 2009 // put audio hardware into standby after short delay 2010 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2011 mSuspended)) { 2012 if (!mStandby) { 2013 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2014 mOutput->stream->common.standby(&mOutput->stream->common); 2015 mStandby = true; 2016 mBytesWritten = 0; 2017 } 2018 2019 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2020 // we're about to wait, flush the binder command buffer 2021 IPCThreadState::self()->flushCommands(); 2022 2023 if (exitPending()) break; 2024 2025 releaseWakeLock_l(); 2026 // wait until we have something to do... 2027 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 2028 mWaitWorkCV.wait(mLock); 2029 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 2030 acquireWakeLock_l(); 2031 2032 mPrevMixerStatus = MIXER_IDLE; 2033 if (!mMasterMute) { 2034 char value[PROPERTY_VALUE_MAX]; 2035 property_get("ro.audio.silent", value, "0"); 2036 if (atoi(value)) { 2037 ALOGD("Silence is golden"); 2038 setMasterMute_l(true); 2039 } 2040 } 2041 2042 standbyTime = systemTime() + mStandbyTimeInNsecs; 2043 sleepTime = idleSleepTime; 2044 sleepTimeShift = 0; 2045 continue; 2046 } 2047 } 2048 2049 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2050 2051 // prevent any changes in effect chain list and in each effect chain 2052 // during mixing and effect process as the audio buffers could be deleted 2053 // or modified if an effect is created or deleted 2054 lockEffectChains_l(effectChains); 2055 } 2056 2057 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2058 // obtain the presentation timestamp of the next output buffer 2059 int64_t pts; 2060 status_t status = INVALID_OPERATION; 2061 2062 if (NULL != mOutput->stream->get_next_write_timestamp) { 2063 status = mOutput->stream->get_next_write_timestamp( 2064 mOutput->stream, &pts); 2065 } 2066 2067 if (status != NO_ERROR) { 2068 pts = AudioBufferProvider::kInvalidPTS; 2069 } 2070 2071 // mix buffers... 2072 mAudioMixer->process(pts); 2073 // increase sleep time progressively when application underrun condition clears. 2074 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2075 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2076 // such that we would underrun the audio HAL. 2077 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2078 sleepTimeShift--; 2079 } 2080 sleepTime = 0; 2081 standbyTime = systemTime() + mStandbyTimeInNsecs; 2082 //TODO: delay standby when effects have a tail 2083 } else { 2084 // If no tracks are ready, sleep once for the duration of an output 2085 // buffer size, then write 0s to the output 2086 if (sleepTime == 0) { 2087 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2088 sleepTime = activeSleepTime >> sleepTimeShift; 2089 if (sleepTime < kMinThreadSleepTimeUs) { 2090 sleepTime = kMinThreadSleepTimeUs; 2091 } 2092 // reduce sleep time in case of consecutive application underruns to avoid 2093 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2094 // duration we would end up writing less data than needed by the audio HAL if 2095 // the condition persists. 2096 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2097 sleepTimeShift++; 2098 } 2099 } else { 2100 sleepTime = idleSleepTime; 2101 } 2102 } else if (mBytesWritten != 0 || 2103 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2104 memset (mMixBuffer, 0, mixBufferSize); 2105 sleepTime = 0; 2106 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2107 } 2108 // TODO add standby time extension fct of effect tail 2109 } 2110 2111 if (mSuspended) { 2112 sleepTime = suspendSleepTimeUs(); 2113 } 2114 // sleepTime == 0 means we must write to audio hardware 2115 if (sleepTime == 0) { 2116 for (size_t i = 0; i < effectChains.size(); i ++) { 2117 effectChains[i]->process_l(); 2118 } 2119 // enable changes in effect chain 2120 unlockEffectChains(effectChains); 2121 mLastWriteTime = systemTime(); 2122 mInWrite = true; 2123 mBytesWritten += mixBufferSize; 2124 2125 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2126 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2127 mNumWrites++; 2128 mInWrite = false; 2129 nsecs_t now = systemTime(); 2130 nsecs_t delta = now - mLastWriteTime; 2131 if (!mStandby && delta > maxPeriod) { 2132 mNumDelayedWrites++; 2133 if ((now - lastWarning) > kWarningThrottleNs) { 2134 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2135 ns2ms(delta), mNumDelayedWrites, this); 2136 lastWarning = now; 2137 } 2138 if (mStandby) { 2139 longStandbyExit = true; 2140 } 2141 } 2142 mStandby = false; 2143 } else { 2144 // enable changes in effect chain 2145 unlockEffectChains(effectChains); 2146 usleep(sleepTime); 2147 } 2148 2149 // finally let go of all our tracks, without the lock held 2150 // since we can't guarantee the destructors won't acquire that 2151 // same lock. 2152 tracksToRemove.clear(); 2153 2154 // Effect chains will be actually deleted here if they were removed from 2155 // mEffectChains list during mixing or effects processing 2156 effectChains.clear(); 2157 } 2158 2159 if (!mStandby) { 2160 mOutput->stream->common.standby(&mOutput->stream->common); 2161 } 2162 2163 releaseWakeLock(); 2164 2165 ALOGV("MixerThread %p exiting", this); 2166 return false; 2167} 2168 2169// prepareTracks_l() must be called with ThreadBase::mLock held 2170AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2171 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2172{ 2173 2174 mixer_state mixerStatus = MIXER_IDLE; 2175 // find out which tracks need to be processed 2176 size_t count = activeTracks.size(); 2177 size_t mixedTracks = 0; 2178 size_t tracksWithEffect = 0; 2179 2180 float masterVolume = mMasterVolume; 2181 bool masterMute = mMasterMute; 2182 2183 if (masterMute) { 2184 masterVolume = 0; 2185 } 2186 // Delegate master volume control to effect in output mix effect chain if needed 2187 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2188 if (chain != 0) { 2189 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2190 chain->setVolume_l(&v, &v); 2191 masterVolume = (float)((v + (1 << 23)) >> 24); 2192 chain.clear(); 2193 } 2194 2195 for (size_t i=0 ; i<count ; i++) { 2196 sp<Track> t = activeTracks[i].promote(); 2197 if (t == 0) continue; 2198 2199 // this const just means the local variable doesn't change 2200 Track* const track = t.get(); 2201 audio_track_cblk_t* cblk = track->cblk(); 2202 2203 // The first time a track is added we wait 2204 // for all its buffers to be filled before processing it 2205 int name = track->name(); 2206 // make sure that we have enough frames to mix one full buffer. 2207 // enforce this condition only once to enable draining the buffer in case the client 2208 // app does not call stop() and relies on underrun to stop: 2209 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2210 // during last round 2211 uint32_t minFrames = 1; 2212 if (!track->isStopped() && !track->isPausing() && 2213 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2214 if (t->sampleRate() == (int)mSampleRate) { 2215 minFrames = mFrameCount; 2216 } else { 2217 // +1 for rounding and +1 for additional sample needed for interpolation 2218 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2219 // add frames already consumed but not yet released by the resampler 2220 // because cblk->framesReady() will include these frames 2221 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2222 // the minimum track buffer size is normally twice the number of frames necessary 2223 // to fill one buffer and the resampler should not leave more than one buffer worth 2224 // of unreleased frames after each pass, but just in case... 2225 ALOG_ASSERT(minFrames <= cblk->frameCount); 2226 } 2227 } 2228 if ((track->framesReady() >= minFrames) && track->isReady() && 2229 !track->isPaused() && !track->isTerminated()) 2230 { 2231 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2232 2233 mixedTracks++; 2234 2235 // track->mainBuffer() != mMixBuffer means there is an effect chain 2236 // connected to the track 2237 chain.clear(); 2238 if (track->mainBuffer() != mMixBuffer) { 2239 chain = getEffectChain_l(track->sessionId()); 2240 // Delegate volume control to effect in track effect chain if needed 2241 if (chain != 0) { 2242 tracksWithEffect++; 2243 } else { 2244 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2245 name, track->sessionId()); 2246 } 2247 } 2248 2249 2250 int param = AudioMixer::VOLUME; 2251 if (track->mFillingUpStatus == Track::FS_FILLED) { 2252 // no ramp for the first volume setting 2253 track->mFillingUpStatus = Track::FS_ACTIVE; 2254 if (track->mState == TrackBase::RESUMING) { 2255 track->mState = TrackBase::ACTIVE; 2256 param = AudioMixer::RAMP_VOLUME; 2257 } 2258 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2259 } else if (cblk->server != 0) { 2260 // If the track is stopped before the first frame was mixed, 2261 // do not apply ramp 2262 param = AudioMixer::RAMP_VOLUME; 2263 } 2264 2265 // compute volume for this track 2266 uint32_t vl, vr, va; 2267 if (track->isMuted() || track->isPausing() || 2268 mStreamTypes[track->streamType()].mute) { 2269 vl = vr = va = 0; 2270 if (track->isPausing()) { 2271 track->setPaused(); 2272 } 2273 } else { 2274 2275 // read original volumes with volume control 2276 float typeVolume = mStreamTypes[track->streamType()].volume; 2277 float v = masterVolume * typeVolume; 2278 uint32_t vlr = cblk->getVolumeLR(); 2279 vl = vlr & 0xFFFF; 2280 vr = vlr >> 16; 2281 // track volumes come from shared memory, so can't be trusted and must be clamped 2282 if (vl > MAX_GAIN_INT) { 2283 ALOGV("Track left volume out of range: %04X", vl); 2284 vl = MAX_GAIN_INT; 2285 } 2286 if (vr > MAX_GAIN_INT) { 2287 ALOGV("Track right volume out of range: %04X", vr); 2288 vr = MAX_GAIN_INT; 2289 } 2290 // now apply the master volume and stream type volume 2291 vl = (uint32_t)(v * vl) << 12; 2292 vr = (uint32_t)(v * vr) << 12; 2293 // assuming master volume and stream type volume each go up to 1.0, 2294 // vl and vr are now in 8.24 format 2295 2296 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2297 // send level comes from shared memory and so may be corrupt 2298 if (sendLevel > MAX_GAIN_INT) { 2299 ALOGV("Track send level out of range: %04X", sendLevel); 2300 sendLevel = MAX_GAIN_INT; 2301 } 2302 va = (uint32_t)(v * sendLevel); 2303 } 2304 // Delegate volume control to effect in track effect chain if needed 2305 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2306 // Do not ramp volume if volume is controlled by effect 2307 param = AudioMixer::VOLUME; 2308 track->mHasVolumeController = true; 2309 } else { 2310 // force no volume ramp when volume controller was just disabled or removed 2311 // from effect chain to avoid volume spike 2312 if (track->mHasVolumeController) { 2313 param = AudioMixer::VOLUME; 2314 } 2315 track->mHasVolumeController = false; 2316 } 2317 2318 // Convert volumes from 8.24 to 4.12 format 2319 // This additional clamping is needed in case chain->setVolume_l() overshot 2320 vl = (vl + (1 << 11)) >> 12; 2321 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2322 vr = (vr + (1 << 11)) >> 12; 2323 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2324 2325 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2326 2327 // XXX: these things DON'T need to be done each time 2328 mAudioMixer->setBufferProvider(name, track); 2329 mAudioMixer->enable(name); 2330 2331 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2332 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2333 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2334 mAudioMixer->setParameter( 2335 name, 2336 AudioMixer::TRACK, 2337 AudioMixer::FORMAT, (void *)track->format()); 2338 mAudioMixer->setParameter( 2339 name, 2340 AudioMixer::TRACK, 2341 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2342 mAudioMixer->setParameter( 2343 name, 2344 AudioMixer::RESAMPLE, 2345 AudioMixer::SAMPLE_RATE, 2346 (void *)(cblk->sampleRate)); 2347 mAudioMixer->setParameter( 2348 name, 2349 AudioMixer::TRACK, 2350 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2351 mAudioMixer->setParameter( 2352 name, 2353 AudioMixer::TRACK, 2354 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2355 2356 // reset retry count 2357 track->mRetryCount = kMaxTrackRetries; 2358 // If one track is ready, set the mixer ready if: 2359 // - the mixer was not ready during previous round OR 2360 // - no other track is not ready 2361 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2362 mixerStatus != MIXER_TRACKS_ENABLED) { 2363 mixerStatus = MIXER_TRACKS_READY; 2364 } 2365 } else { 2366 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2367 if (track->isStopped()) { 2368 track->reset(); 2369 } 2370 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2371 // We have consumed all the buffers of this track. 2372 // Remove it from the list of active tracks. 2373 tracksToRemove->add(track); 2374 } else { 2375 // No buffers for this track. Give it a few chances to 2376 // fill a buffer, then remove it from active list. 2377 if (--(track->mRetryCount) <= 0) { 2378 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2379 tracksToRemove->add(track); 2380 // indicate to client process that the track was disabled because of underrun 2381 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2382 // If one track is not ready, mark the mixer also not ready if: 2383 // - the mixer was ready during previous round OR 2384 // - no other track is ready 2385 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2386 mixerStatus != MIXER_TRACKS_READY) { 2387 mixerStatus = MIXER_TRACKS_ENABLED; 2388 } 2389 } 2390 mAudioMixer->disable(name); 2391 } 2392 } 2393 2394 // remove all the tracks that need to be... 2395 count = tracksToRemove->size(); 2396 if (CC_UNLIKELY(count)) { 2397 for (size_t i=0 ; i<count ; i++) { 2398 const sp<Track>& track = tracksToRemove->itemAt(i); 2399 mActiveTracks.remove(track); 2400 if (track->mainBuffer() != mMixBuffer) { 2401 chain = getEffectChain_l(track->sessionId()); 2402 if (chain != 0) { 2403 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2404 chain->decActiveTrackCnt(); 2405 } 2406 } 2407 if (track->isTerminated()) { 2408 removeTrack_l(track); 2409 } 2410 } 2411 } 2412 2413 // mix buffer must be cleared if all tracks are connected to an 2414 // effect chain as in this case the mixer will not write to 2415 // mix buffer and track effects will accumulate into it 2416 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2417 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2418 } 2419 2420 mPrevMixerStatus = mixerStatus; 2421 return mixerStatus; 2422} 2423 2424void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2425{ 2426 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2427 this, streamType, mTracks.size()); 2428 Mutex::Autolock _l(mLock); 2429 2430 size_t size = mTracks.size(); 2431 for (size_t i = 0; i < size; i++) { 2432 sp<Track> t = mTracks[i]; 2433 if (t->streamType() == streamType) { 2434 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2435 t->mCblk->cv.signal(); 2436 } 2437 } 2438} 2439 2440void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2441{ 2442 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2443 this, streamType, valid); 2444 Mutex::Autolock _l(mLock); 2445 2446 mStreamTypes[streamType].valid = valid; 2447} 2448 2449// getTrackName_l() must be called with ThreadBase::mLock held 2450int AudioFlinger::MixerThread::getTrackName_l() 2451{ 2452 return mAudioMixer->getTrackName(); 2453} 2454 2455// deleteTrackName_l() must be called with ThreadBase::mLock held 2456void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2457{ 2458 ALOGV("remove track (%d) and delete from mixer", name); 2459 mAudioMixer->deleteTrackName(name); 2460} 2461 2462// checkForNewParameters_l() must be called with ThreadBase::mLock held 2463bool AudioFlinger::MixerThread::checkForNewParameters_l() 2464{ 2465 bool reconfig = false; 2466 2467 while (!mNewParameters.isEmpty()) { 2468 status_t status = NO_ERROR; 2469 String8 keyValuePair = mNewParameters[0]; 2470 AudioParameter param = AudioParameter(keyValuePair); 2471 int value; 2472 2473 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2474 reconfig = true; 2475 } 2476 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2477 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2478 status = BAD_VALUE; 2479 } else { 2480 reconfig = true; 2481 } 2482 } 2483 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2484 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2485 status = BAD_VALUE; 2486 } else { 2487 reconfig = true; 2488 } 2489 } 2490 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2491 // do not accept frame count changes if tracks are open as the track buffer 2492 // size depends on frame count and correct behavior would not be guaranteed 2493 // if frame count is changed after track creation 2494 if (!mTracks.isEmpty()) { 2495 status = INVALID_OPERATION; 2496 } else { 2497 reconfig = true; 2498 } 2499 } 2500 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2501 // when changing the audio output device, call addBatteryData to notify 2502 // the change 2503 if ((int)mDevice != value) { 2504 uint32_t params = 0; 2505 // check whether speaker is on 2506 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2507 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2508 } 2509 2510 int deviceWithoutSpeaker 2511 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2512 // check if any other device (except speaker) is on 2513 if (value & deviceWithoutSpeaker ) { 2514 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2515 } 2516 2517 if (params != 0) { 2518 addBatteryData(params); 2519 } 2520 } 2521 2522 // forward device change to effects that have requested to be 2523 // aware of attached audio device. 2524 mDevice = (uint32_t)value; 2525 for (size_t i = 0; i < mEffectChains.size(); i++) { 2526 mEffectChains[i]->setDevice_l(mDevice); 2527 } 2528 } 2529 2530 if (status == NO_ERROR) { 2531 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2532 keyValuePair.string()); 2533 if (!mStandby && status == INVALID_OPERATION) { 2534 mOutput->stream->common.standby(&mOutput->stream->common); 2535 mStandby = true; 2536 mBytesWritten = 0; 2537 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2538 keyValuePair.string()); 2539 } 2540 if (status == NO_ERROR && reconfig) { 2541 delete mAudioMixer; 2542 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2543 mAudioMixer = NULL; 2544 readOutputParameters(); 2545 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2546 for (size_t i = 0; i < mTracks.size() ; i++) { 2547 int name = getTrackName_l(); 2548 if (name < 0) break; 2549 mTracks[i]->mName = name; 2550 // limit track sample rate to 2 x new output sample rate 2551 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2552 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2553 } 2554 } 2555 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2556 } 2557 } 2558 2559 mNewParameters.removeAt(0); 2560 2561 mParamStatus = status; 2562 mParamCond.signal(); 2563 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2564 // already timed out waiting for the status and will never signal the condition. 2565 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2566 } 2567 return reconfig; 2568} 2569 2570status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2571{ 2572 const size_t SIZE = 256; 2573 char buffer[SIZE]; 2574 String8 result; 2575 2576 PlaybackThread::dumpInternals(fd, args); 2577 2578 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2579 result.append(buffer); 2580 write(fd, result.string(), result.size()); 2581 return NO_ERROR; 2582} 2583 2584uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2585{ 2586 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2587} 2588 2589uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2590{ 2591 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2592} 2593 2594// ---------------------------------------------------------------------------- 2595AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2596 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2597 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2598 // mLeftVolFloat, mRightVolFloat 2599 // mLeftVolShort, mRightVolShort 2600{ 2601} 2602 2603AudioFlinger::DirectOutputThread::~DirectOutputThread() 2604{ 2605} 2606 2607void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2608{ 2609 // Do not apply volume on compressed audio 2610 if (!audio_is_linear_pcm(mFormat)) { 2611 return; 2612 } 2613 2614 // convert to signed 16 bit before volume calculation 2615 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2616 size_t count = mFrameCount * mChannelCount; 2617 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2618 int16_t *dst = mMixBuffer + count-1; 2619 while(count--) { 2620 *dst-- = (int16_t)(*src--^0x80) << 8; 2621 } 2622 } 2623 2624 size_t frameCount = mFrameCount; 2625 int16_t *out = mMixBuffer; 2626 if (ramp) { 2627 if (mChannelCount == 1) { 2628 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2629 int32_t vlInc = d / (int32_t)frameCount; 2630 int32_t vl = ((int32_t)mLeftVolShort << 16); 2631 do { 2632 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2633 out++; 2634 vl += vlInc; 2635 } while (--frameCount); 2636 2637 } else { 2638 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2639 int32_t vlInc = d / (int32_t)frameCount; 2640 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2641 int32_t vrInc = d / (int32_t)frameCount; 2642 int32_t vl = ((int32_t)mLeftVolShort << 16); 2643 int32_t vr = ((int32_t)mRightVolShort << 16); 2644 do { 2645 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2646 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2647 out += 2; 2648 vl += vlInc; 2649 vr += vrInc; 2650 } while (--frameCount); 2651 } 2652 } else { 2653 if (mChannelCount == 1) { 2654 do { 2655 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2656 out++; 2657 } while (--frameCount); 2658 } else { 2659 do { 2660 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2661 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2662 out += 2; 2663 } while (--frameCount); 2664 } 2665 } 2666 2667 // convert back to unsigned 8 bit after volume calculation 2668 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2669 size_t count = mFrameCount * mChannelCount; 2670 int16_t *src = mMixBuffer; 2671 uint8_t *dst = (uint8_t *)mMixBuffer; 2672 while(count--) { 2673 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2674 } 2675 } 2676 2677 mLeftVolShort = leftVol; 2678 mRightVolShort = rightVol; 2679} 2680 2681bool AudioFlinger::DirectOutputThread::threadLoop() 2682{ 2683 mixer_state mixerStatus = MIXER_IDLE; 2684 sp<Track> trackToRemove; 2685 sp<Track> activeTrack; 2686 nsecs_t standbyTime = systemTime(); 2687 size_t mixBufferSize = mFrameCount*mFrameSize; 2688 uint32_t activeSleepTime = activeSleepTimeUs(); 2689 uint32_t idleSleepTime = idleSleepTimeUs(); 2690 uint32_t sleepTime = idleSleepTime; 2691 // use shorter standby delay as on normal output to release 2692 // hardware resources as soon as possible 2693 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2694 2695 acquireWakeLock(); 2696 2697 while (!exitPending()) 2698 { 2699 bool rampVolume; 2700 uint16_t leftVol; 2701 uint16_t rightVol; 2702 Vector< sp<EffectChain> > effectChains; 2703 2704 processConfigEvents(); 2705 2706 mixerStatus = MIXER_IDLE; 2707 2708 { // scope for the mLock 2709 2710 Mutex::Autolock _l(mLock); 2711 2712 if (checkForNewParameters_l()) { 2713 mixBufferSize = mFrameCount*mFrameSize; 2714 activeSleepTime = activeSleepTimeUs(); 2715 idleSleepTime = idleSleepTimeUs(); 2716 standbyDelay = microseconds(activeSleepTime*2); 2717 } 2718 2719 // put audio hardware into standby after short delay 2720 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2721 mSuspended)) { 2722 // wait until we have something to do... 2723 if (!mStandby) { 2724 ALOGV("Audio hardware entering standby, mixer %p", this); 2725 mOutput->stream->common.standby(&mOutput->stream->common); 2726 mStandby = true; 2727 mBytesWritten = 0; 2728 } 2729 2730 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2731 // we're about to wait, flush the binder command buffer 2732 IPCThreadState::self()->flushCommands(); 2733 2734 if (exitPending()) break; 2735 2736 releaseWakeLock_l(); 2737 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2738 mWaitWorkCV.wait(mLock); 2739 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2740 acquireWakeLock_l(); 2741 2742 if (!mMasterMute) { 2743 char value[PROPERTY_VALUE_MAX]; 2744 property_get("ro.audio.silent", value, "0"); 2745 if (atoi(value)) { 2746 ALOGD("Silence is golden"); 2747 setMasterMute_l(true); 2748 } 2749 } 2750 2751 standbyTime = systemTime() + standbyDelay; 2752 sleepTime = idleSleepTime; 2753 continue; 2754 } 2755 } 2756 2757 effectChains = mEffectChains; 2758 2759 // find out which tracks need to be processed 2760 if (mActiveTracks.size() != 0) { 2761 sp<Track> t = mActiveTracks[0].promote(); 2762 if (t == 0) continue; 2763 2764 Track* const track = t.get(); 2765 audio_track_cblk_t* cblk = track->cblk(); 2766 2767 // The first time a track is added we wait 2768 // for all its buffers to be filled before processing it 2769 if (cblk->framesReady() && track->isReady() && 2770 !track->isPaused() && !track->isTerminated()) 2771 { 2772 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2773 2774 if (track->mFillingUpStatus == Track::FS_FILLED) { 2775 track->mFillingUpStatus = Track::FS_ACTIVE; 2776 mLeftVolFloat = mRightVolFloat = 0; 2777 mLeftVolShort = mRightVolShort = 0; 2778 if (track->mState == TrackBase::RESUMING) { 2779 track->mState = TrackBase::ACTIVE; 2780 rampVolume = true; 2781 } 2782 } else if (cblk->server != 0) { 2783 // If the track is stopped before the first frame was mixed, 2784 // do not apply ramp 2785 rampVolume = true; 2786 } 2787 // compute volume for this track 2788 float left, right; 2789 if (track->isMuted() || mMasterMute || track->isPausing() || 2790 mStreamTypes[track->streamType()].mute) { 2791 left = right = 0; 2792 if (track->isPausing()) { 2793 track->setPaused(); 2794 } 2795 } else { 2796 float typeVolume = mStreamTypes[track->streamType()].volume; 2797 float v = mMasterVolume * typeVolume; 2798 uint32_t vlr = cblk->getVolumeLR(); 2799 float v_clamped = v * (vlr & 0xFFFF); 2800 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2801 left = v_clamped/MAX_GAIN; 2802 v_clamped = v * (vlr >> 16); 2803 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2804 right = v_clamped/MAX_GAIN; 2805 } 2806 2807 if (left != mLeftVolFloat || right != mRightVolFloat) { 2808 mLeftVolFloat = left; 2809 mRightVolFloat = right; 2810 2811 // If audio HAL implements volume control, 2812 // force software volume to nominal value 2813 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2814 left = 1.0f; 2815 right = 1.0f; 2816 } 2817 2818 // Convert volumes from float to 8.24 2819 uint32_t vl = (uint32_t)(left * (1 << 24)); 2820 uint32_t vr = (uint32_t)(right * (1 << 24)); 2821 2822 // Delegate volume control to effect in track effect chain if needed 2823 // only one effect chain can be present on DirectOutputThread, so if 2824 // there is one, the track is connected to it 2825 if (!effectChains.isEmpty()) { 2826 // Do not ramp volume if volume is controlled by effect 2827 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2828 rampVolume = false; 2829 } 2830 } 2831 2832 // Convert volumes from 8.24 to 4.12 format 2833 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2834 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2835 leftVol = (uint16_t)v_clamped; 2836 v_clamped = (vr + (1 << 11)) >> 12; 2837 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2838 rightVol = (uint16_t)v_clamped; 2839 } else { 2840 leftVol = mLeftVolShort; 2841 rightVol = mRightVolShort; 2842 rampVolume = false; 2843 } 2844 2845 // reset retry count 2846 track->mRetryCount = kMaxTrackRetriesDirect; 2847 activeTrack = t; 2848 mixerStatus = MIXER_TRACKS_READY; 2849 } else { 2850 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2851 if (track->isStopped()) { 2852 track->reset(); 2853 } 2854 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2855 // We have consumed all the buffers of this track. 2856 // Remove it from the list of active tracks. 2857 trackToRemove = track; 2858 } else { 2859 // No buffers for this track. Give it a few chances to 2860 // fill a buffer, then remove it from active list. 2861 if (--(track->mRetryCount) <= 0) { 2862 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2863 trackToRemove = track; 2864 } else { 2865 mixerStatus = MIXER_TRACKS_ENABLED; 2866 } 2867 } 2868 } 2869 } 2870 2871 // remove all the tracks that need to be... 2872 if (CC_UNLIKELY(trackToRemove != 0)) { 2873 mActiveTracks.remove(trackToRemove); 2874 if (!effectChains.isEmpty()) { 2875 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2876 trackToRemove->sessionId()); 2877 effectChains[0]->decActiveTrackCnt(); 2878 } 2879 if (trackToRemove->isTerminated()) { 2880 removeTrack_l(trackToRemove); 2881 } 2882 } 2883 2884 lockEffectChains_l(effectChains); 2885 } 2886 2887 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2888 AudioBufferProvider::Buffer buffer; 2889 size_t frameCount = mFrameCount; 2890 int8_t *curBuf = (int8_t *)mMixBuffer; 2891 // output audio to hardware 2892 while (frameCount) { 2893 buffer.frameCount = frameCount; 2894 activeTrack->getNextBuffer(&buffer, 2895 AudioBufferProvider::kInvalidPTS); 2896 if (CC_UNLIKELY(buffer.raw == NULL)) { 2897 memset(curBuf, 0, frameCount * mFrameSize); 2898 break; 2899 } 2900 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2901 frameCount -= buffer.frameCount; 2902 curBuf += buffer.frameCount * mFrameSize; 2903 activeTrack->releaseBuffer(&buffer); 2904 } 2905 sleepTime = 0; 2906 standbyTime = systemTime() + standbyDelay; 2907 } else { 2908 if (sleepTime == 0) { 2909 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2910 sleepTime = activeSleepTime; 2911 } else { 2912 sleepTime = idleSleepTime; 2913 } 2914 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2915 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2916 sleepTime = 0; 2917 } 2918 } 2919 2920 if (mSuspended) { 2921 sleepTime = suspendSleepTimeUs(); 2922 } 2923 // sleepTime == 0 means we must write to audio hardware 2924 if (sleepTime == 0) { 2925 if (mixerStatus == MIXER_TRACKS_READY) { 2926 applyVolume(leftVol, rightVol, rampVolume); 2927 } 2928 for (size_t i = 0; i < effectChains.size(); i ++) { 2929 effectChains[i]->process_l(); 2930 } 2931 unlockEffectChains(effectChains); 2932 2933 mLastWriteTime = systemTime(); 2934 mInWrite = true; 2935 mBytesWritten += mixBufferSize; 2936 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2937 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2938 mNumWrites++; 2939 mInWrite = false; 2940 mStandby = false; 2941 } else { 2942 unlockEffectChains(effectChains); 2943 usleep(sleepTime); 2944 } 2945 2946 // finally let go of removed track, without the lock held 2947 // since we can't guarantee the destructors won't acquire that 2948 // same lock. 2949 trackToRemove.clear(); 2950 activeTrack.clear(); 2951 2952 // Effect chains will be actually deleted here if they were removed from 2953 // mEffectChains list during mixing or effects processing 2954 effectChains.clear(); 2955 } 2956 2957 if (!mStandby) { 2958 mOutput->stream->common.standby(&mOutput->stream->common); 2959 } 2960 2961 releaseWakeLock(); 2962 2963 ALOGV("DirectOutputThread %p exiting", this); 2964 return false; 2965} 2966 2967// getTrackName_l() must be called with ThreadBase::mLock held 2968int AudioFlinger::DirectOutputThread::getTrackName_l() 2969{ 2970 return 0; 2971} 2972 2973// deleteTrackName_l() must be called with ThreadBase::mLock held 2974void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2975{ 2976} 2977 2978// checkForNewParameters_l() must be called with ThreadBase::mLock held 2979bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2980{ 2981 bool reconfig = false; 2982 2983 while (!mNewParameters.isEmpty()) { 2984 status_t status = NO_ERROR; 2985 String8 keyValuePair = mNewParameters[0]; 2986 AudioParameter param = AudioParameter(keyValuePair); 2987 int value; 2988 2989 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2990 // do not accept frame count changes if tracks are open as the track buffer 2991 // size depends on frame count and correct behavior would not be garantied 2992 // if frame count is changed after track creation 2993 if (!mTracks.isEmpty()) { 2994 status = INVALID_OPERATION; 2995 } else { 2996 reconfig = true; 2997 } 2998 } 2999 if (status == NO_ERROR) { 3000 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3001 keyValuePair.string()); 3002 if (!mStandby && status == INVALID_OPERATION) { 3003 mOutput->stream->common.standby(&mOutput->stream->common); 3004 mStandby = true; 3005 mBytesWritten = 0; 3006 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3007 keyValuePair.string()); 3008 } 3009 if (status == NO_ERROR && reconfig) { 3010 readOutputParameters(); 3011 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3012 } 3013 } 3014 3015 mNewParameters.removeAt(0); 3016 3017 mParamStatus = status; 3018 mParamCond.signal(); 3019 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3020 // already timed out waiting for the status and will never signal the condition. 3021 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3022 } 3023 return reconfig; 3024} 3025 3026uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3027{ 3028 uint32_t time; 3029 if (audio_is_linear_pcm(mFormat)) { 3030 time = PlaybackThread::activeSleepTimeUs(); 3031 } else { 3032 time = 10000; 3033 } 3034 return time; 3035} 3036 3037uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3038{ 3039 uint32_t time; 3040 if (audio_is_linear_pcm(mFormat)) { 3041 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3042 } else { 3043 time = 10000; 3044 } 3045 return time; 3046} 3047 3048uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3049{ 3050 uint32_t time; 3051 if (audio_is_linear_pcm(mFormat)) { 3052 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3053 } else { 3054 time = 10000; 3055 } 3056 return time; 3057} 3058 3059 3060// ---------------------------------------------------------------------------- 3061 3062AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3063 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3064 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3065 mWaitTimeMs(UINT_MAX) 3066{ 3067 addOutputTrack(mainThread); 3068} 3069 3070AudioFlinger::DuplicatingThread::~DuplicatingThread() 3071{ 3072 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3073 mOutputTracks[i]->destroy(); 3074 } 3075} 3076 3077bool AudioFlinger::DuplicatingThread::threadLoop() 3078{ 3079 Vector< sp<Track> > tracksToRemove; 3080 mixer_state mixerStatus = MIXER_IDLE; 3081 nsecs_t standbyTime = systemTime(); 3082 size_t mixBufferSize = mFrameCount*mFrameSize; 3083 SortedVector< sp<OutputTrack> > outputTracks; 3084 uint32_t writeFrames = 0; 3085 uint32_t activeSleepTime = activeSleepTimeUs(); 3086 uint32_t idleSleepTime = idleSleepTimeUs(); 3087 uint32_t sleepTime = idleSleepTime; 3088 Vector< sp<EffectChain> > effectChains; 3089 3090 acquireWakeLock(); 3091 3092 while (!exitPending()) 3093 { 3094 processConfigEvents(); 3095 3096 mixerStatus = MIXER_IDLE; 3097 { // scope for the mLock 3098 3099 Mutex::Autolock _l(mLock); 3100 3101 if (checkForNewParameters_l()) { 3102 mixBufferSize = mFrameCount*mFrameSize; 3103 updateWaitTime(); 3104 activeSleepTime = activeSleepTimeUs(); 3105 idleSleepTime = idleSleepTimeUs(); 3106 } 3107 3108 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3109 3110 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3111 outputTracks.add(mOutputTracks[i]); 3112 } 3113 3114 // put audio hardware into standby after short delay 3115 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3116 mSuspended)) { 3117 if (!mStandby) { 3118 for (size_t i = 0; i < outputTracks.size(); i++) { 3119 outputTracks[i]->stop(); 3120 } 3121 mStandby = true; 3122 mBytesWritten = 0; 3123 } 3124 3125 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3126 // we're about to wait, flush the binder command buffer 3127 IPCThreadState::self()->flushCommands(); 3128 outputTracks.clear(); 3129 3130 if (exitPending()) break; 3131 3132 releaseWakeLock_l(); 3133 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3134 mWaitWorkCV.wait(mLock); 3135 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3136 acquireWakeLock_l(); 3137 3138 mPrevMixerStatus = MIXER_IDLE; 3139 if (!mMasterMute) { 3140 char value[PROPERTY_VALUE_MAX]; 3141 property_get("ro.audio.silent", value, "0"); 3142 if (atoi(value)) { 3143 ALOGD("Silence is golden"); 3144 setMasterMute_l(true); 3145 } 3146 } 3147 3148 standbyTime = systemTime() + mStandbyTimeInNsecs; 3149 sleepTime = idleSleepTime; 3150 continue; 3151 } 3152 } 3153 3154 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3155 3156 // prevent any changes in effect chain list and in each effect chain 3157 // during mixing and effect process as the audio buffers could be deleted 3158 // or modified if an effect is created or deleted 3159 lockEffectChains_l(effectChains); 3160 } 3161 3162 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3163 // mix buffers... 3164 if (outputsReady(outputTracks)) { 3165 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3166 } else { 3167 memset(mMixBuffer, 0, mixBufferSize); 3168 } 3169 sleepTime = 0; 3170 writeFrames = mFrameCount; 3171 } else { 3172 if (sleepTime == 0) { 3173 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3174 sleepTime = activeSleepTime; 3175 } else { 3176 sleepTime = idleSleepTime; 3177 } 3178 } else if (mBytesWritten != 0) { 3179 // flush remaining overflow buffers in output tracks 3180 for (size_t i = 0; i < outputTracks.size(); i++) { 3181 if (outputTracks[i]->isActive()) { 3182 sleepTime = 0; 3183 writeFrames = 0; 3184 memset(mMixBuffer, 0, mixBufferSize); 3185 break; 3186 } 3187 } 3188 } 3189 } 3190 3191 if (mSuspended) { 3192 sleepTime = suspendSleepTimeUs(); 3193 } 3194 // sleepTime == 0 means we must write to audio hardware 3195 if (sleepTime == 0) { 3196 for (size_t i = 0; i < effectChains.size(); i ++) { 3197 effectChains[i]->process_l(); 3198 } 3199 // enable changes in effect chain 3200 unlockEffectChains(effectChains); 3201 3202 standbyTime = systemTime() + mStandbyTimeInNsecs; 3203 for (size_t i = 0; i < outputTracks.size(); i++) { 3204 outputTracks[i]->write(mMixBuffer, writeFrames); 3205 } 3206 mStandby = false; 3207 mBytesWritten += mixBufferSize; 3208 } else { 3209 // enable changes in effect chain 3210 unlockEffectChains(effectChains); 3211 usleep(sleepTime); 3212 } 3213 3214 // finally let go of all our tracks, without the lock held 3215 // since we can't guarantee the destructors won't acquire that 3216 // same lock. 3217 tracksToRemove.clear(); 3218 outputTracks.clear(); 3219 3220 // Effect chains will be actually deleted here if they were removed from 3221 // mEffectChains list during mixing or effects processing 3222 effectChains.clear(); 3223 } 3224 3225 releaseWakeLock(); 3226 3227 return false; 3228} 3229 3230void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3231{ 3232 // FIXME explain this formula 3233 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3234 OutputTrack *outputTrack = new OutputTrack(thread, 3235 this, 3236 mSampleRate, 3237 mFormat, 3238 mChannelMask, 3239 frameCount); 3240 if (outputTrack->cblk() != NULL) { 3241 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3242 mOutputTracks.add(outputTrack); 3243 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3244 updateWaitTime(); 3245 } 3246} 3247 3248void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3249{ 3250 Mutex::Autolock _l(mLock); 3251 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3252 if (mOutputTracks[i]->thread() == thread) { 3253 mOutputTracks[i]->destroy(); 3254 mOutputTracks.removeAt(i); 3255 updateWaitTime(); 3256 return; 3257 } 3258 } 3259 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3260} 3261 3262void AudioFlinger::DuplicatingThread::updateWaitTime() 3263{ 3264 mWaitTimeMs = UINT_MAX; 3265 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3266 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3267 if (strong != 0) { 3268 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3269 if (waitTimeMs < mWaitTimeMs) { 3270 mWaitTimeMs = waitTimeMs; 3271 } 3272 } 3273 } 3274} 3275 3276 3277bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3278{ 3279 for (size_t i = 0; i < outputTracks.size(); i++) { 3280 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3281 if (thread == 0) { 3282 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3283 return false; 3284 } 3285 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3286 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3287 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3288 return false; 3289 } 3290 } 3291 return true; 3292} 3293 3294uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3295{ 3296 return (mWaitTimeMs * 1000) / 2; 3297} 3298 3299// ---------------------------------------------------------------------------- 3300 3301// TrackBase constructor must be called with AudioFlinger::mLock held 3302AudioFlinger::ThreadBase::TrackBase::TrackBase( 3303 ThreadBase *thread, 3304 const sp<Client>& client, 3305 uint32_t sampleRate, 3306 audio_format_t format, 3307 uint32_t channelMask, 3308 int frameCount, 3309 uint32_t flags, 3310 const sp<IMemory>& sharedBuffer, 3311 int sessionId) 3312 : RefBase(), 3313 mThread(thread), 3314 mClient(client), 3315 mCblk(NULL), 3316 // mBuffer 3317 // mBufferEnd 3318 mFrameCount(0), 3319 mState(IDLE), 3320 mFormat(format), 3321 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3322 mSessionId(sessionId) 3323 // mChannelCount 3324 // mChannelMask 3325{ 3326 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3327 3328 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3329 size_t size = sizeof(audio_track_cblk_t); 3330 uint8_t channelCount = popcount(channelMask); 3331 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3332 if (sharedBuffer == 0) { 3333 size += bufferSize; 3334 } 3335 3336 if (client != NULL) { 3337 mCblkMemory = client->heap()->allocate(size); 3338 if (mCblkMemory != 0) { 3339 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3340 if (mCblk != NULL) { // construct the shared structure in-place. 3341 new(mCblk) audio_track_cblk_t(); 3342 // clear all buffers 3343 mCblk->frameCount = frameCount; 3344 mCblk->sampleRate = sampleRate; 3345 mChannelCount = channelCount; 3346 mChannelMask = channelMask; 3347 if (sharedBuffer == 0) { 3348 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3349 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3350 // Force underrun condition to avoid false underrun callback until first data is 3351 // written to buffer (other flags are cleared) 3352 mCblk->flags = CBLK_UNDERRUN_ON; 3353 } else { 3354 mBuffer = sharedBuffer->pointer(); 3355 } 3356 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3357 } 3358 } else { 3359 ALOGE("not enough memory for AudioTrack size=%u", size); 3360 client->heap()->dump("AudioTrack"); 3361 return; 3362 } 3363 } else { 3364 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3365 // construct the shared structure in-place. 3366 new(mCblk) audio_track_cblk_t(); 3367 // clear all buffers 3368 mCblk->frameCount = frameCount; 3369 mCblk->sampleRate = sampleRate; 3370 mChannelCount = channelCount; 3371 mChannelMask = channelMask; 3372 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3373 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3374 // Force underrun condition to avoid false underrun callback until first data is 3375 // written to buffer (other flags are cleared) 3376 mCblk->flags = CBLK_UNDERRUN_ON; 3377 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3378 } 3379} 3380 3381AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3382{ 3383 if (mCblk != NULL) { 3384 if (mClient == 0) { 3385 delete mCblk; 3386 } else { 3387 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3388 } 3389 } 3390 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3391 if (mClient != 0) { 3392 // Client destructor must run with AudioFlinger mutex locked 3393 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3394 // If the client's reference count drops to zero, the associated destructor 3395 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3396 // relying on the automatic clear() at end of scope. 3397 mClient.clear(); 3398 } 3399} 3400 3401void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3402{ 3403 buffer->raw = NULL; 3404 mFrameCount = buffer->frameCount; 3405 step(); 3406 buffer->frameCount = 0; 3407} 3408 3409bool AudioFlinger::ThreadBase::TrackBase::step() { 3410 bool result; 3411 audio_track_cblk_t* cblk = this->cblk(); 3412 3413 result = cblk->stepServer(mFrameCount); 3414 if (!result) { 3415 ALOGV("stepServer failed acquiring cblk mutex"); 3416 mFlags |= STEPSERVER_FAILED; 3417 } 3418 return result; 3419} 3420 3421void AudioFlinger::ThreadBase::TrackBase::reset() { 3422 audio_track_cblk_t* cblk = this->cblk(); 3423 3424 cblk->user = 0; 3425 cblk->server = 0; 3426 cblk->userBase = 0; 3427 cblk->serverBase = 0; 3428 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3429 ALOGV("TrackBase::reset"); 3430} 3431 3432int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3433 return (int)mCblk->sampleRate; 3434} 3435 3436void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3437 audio_track_cblk_t* cblk = this->cblk(); 3438 size_t frameSize = cblk->frameSize; 3439 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3440 int8_t *bufferEnd = bufferStart + frames * frameSize; 3441 3442 // Check validity of returned pointer in case the track control block would have been corrupted. 3443 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3444 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3445 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3446 server %d, serverBase %d, user %d, userBase %d", 3447 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3448 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3449 return NULL; 3450 } 3451 3452 return bufferStart; 3453} 3454 3455// ---------------------------------------------------------------------------- 3456 3457// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3458AudioFlinger::PlaybackThread::Track::Track( 3459 PlaybackThread *thread, 3460 const sp<Client>& client, 3461 audio_stream_type_t streamType, 3462 uint32_t sampleRate, 3463 audio_format_t format, 3464 uint32_t channelMask, 3465 int frameCount, 3466 const sp<IMemory>& sharedBuffer, 3467 int sessionId) 3468 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3469 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3470 mAuxEffectId(0), mHasVolumeController(false) 3471{ 3472 if (mCblk != NULL) { 3473 if (thread != NULL) { 3474 mName = thread->getTrackName_l(); 3475 mMainBuffer = thread->mixBuffer(); 3476 } 3477 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3478 if (mName < 0) { 3479 ALOGE("no more track names available"); 3480 } 3481 mStreamType = streamType; 3482 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3483 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3484 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3485 } 3486} 3487 3488AudioFlinger::PlaybackThread::Track::~Track() 3489{ 3490 ALOGV("PlaybackThread::Track destructor"); 3491 sp<ThreadBase> thread = mThread.promote(); 3492 if (thread != 0) { 3493 Mutex::Autolock _l(thread->mLock); 3494 mState = TERMINATED; 3495 } 3496} 3497 3498void AudioFlinger::PlaybackThread::Track::destroy() 3499{ 3500 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3501 // by removing it from mTracks vector, so there is a risk that this Tracks's 3502 // destructor is called. As the destructor needs to lock mLock, 3503 // we must acquire a strong reference on this Track before locking mLock 3504 // here so that the destructor is called only when exiting this function. 3505 // On the other hand, as long as Track::destroy() is only called by 3506 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3507 // this Track with its member mTrack. 3508 sp<Track> keep(this); 3509 { // scope for mLock 3510 sp<ThreadBase> thread = mThread.promote(); 3511 if (thread != 0) { 3512 if (!isOutputTrack()) { 3513 if (mState == ACTIVE || mState == RESUMING) { 3514 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3515 3516 // to track the speaker usage 3517 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3518 } 3519 AudioSystem::releaseOutput(thread->id()); 3520 } 3521 Mutex::Autolock _l(thread->mLock); 3522 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3523 playbackThread->destroyTrack_l(this); 3524 } 3525 } 3526} 3527 3528void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3529{ 3530 uint32_t vlr = mCblk->getVolumeLR(); 3531 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", 3532 mName - AudioMixer::TRACK0, 3533 (mClient == 0) ? getpid_cached : mClient->pid(), 3534 mStreamType, 3535 mFormat, 3536 mChannelMask, 3537 mSessionId, 3538 mFrameCount, 3539 mState, 3540 mMute, 3541 mFillingUpStatus, 3542 mCblk->sampleRate, 3543 vlr & 0xFFFF, 3544 vlr >> 16, 3545 mCblk->server, 3546 mCblk->user, 3547 (int)mMainBuffer, 3548 (int)mAuxBuffer); 3549} 3550 3551status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3552 AudioBufferProvider::Buffer* buffer, int64_t pts) 3553{ 3554 audio_track_cblk_t* cblk = this->cblk(); 3555 uint32_t framesReady; 3556 uint32_t framesReq = buffer->frameCount; 3557 3558 // Check if last stepServer failed, try to step now 3559 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3560 if (!step()) goto getNextBuffer_exit; 3561 ALOGV("stepServer recovered"); 3562 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3563 } 3564 3565 framesReady = cblk->framesReady(); 3566 3567 if (CC_LIKELY(framesReady)) { 3568 uint32_t s = cblk->server; 3569 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3570 3571 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3572 if (framesReq > framesReady) { 3573 framesReq = framesReady; 3574 } 3575 if (s + framesReq > bufferEnd) { 3576 framesReq = bufferEnd - s; 3577 } 3578 3579 buffer->raw = getBuffer(s, framesReq); 3580 if (buffer->raw == NULL) goto getNextBuffer_exit; 3581 3582 buffer->frameCount = framesReq; 3583 return NO_ERROR; 3584 } 3585 3586getNextBuffer_exit: 3587 buffer->raw = NULL; 3588 buffer->frameCount = 0; 3589 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3590 return NOT_ENOUGH_DATA; 3591} 3592 3593uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3594 return mCblk->framesReady(); 3595} 3596 3597bool AudioFlinger::PlaybackThread::Track::isReady() const { 3598 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3599 3600 if (framesReady() >= mCblk->frameCount || 3601 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3602 mFillingUpStatus = FS_FILLED; 3603 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3604 return true; 3605 } 3606 return false; 3607} 3608 3609status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3610{ 3611 status_t status = NO_ERROR; 3612 ALOGV("start(%d), calling pid %d session %d tid %d", 3613 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3614 sp<ThreadBase> thread = mThread.promote(); 3615 if (thread != 0) { 3616 Mutex::Autolock _l(thread->mLock); 3617 track_state state = mState; 3618 // here the track could be either new, or restarted 3619 // in both cases "unstop" the track 3620 if (mState == PAUSED) { 3621 mState = TrackBase::RESUMING; 3622 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3623 } else { 3624 mState = TrackBase::ACTIVE; 3625 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3626 } 3627 3628 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3629 thread->mLock.unlock(); 3630 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3631 thread->mLock.lock(); 3632 3633 // to track the speaker usage 3634 if (status == NO_ERROR) { 3635 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3636 } 3637 } 3638 if (status == NO_ERROR) { 3639 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3640 playbackThread->addTrack_l(this); 3641 } else { 3642 mState = state; 3643 } 3644 } else { 3645 status = BAD_VALUE; 3646 } 3647 return status; 3648} 3649 3650void AudioFlinger::PlaybackThread::Track::stop() 3651{ 3652 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3653 sp<ThreadBase> thread = mThread.promote(); 3654 if (thread != 0) { 3655 Mutex::Autolock _l(thread->mLock); 3656 track_state state = mState; 3657 if (mState > STOPPED) { 3658 mState = STOPPED; 3659 // If the track is not active (PAUSED and buffers full), flush buffers 3660 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3661 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3662 reset(); 3663 } 3664 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3665 } 3666 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3667 thread->mLock.unlock(); 3668 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3669 thread->mLock.lock(); 3670 3671 // to track the speaker usage 3672 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3673 } 3674 } 3675} 3676 3677void AudioFlinger::PlaybackThread::Track::pause() 3678{ 3679 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3680 sp<ThreadBase> thread = mThread.promote(); 3681 if (thread != 0) { 3682 Mutex::Autolock _l(thread->mLock); 3683 if (mState == ACTIVE || mState == RESUMING) { 3684 mState = PAUSING; 3685 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3686 if (!isOutputTrack()) { 3687 thread->mLock.unlock(); 3688 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3689 thread->mLock.lock(); 3690 3691 // to track the speaker usage 3692 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3693 } 3694 } 3695 } 3696} 3697 3698void AudioFlinger::PlaybackThread::Track::flush() 3699{ 3700 ALOGV("flush(%d)", mName); 3701 sp<ThreadBase> thread = mThread.promote(); 3702 if (thread != 0) { 3703 Mutex::Autolock _l(thread->mLock); 3704 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3705 return; 3706 } 3707 // No point remaining in PAUSED state after a flush => go to 3708 // STOPPED state 3709 mState = STOPPED; 3710 3711 // do not reset the track if it is still in the process of being stopped or paused. 3712 // this will be done by prepareTracks_l() when the track is stopped. 3713 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3714 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3715 reset(); 3716 } 3717 } 3718} 3719 3720void AudioFlinger::PlaybackThread::Track::reset() 3721{ 3722 // Do not reset twice to avoid discarding data written just after a flush and before 3723 // the audioflinger thread detects the track is stopped. 3724 if (!mResetDone) { 3725 TrackBase::reset(); 3726 // Force underrun condition to avoid false underrun callback until first data is 3727 // written to buffer 3728 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3729 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3730 mFillingUpStatus = FS_FILLING; 3731 mResetDone = true; 3732 } 3733} 3734 3735void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3736{ 3737 mMute = muted; 3738} 3739 3740status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3741{ 3742 status_t status = DEAD_OBJECT; 3743 sp<ThreadBase> thread = mThread.promote(); 3744 if (thread != 0) { 3745 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3746 status = playbackThread->attachAuxEffect(this, EffectId); 3747 } 3748 return status; 3749} 3750 3751void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3752{ 3753 mAuxEffectId = EffectId; 3754 mAuxBuffer = buffer; 3755} 3756 3757// timed audio tracks 3758 3759sp<AudioFlinger::PlaybackThread::TimedTrack> 3760AudioFlinger::PlaybackThread::TimedTrack::create( 3761 PlaybackThread *thread, 3762 const sp<Client>& client, 3763 audio_stream_type_t streamType, 3764 uint32_t sampleRate, 3765 audio_format_t format, 3766 uint32_t channelMask, 3767 int frameCount, 3768 const sp<IMemory>& sharedBuffer, 3769 int sessionId) { 3770 if (!client->reserveTimedTrack()) 3771 return NULL; 3772 3773 sp<TimedTrack> track = new TimedTrack( 3774 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3775 sharedBuffer, sessionId); 3776 3777 if (track == NULL) { 3778 client->releaseTimedTrack(); 3779 return NULL; 3780 } 3781 3782 return track; 3783} 3784 3785AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3786 PlaybackThread *thread, 3787 const sp<Client>& client, 3788 audio_stream_type_t streamType, 3789 uint32_t sampleRate, 3790 audio_format_t format, 3791 uint32_t channelMask, 3792 int frameCount, 3793 const sp<IMemory>& sharedBuffer, 3794 int sessionId) 3795 : Track(thread, client, streamType, sampleRate, format, channelMask, 3796 frameCount, sharedBuffer, sessionId), 3797 mTimedSilenceBuffer(NULL), 3798 mTimedSilenceBufferSize(0), 3799 mTimedAudioOutputOnTime(false), 3800 mMediaTimeTransformValid(false) 3801{ 3802 LocalClock lc; 3803 mLocalTimeFreq = lc.getLocalFreq(); 3804 3805 mLocalTimeToSampleTransform.a_zero = 0; 3806 mLocalTimeToSampleTransform.b_zero = 0; 3807 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3808 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3809 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3810 &mLocalTimeToSampleTransform.a_to_b_denom); 3811} 3812 3813AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3814 mClient->releaseTimedTrack(); 3815 delete [] mTimedSilenceBuffer; 3816} 3817 3818status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3819 size_t size, sp<IMemory>* buffer) { 3820 3821 Mutex::Autolock _l(mTimedBufferQueueLock); 3822 3823 trimTimedBufferQueue_l(); 3824 3825 // lazily initialize the shared memory heap for timed buffers 3826 if (mTimedMemoryDealer == NULL) { 3827 const int kTimedBufferHeapSize = 512 << 10; 3828 3829 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3830 "AudioFlingerTimed"); 3831 if (mTimedMemoryDealer == NULL) 3832 return NO_MEMORY; 3833 } 3834 3835 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3836 if (newBuffer == NULL) { 3837 newBuffer = mTimedMemoryDealer->allocate(size); 3838 if (newBuffer == NULL) 3839 return NO_MEMORY; 3840 } 3841 3842 *buffer = newBuffer; 3843 return NO_ERROR; 3844} 3845 3846// caller must hold mTimedBufferQueueLock 3847void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3848 int64_t mediaTimeNow; 3849 { 3850 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3851 if (!mMediaTimeTransformValid) 3852 return; 3853 3854 int64_t targetTimeNow; 3855 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3856 ? mCCHelper.getCommonTime(&targetTimeNow) 3857 : mCCHelper.getLocalTime(&targetTimeNow); 3858 3859 if (OK != res) 3860 return; 3861 3862 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3863 &mediaTimeNow)) { 3864 return; 3865 } 3866 } 3867 3868 size_t trimIndex; 3869 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3870 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3871 break; 3872 } 3873 3874 if (trimIndex) { 3875 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3876 } 3877} 3878 3879status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3880 const sp<IMemory>& buffer, int64_t pts) { 3881 3882 { 3883 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3884 if (!mMediaTimeTransformValid) 3885 return INVALID_OPERATION; 3886 } 3887 3888 Mutex::Autolock _l(mTimedBufferQueueLock); 3889 3890 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3891 3892 return NO_ERROR; 3893} 3894 3895status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3896 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3897 3898 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3899 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3900 target); 3901 3902 if (!(target == TimedAudioTrack::LOCAL_TIME || 3903 target == TimedAudioTrack::COMMON_TIME)) { 3904 return BAD_VALUE; 3905 } 3906 3907 Mutex::Autolock lock(mMediaTimeTransformLock); 3908 mMediaTimeTransform = xform; 3909 mMediaTimeTransformTarget = target; 3910 mMediaTimeTransformValid = true; 3911 3912 return NO_ERROR; 3913} 3914 3915#define min(a, b) ((a) < (b) ? (a) : (b)) 3916 3917// implementation of getNextBuffer for tracks whose buffers have timestamps 3918status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3919 AudioBufferProvider::Buffer* buffer, int64_t pts) 3920{ 3921 if (pts == AudioBufferProvider::kInvalidPTS) { 3922 buffer->raw = 0; 3923 buffer->frameCount = 0; 3924 return INVALID_OPERATION; 3925 } 3926 3927 Mutex::Autolock _l(mTimedBufferQueueLock); 3928 3929 while (true) { 3930 3931 // if we have no timed buffers, then fail 3932 if (mTimedBufferQueue.isEmpty()) { 3933 buffer->raw = 0; 3934 buffer->frameCount = 0; 3935 return NOT_ENOUGH_DATA; 3936 } 3937 3938 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3939 3940 // calculate the PTS of the head of the timed buffer queue expressed in 3941 // local time 3942 int64_t headLocalPTS; 3943 { 3944 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3945 3946 assert(mMediaTimeTransformValid); 3947 3948 if (mMediaTimeTransform.a_to_b_denom == 0) { 3949 // the transform represents a pause, so yield silence 3950 timedYieldSilence(buffer->frameCount, buffer); 3951 return NO_ERROR; 3952 } 3953 3954 int64_t transformedPTS; 3955 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3956 &transformedPTS)) { 3957 // the transform failed. this shouldn't happen, but if it does 3958 // then just drop this buffer 3959 ALOGW("timedGetNextBuffer transform failed"); 3960 buffer->raw = 0; 3961 buffer->frameCount = 0; 3962 mTimedBufferQueue.removeAt(0); 3963 return NO_ERROR; 3964 } 3965 3966 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3967 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3968 &headLocalPTS)) { 3969 buffer->raw = 0; 3970 buffer->frameCount = 0; 3971 return INVALID_OPERATION; 3972 } 3973 } else { 3974 headLocalPTS = transformedPTS; 3975 } 3976 } 3977 3978 // adjust the head buffer's PTS to reflect the portion of the head buffer 3979 // that has already been consumed 3980 int64_t effectivePTS = headLocalPTS + 3981 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3982 3983 // Calculate the delta in samples between the head of the input buffer 3984 // queue and the start of the next output buffer that will be written. 3985 // If the transformation fails because of over or underflow, it means 3986 // that the sample's position in the output stream is so far out of 3987 // whack that it should just be dropped. 3988 int64_t sampleDelta; 3989 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3990 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3991 mTimedBufferQueue.removeAt(0); 3992 continue; 3993 } 3994 if (!mLocalTimeToSampleTransform.doForwardTransform( 3995 (effectivePTS - pts) << 32, &sampleDelta)) { 3996 ALOGV("*** too late during sample rate transform: dropped buffer"); 3997 mTimedBufferQueue.removeAt(0); 3998 continue; 3999 } 4000 4001 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4002 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4003 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4004 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4005 4006 // if the delta between the ideal placement for the next input sample and 4007 // the current output position is within this threshold, then we will 4008 // concatenate the next input samples to the previous output 4009 const int64_t kSampleContinuityThreshold = 4010 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4011 4012 // if this is the first buffer of audio that we're emitting from this track 4013 // then it should be almost exactly on time. 4014 const int64_t kSampleStartupThreshold = 1LL << 32; 4015 4016 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4017 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4018 // the next input is close enough to being on time, so concatenate it 4019 // with the last output 4020 timedYieldSamples(buffer); 4021 4022 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4023 return NO_ERROR; 4024 } else if (sampleDelta > 0) { 4025 // the gap between the current output position and the proper start of 4026 // the next input sample is too big, so fill it with silence 4027 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4028 4029 timedYieldSilence(framesUntilNextInput, buffer); 4030 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4031 return NO_ERROR; 4032 } else { 4033 // the next input sample is late 4034 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4035 size_t onTimeSamplePosition = 4036 head.position() + lateFrames * mCblk->frameSize; 4037 4038 if (onTimeSamplePosition > head.buffer()->size()) { 4039 // all the remaining samples in the head are too late, so 4040 // drop it and move on 4041 ALOGV("*** too late: dropped buffer"); 4042 mTimedBufferQueue.removeAt(0); 4043 continue; 4044 } else { 4045 // skip over the late samples 4046 head.setPosition(onTimeSamplePosition); 4047 4048 // yield the available samples 4049 timedYieldSamples(buffer); 4050 4051 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4052 return NO_ERROR; 4053 } 4054 } 4055 } 4056} 4057 4058// Yield samples from the timed buffer queue head up to the given output 4059// buffer's capacity. 4060// 4061// Caller must hold mTimedBufferQueueLock 4062void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4063 AudioBufferProvider::Buffer* buffer) { 4064 4065 const TimedBuffer& head = mTimedBufferQueue[0]; 4066 4067 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4068 head.position()); 4069 4070 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4071 mCblk->frameSize); 4072 size_t framesRequested = buffer->frameCount; 4073 buffer->frameCount = min(framesLeftInHead, framesRequested); 4074 4075 mTimedAudioOutputOnTime = true; 4076} 4077 4078// Yield samples of silence up to the given output buffer's capacity 4079// 4080// Caller must hold mTimedBufferQueueLock 4081void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4082 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4083 4084 // lazily allocate a buffer filled with silence 4085 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4086 delete [] mTimedSilenceBuffer; 4087 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4088 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4089 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4090 } 4091 4092 buffer->raw = mTimedSilenceBuffer; 4093 size_t framesRequested = buffer->frameCount; 4094 buffer->frameCount = min(numFrames, framesRequested); 4095 4096 mTimedAudioOutputOnTime = false; 4097} 4098 4099void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4100 AudioBufferProvider::Buffer* buffer) { 4101 4102 Mutex::Autolock _l(mTimedBufferQueueLock); 4103 4104 // If the buffer which was just released is part of the buffer at the head 4105 // of the queue, be sure to update the amt of the buffer which has been 4106 // consumed. If the buffer being returned is not part of the head of the 4107 // queue, its either because the buffer is part of the silence buffer, or 4108 // because the head of the timed queue was trimmed after the mixer called 4109 // getNextBuffer but before the mixer called releaseBuffer. 4110 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4111 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4112 4113 void* start = head.buffer()->pointer(); 4114 void* end = head.buffer()->pointer() + head.buffer()->size(); 4115 4116 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4117 head.setPosition(head.position() + 4118 (buffer->frameCount * mCblk->frameSize)); 4119 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4120 mTimedBufferQueue.removeAt(0); 4121 } 4122 } 4123 } 4124 4125 buffer->raw = 0; 4126 buffer->frameCount = 0; 4127} 4128 4129uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4130 Mutex::Autolock _l(mTimedBufferQueueLock); 4131 4132 uint32_t frames = 0; 4133 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4134 const TimedBuffer& tb = mTimedBufferQueue[i]; 4135 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4136 } 4137 4138 return frames; 4139} 4140 4141AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4142 : mPTS(0), mPosition(0) {} 4143 4144AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4145 const sp<IMemory>& buffer, int64_t pts) 4146 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4147 4148// ---------------------------------------------------------------------------- 4149 4150// RecordTrack constructor must be called with AudioFlinger::mLock held 4151AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4152 RecordThread *thread, 4153 const sp<Client>& client, 4154 uint32_t sampleRate, 4155 audio_format_t format, 4156 uint32_t channelMask, 4157 int frameCount, 4158 uint32_t flags, 4159 int sessionId) 4160 : TrackBase(thread, client, sampleRate, format, 4161 channelMask, frameCount, flags, 0, sessionId), 4162 mOverflow(false) 4163{ 4164 if (mCblk != NULL) { 4165 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4166 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4167 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4168 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4169 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4170 } else { 4171 mCblk->frameSize = sizeof(int8_t); 4172 } 4173 } 4174} 4175 4176AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4177{ 4178 sp<ThreadBase> thread = mThread.promote(); 4179 if (thread != 0) { 4180 AudioSystem::releaseInput(thread->id()); 4181 } 4182} 4183 4184status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4185{ 4186 audio_track_cblk_t* cblk = this->cblk(); 4187 uint32_t framesAvail; 4188 uint32_t framesReq = buffer->frameCount; 4189 4190 // Check if last stepServer failed, try to step now 4191 if (mFlags & TrackBase::STEPSERVER_FAILED) { 4192 if (!step()) goto getNextBuffer_exit; 4193 ALOGV("stepServer recovered"); 4194 mFlags &= ~TrackBase::STEPSERVER_FAILED; 4195 } 4196 4197 framesAvail = cblk->framesAvailable_l(); 4198 4199 if (CC_LIKELY(framesAvail)) { 4200 uint32_t s = cblk->server; 4201 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4202 4203 if (framesReq > framesAvail) { 4204 framesReq = framesAvail; 4205 } 4206 if (s + framesReq > bufferEnd) { 4207 framesReq = bufferEnd - s; 4208 } 4209 4210 buffer->raw = getBuffer(s, framesReq); 4211 if (buffer->raw == NULL) goto getNextBuffer_exit; 4212 4213 buffer->frameCount = framesReq; 4214 return NO_ERROR; 4215 } 4216 4217getNextBuffer_exit: 4218 buffer->raw = NULL; 4219 buffer->frameCount = 0; 4220 return NOT_ENOUGH_DATA; 4221} 4222 4223status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4224{ 4225 sp<ThreadBase> thread = mThread.promote(); 4226 if (thread != 0) { 4227 RecordThread *recordThread = (RecordThread *)thread.get(); 4228 return recordThread->start(this, tid); 4229 } else { 4230 return BAD_VALUE; 4231 } 4232} 4233 4234void AudioFlinger::RecordThread::RecordTrack::stop() 4235{ 4236 sp<ThreadBase> thread = mThread.promote(); 4237 if (thread != 0) { 4238 RecordThread *recordThread = (RecordThread *)thread.get(); 4239 recordThread->stop(this); 4240 TrackBase::reset(); 4241 // Force overerrun condition to avoid false overrun callback until first data is 4242 // read from buffer 4243 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4244 } 4245} 4246 4247void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4248{ 4249 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4250 (mClient == 0) ? getpid_cached : mClient->pid(), 4251 mFormat, 4252 mChannelMask, 4253 mSessionId, 4254 mFrameCount, 4255 mState, 4256 mCblk->sampleRate, 4257 mCblk->server, 4258 mCblk->user); 4259} 4260 4261 4262// ---------------------------------------------------------------------------- 4263 4264AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4265 PlaybackThread *playbackThread, 4266 DuplicatingThread *sourceThread, 4267 uint32_t sampleRate, 4268 audio_format_t format, 4269 uint32_t channelMask, 4270 int frameCount) 4271 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4272 mActive(false), mSourceThread(sourceThread) 4273{ 4274 4275 if (mCblk != NULL) { 4276 mCblk->flags |= CBLK_DIRECTION_OUT; 4277 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4278 mOutBuffer.frameCount = 0; 4279 playbackThread->mTracks.add(this); 4280 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4281 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4282 mCblk, mBuffer, mCblk->buffers, 4283 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4284 } else { 4285 ALOGW("Error creating output track on thread %p", playbackThread); 4286 } 4287} 4288 4289AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4290{ 4291 clearBufferQueue(); 4292} 4293 4294status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4295{ 4296 status_t status = Track::start(tid); 4297 if (status != NO_ERROR) { 4298 return status; 4299 } 4300 4301 mActive = true; 4302 mRetryCount = 127; 4303 return status; 4304} 4305 4306void AudioFlinger::PlaybackThread::OutputTrack::stop() 4307{ 4308 Track::stop(); 4309 clearBufferQueue(); 4310 mOutBuffer.frameCount = 0; 4311 mActive = false; 4312} 4313 4314bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4315{ 4316 Buffer *pInBuffer; 4317 Buffer inBuffer; 4318 uint32_t channelCount = mChannelCount; 4319 bool outputBufferFull = false; 4320 inBuffer.frameCount = frames; 4321 inBuffer.i16 = data; 4322 4323 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4324 4325 if (!mActive && frames != 0) { 4326 start(0); 4327 sp<ThreadBase> thread = mThread.promote(); 4328 if (thread != 0) { 4329 MixerThread *mixerThread = (MixerThread *)thread.get(); 4330 if (mCblk->frameCount > frames){ 4331 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4332 uint32_t startFrames = (mCblk->frameCount - frames); 4333 pInBuffer = new Buffer; 4334 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4335 pInBuffer->frameCount = startFrames; 4336 pInBuffer->i16 = pInBuffer->mBuffer; 4337 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4338 mBufferQueue.add(pInBuffer); 4339 } else { 4340 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4341 } 4342 } 4343 } 4344 } 4345 4346 while (waitTimeLeftMs) { 4347 // First write pending buffers, then new data 4348 if (mBufferQueue.size()) { 4349 pInBuffer = mBufferQueue.itemAt(0); 4350 } else { 4351 pInBuffer = &inBuffer; 4352 } 4353 4354 if (pInBuffer->frameCount == 0) { 4355 break; 4356 } 4357 4358 if (mOutBuffer.frameCount == 0) { 4359 mOutBuffer.frameCount = pInBuffer->frameCount; 4360 nsecs_t startTime = systemTime(); 4361 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4362 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4363 outputBufferFull = true; 4364 break; 4365 } 4366 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4367 if (waitTimeLeftMs >= waitTimeMs) { 4368 waitTimeLeftMs -= waitTimeMs; 4369 } else { 4370 waitTimeLeftMs = 0; 4371 } 4372 } 4373 4374 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4375 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4376 mCblk->stepUser(outFrames); 4377 pInBuffer->frameCount -= outFrames; 4378 pInBuffer->i16 += outFrames * channelCount; 4379 mOutBuffer.frameCount -= outFrames; 4380 mOutBuffer.i16 += outFrames * channelCount; 4381 4382 if (pInBuffer->frameCount == 0) { 4383 if (mBufferQueue.size()) { 4384 mBufferQueue.removeAt(0); 4385 delete [] pInBuffer->mBuffer; 4386 delete pInBuffer; 4387 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4388 } else { 4389 break; 4390 } 4391 } 4392 } 4393 4394 // If we could not write all frames, allocate a buffer and queue it for next time. 4395 if (inBuffer.frameCount) { 4396 sp<ThreadBase> thread = mThread.promote(); 4397 if (thread != 0 && !thread->standby()) { 4398 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4399 pInBuffer = new Buffer; 4400 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4401 pInBuffer->frameCount = inBuffer.frameCount; 4402 pInBuffer->i16 = pInBuffer->mBuffer; 4403 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4404 mBufferQueue.add(pInBuffer); 4405 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4406 } else { 4407 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4408 } 4409 } 4410 } 4411 4412 // Calling write() with a 0 length buffer, means that no more data will be written: 4413 // If no more buffers are pending, fill output track buffer to make sure it is started 4414 // by output mixer. 4415 if (frames == 0 && mBufferQueue.size() == 0) { 4416 if (mCblk->user < mCblk->frameCount) { 4417 frames = mCblk->frameCount - mCblk->user; 4418 pInBuffer = new Buffer; 4419 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4420 pInBuffer->frameCount = frames; 4421 pInBuffer->i16 = pInBuffer->mBuffer; 4422 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4423 mBufferQueue.add(pInBuffer); 4424 } else if (mActive) { 4425 stop(); 4426 } 4427 } 4428 4429 return outputBufferFull; 4430} 4431 4432status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4433{ 4434 int active; 4435 status_t result; 4436 audio_track_cblk_t* cblk = mCblk; 4437 uint32_t framesReq = buffer->frameCount; 4438 4439// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4440 buffer->frameCount = 0; 4441 4442 uint32_t framesAvail = cblk->framesAvailable(); 4443 4444 4445 if (framesAvail == 0) { 4446 Mutex::Autolock _l(cblk->lock); 4447 goto start_loop_here; 4448 while (framesAvail == 0) { 4449 active = mActive; 4450 if (CC_UNLIKELY(!active)) { 4451 ALOGV("Not active and NO_MORE_BUFFERS"); 4452 return NO_MORE_BUFFERS; 4453 } 4454 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4455 if (result != NO_ERROR) { 4456 return NO_MORE_BUFFERS; 4457 } 4458 // read the server count again 4459 start_loop_here: 4460 framesAvail = cblk->framesAvailable_l(); 4461 } 4462 } 4463 4464// if (framesAvail < framesReq) { 4465// return NO_MORE_BUFFERS; 4466// } 4467 4468 if (framesReq > framesAvail) { 4469 framesReq = framesAvail; 4470 } 4471 4472 uint32_t u = cblk->user; 4473 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4474 4475 if (u + framesReq > bufferEnd) { 4476 framesReq = bufferEnd - u; 4477 } 4478 4479 buffer->frameCount = framesReq; 4480 buffer->raw = (void *)cblk->buffer(u); 4481 return NO_ERROR; 4482} 4483 4484 4485void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4486{ 4487 size_t size = mBufferQueue.size(); 4488 4489 for (size_t i = 0; i < size; i++) { 4490 Buffer *pBuffer = mBufferQueue.itemAt(i); 4491 delete [] pBuffer->mBuffer; 4492 delete pBuffer; 4493 } 4494 mBufferQueue.clear(); 4495} 4496 4497// ---------------------------------------------------------------------------- 4498 4499AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4500 : RefBase(), 4501 mAudioFlinger(audioFlinger), 4502 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4503 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4504 mPid(pid), 4505 mTimedTrackCount(0) 4506{ 4507 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4508} 4509 4510// Client destructor must be called with AudioFlinger::mLock held 4511AudioFlinger::Client::~Client() 4512{ 4513 mAudioFlinger->removeClient_l(mPid); 4514} 4515 4516sp<MemoryDealer> AudioFlinger::Client::heap() const 4517{ 4518 return mMemoryDealer; 4519} 4520 4521// Reserve one of the limited slots for a timed audio track associated 4522// with this client 4523bool AudioFlinger::Client::reserveTimedTrack() 4524{ 4525 const int kMaxTimedTracksPerClient = 4; 4526 4527 Mutex::Autolock _l(mTimedTrackLock); 4528 4529 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4530 ALOGW("can not create timed track - pid %d has exceeded the limit", 4531 mPid); 4532 return false; 4533 } 4534 4535 mTimedTrackCount++; 4536 return true; 4537} 4538 4539// Release a slot for a timed audio track 4540void AudioFlinger::Client::releaseTimedTrack() 4541{ 4542 Mutex::Autolock _l(mTimedTrackLock); 4543 mTimedTrackCount--; 4544} 4545 4546// ---------------------------------------------------------------------------- 4547 4548AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4549 const sp<IAudioFlingerClient>& client, 4550 pid_t pid) 4551 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4552{ 4553} 4554 4555AudioFlinger::NotificationClient::~NotificationClient() 4556{ 4557} 4558 4559void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4560{ 4561 sp<NotificationClient> keep(this); 4562 mAudioFlinger->removeNotificationClient(mPid); 4563} 4564 4565// ---------------------------------------------------------------------------- 4566 4567AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4568 : BnAudioTrack(), 4569 mTrack(track) 4570{ 4571} 4572 4573AudioFlinger::TrackHandle::~TrackHandle() { 4574 // just stop the track on deletion, associated resources 4575 // will be freed from the main thread once all pending buffers have 4576 // been played. Unless it's not in the active track list, in which 4577 // case we free everything now... 4578 mTrack->destroy(); 4579} 4580 4581sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4582 return mTrack->getCblk(); 4583} 4584 4585status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4586 return mTrack->start(tid); 4587} 4588 4589void AudioFlinger::TrackHandle::stop() { 4590 mTrack->stop(); 4591} 4592 4593void AudioFlinger::TrackHandle::flush() { 4594 mTrack->flush(); 4595} 4596 4597void AudioFlinger::TrackHandle::mute(bool e) { 4598 mTrack->mute(e); 4599} 4600 4601void AudioFlinger::TrackHandle::pause() { 4602 mTrack->pause(); 4603} 4604 4605status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4606{ 4607 return mTrack->attachAuxEffect(EffectId); 4608} 4609 4610status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4611 sp<IMemory>* buffer) { 4612 if (!mTrack->isTimedTrack()) 4613 return INVALID_OPERATION; 4614 4615 PlaybackThread::TimedTrack* tt = 4616 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4617 return tt->allocateTimedBuffer(size, buffer); 4618} 4619 4620status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4621 int64_t pts) { 4622 if (!mTrack->isTimedTrack()) 4623 return INVALID_OPERATION; 4624 4625 PlaybackThread::TimedTrack* tt = 4626 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4627 return tt->queueTimedBuffer(buffer, pts); 4628} 4629 4630status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4631 const LinearTransform& xform, int target) { 4632 4633 if (!mTrack->isTimedTrack()) 4634 return INVALID_OPERATION; 4635 4636 PlaybackThread::TimedTrack* tt = 4637 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4638 return tt->setMediaTimeTransform( 4639 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4640} 4641 4642status_t AudioFlinger::TrackHandle::onTransact( 4643 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4644{ 4645 return BnAudioTrack::onTransact(code, data, reply, flags); 4646} 4647 4648// ---------------------------------------------------------------------------- 4649 4650sp<IAudioRecord> AudioFlinger::openRecord( 4651 pid_t pid, 4652 audio_io_handle_t input, 4653 uint32_t sampleRate, 4654 audio_format_t format, 4655 uint32_t channelMask, 4656 int frameCount, 4657 uint32_t flags, 4658 int *sessionId, 4659 status_t *status) 4660{ 4661 sp<RecordThread::RecordTrack> recordTrack; 4662 sp<RecordHandle> recordHandle; 4663 sp<Client> client; 4664 status_t lStatus; 4665 RecordThread *thread; 4666 size_t inFrameCount; 4667 int lSessionId; 4668 4669 // check calling permissions 4670 if (!recordingAllowed()) { 4671 lStatus = PERMISSION_DENIED; 4672 goto Exit; 4673 } 4674 4675 // add client to list 4676 { // scope for mLock 4677 Mutex::Autolock _l(mLock); 4678 thread = checkRecordThread_l(input); 4679 if (thread == NULL) { 4680 lStatus = BAD_VALUE; 4681 goto Exit; 4682 } 4683 4684 client = registerPid_l(pid); 4685 4686 // If no audio session id is provided, create one here 4687 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4688 lSessionId = *sessionId; 4689 } else { 4690 lSessionId = nextUniqueId(); 4691 if (sessionId != NULL) { 4692 *sessionId = lSessionId; 4693 } 4694 } 4695 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4696 recordTrack = thread->createRecordTrack_l(client, 4697 sampleRate, 4698 format, 4699 channelMask, 4700 frameCount, 4701 flags, 4702 lSessionId, 4703 &lStatus); 4704 } 4705 if (lStatus != NO_ERROR) { 4706 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4707 // destructor is called by the TrackBase destructor with mLock held 4708 client.clear(); 4709 recordTrack.clear(); 4710 goto Exit; 4711 } 4712 4713 // return to handle to client 4714 recordHandle = new RecordHandle(recordTrack); 4715 lStatus = NO_ERROR; 4716 4717Exit: 4718 if (status) { 4719 *status = lStatus; 4720 } 4721 return recordHandle; 4722} 4723 4724// ---------------------------------------------------------------------------- 4725 4726AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4727 : BnAudioRecord(), 4728 mRecordTrack(recordTrack) 4729{ 4730} 4731 4732AudioFlinger::RecordHandle::~RecordHandle() { 4733 stop(); 4734} 4735 4736sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4737 return mRecordTrack->getCblk(); 4738} 4739 4740status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4741 ALOGV("RecordHandle::start()"); 4742 return mRecordTrack->start(tid); 4743} 4744 4745void AudioFlinger::RecordHandle::stop() { 4746 ALOGV("RecordHandle::stop()"); 4747 mRecordTrack->stop(); 4748} 4749 4750status_t AudioFlinger::RecordHandle::onTransact( 4751 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4752{ 4753 return BnAudioRecord::onTransact(code, data, reply, flags); 4754} 4755 4756// ---------------------------------------------------------------------------- 4757 4758AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4759 AudioStreamIn *input, 4760 uint32_t sampleRate, 4761 uint32_t channels, 4762 audio_io_handle_t id, 4763 uint32_t device) : 4764 ThreadBase(audioFlinger, id, device, RECORD), 4765 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4766 // mRsmpInIndex and mInputBytes set by readInputParameters() 4767 mReqChannelCount(popcount(channels)), 4768 mReqSampleRate(sampleRate) 4769 // mBytesRead is only meaningful while active, and so is cleared in start() 4770 // (but might be better to also clear here for dump?) 4771{ 4772 snprintf(mName, kNameLength, "AudioIn_%d", id); 4773 4774 readInputParameters(); 4775} 4776 4777 4778AudioFlinger::RecordThread::~RecordThread() 4779{ 4780 delete[] mRsmpInBuffer; 4781 delete mResampler; 4782 delete[] mRsmpOutBuffer; 4783} 4784 4785void AudioFlinger::RecordThread::onFirstRef() 4786{ 4787 run(mName, PRIORITY_URGENT_AUDIO); 4788} 4789 4790status_t AudioFlinger::RecordThread::readyToRun() 4791{ 4792 status_t status = initCheck(); 4793 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4794 return status; 4795} 4796 4797bool AudioFlinger::RecordThread::threadLoop() 4798{ 4799 AudioBufferProvider::Buffer buffer; 4800 sp<RecordTrack> activeTrack; 4801 Vector< sp<EffectChain> > effectChains; 4802 4803 nsecs_t lastWarning = 0; 4804 4805 acquireWakeLock(); 4806 4807 // start recording 4808 while (!exitPending()) { 4809 4810 processConfigEvents(); 4811 4812 { // scope for mLock 4813 Mutex::Autolock _l(mLock); 4814 checkForNewParameters_l(); 4815 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4816 if (!mStandby) { 4817 mInput->stream->common.standby(&mInput->stream->common); 4818 mStandby = true; 4819 } 4820 4821 if (exitPending()) break; 4822 4823 releaseWakeLock_l(); 4824 ALOGV("RecordThread: loop stopping"); 4825 // go to sleep 4826 mWaitWorkCV.wait(mLock); 4827 ALOGV("RecordThread: loop starting"); 4828 acquireWakeLock_l(); 4829 continue; 4830 } 4831 if (mActiveTrack != 0) { 4832 if (mActiveTrack->mState == TrackBase::PAUSING) { 4833 if (!mStandby) { 4834 mInput->stream->common.standby(&mInput->stream->common); 4835 mStandby = true; 4836 } 4837 mActiveTrack.clear(); 4838 mStartStopCond.broadcast(); 4839 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4840 if (mReqChannelCount != mActiveTrack->channelCount()) { 4841 mActiveTrack.clear(); 4842 mStartStopCond.broadcast(); 4843 } else if (mBytesRead != 0) { 4844 // record start succeeds only if first read from audio input 4845 // succeeds 4846 if (mBytesRead > 0) { 4847 mActiveTrack->mState = TrackBase::ACTIVE; 4848 } else { 4849 mActiveTrack.clear(); 4850 } 4851 mStartStopCond.broadcast(); 4852 } 4853 mStandby = false; 4854 } 4855 } 4856 lockEffectChains_l(effectChains); 4857 } 4858 4859 if (mActiveTrack != 0) { 4860 if (mActiveTrack->mState != TrackBase::ACTIVE && 4861 mActiveTrack->mState != TrackBase::RESUMING) { 4862 unlockEffectChains(effectChains); 4863 usleep(kRecordThreadSleepUs); 4864 continue; 4865 } 4866 for (size_t i = 0; i < effectChains.size(); i ++) { 4867 effectChains[i]->process_l(); 4868 } 4869 4870 buffer.frameCount = mFrameCount; 4871 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4872 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4873 size_t framesOut = buffer.frameCount; 4874 if (mResampler == NULL) { 4875 // no resampling 4876 while (framesOut) { 4877 size_t framesIn = mFrameCount - mRsmpInIndex; 4878 if (framesIn) { 4879 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4880 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4881 if (framesIn > framesOut) 4882 framesIn = framesOut; 4883 mRsmpInIndex += framesIn; 4884 framesOut -= framesIn; 4885 if ((int)mChannelCount == mReqChannelCount || 4886 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4887 memcpy(dst, src, framesIn * mFrameSize); 4888 } else { 4889 int16_t *src16 = (int16_t *)src; 4890 int16_t *dst16 = (int16_t *)dst; 4891 if (mChannelCount == 1) { 4892 while (framesIn--) { 4893 *dst16++ = *src16; 4894 *dst16++ = *src16++; 4895 } 4896 } else { 4897 while (framesIn--) { 4898 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4899 src16 += 2; 4900 } 4901 } 4902 } 4903 } 4904 if (framesOut && mFrameCount == mRsmpInIndex) { 4905 if (framesOut == mFrameCount && 4906 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4907 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4908 framesOut = 0; 4909 } else { 4910 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4911 mRsmpInIndex = 0; 4912 } 4913 if (mBytesRead < 0) { 4914 ALOGE("Error reading audio input"); 4915 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4916 // Force input into standby so that it tries to 4917 // recover at next read attempt 4918 mInput->stream->common.standby(&mInput->stream->common); 4919 usleep(kRecordThreadSleepUs); 4920 } 4921 mRsmpInIndex = mFrameCount; 4922 framesOut = 0; 4923 buffer.frameCount = 0; 4924 } 4925 } 4926 } 4927 } else { 4928 // resampling 4929 4930 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4931 // alter output frame count as if we were expecting stereo samples 4932 if (mChannelCount == 1 && mReqChannelCount == 1) { 4933 framesOut >>= 1; 4934 } 4935 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4936 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4937 // are 32 bit aligned which should be always true. 4938 if (mChannelCount == 2 && mReqChannelCount == 1) { 4939 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4940 // the resampler always outputs stereo samples: do post stereo to mono conversion 4941 int16_t *src = (int16_t *)mRsmpOutBuffer; 4942 int16_t *dst = buffer.i16; 4943 while (framesOut--) { 4944 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4945 src += 2; 4946 } 4947 } else { 4948 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4949 } 4950 4951 } 4952 mActiveTrack->releaseBuffer(&buffer); 4953 mActiveTrack->overflow(); 4954 } 4955 // client isn't retrieving buffers fast enough 4956 else { 4957 if (!mActiveTrack->setOverflow()) { 4958 nsecs_t now = systemTime(); 4959 if ((now - lastWarning) > kWarningThrottleNs) { 4960 ALOGW("RecordThread: buffer overflow"); 4961 lastWarning = now; 4962 } 4963 } 4964 // Release the processor for a while before asking for a new buffer. 4965 // This will give the application more chance to read from the buffer and 4966 // clear the overflow. 4967 usleep(kRecordThreadSleepUs); 4968 } 4969 } 4970 // enable changes in effect chain 4971 unlockEffectChains(effectChains); 4972 effectChains.clear(); 4973 } 4974 4975 if (!mStandby) { 4976 mInput->stream->common.standby(&mInput->stream->common); 4977 } 4978 mActiveTrack.clear(); 4979 4980 mStartStopCond.broadcast(); 4981 4982 releaseWakeLock(); 4983 4984 ALOGV("RecordThread %p exiting", this); 4985 return false; 4986} 4987 4988 4989sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4990 const sp<AudioFlinger::Client>& client, 4991 uint32_t sampleRate, 4992 audio_format_t format, 4993 int channelMask, 4994 int frameCount, 4995 uint32_t flags, 4996 int sessionId, 4997 status_t *status) 4998{ 4999 sp<RecordTrack> track; 5000 status_t lStatus; 5001 5002 lStatus = initCheck(); 5003 if (lStatus != NO_ERROR) { 5004 ALOGE("Audio driver not initialized."); 5005 goto Exit; 5006 } 5007 5008 { // scope for mLock 5009 Mutex::Autolock _l(mLock); 5010 5011 track = new RecordTrack(this, client, sampleRate, 5012 format, channelMask, frameCount, flags, sessionId); 5013 5014 if (track->getCblk() == 0) { 5015 lStatus = NO_MEMORY; 5016 goto Exit; 5017 } 5018 5019 mTrack = track.get(); 5020 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5021 bool suspend = audio_is_bluetooth_sco_device( 5022 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5023 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5024 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5025 } 5026 lStatus = NO_ERROR; 5027 5028Exit: 5029 if (status) { 5030 *status = lStatus; 5031 } 5032 return track; 5033} 5034 5035status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5036{ 5037 ALOGV("RecordThread::start tid=%d", tid); 5038 sp <ThreadBase> strongMe = this; 5039 status_t status = NO_ERROR; 5040 { 5041 AutoMutex lock(mLock); 5042 if (mActiveTrack != 0) { 5043 if (recordTrack != mActiveTrack.get()) { 5044 status = -EBUSY; 5045 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5046 mActiveTrack->mState = TrackBase::ACTIVE; 5047 } 5048 return status; 5049 } 5050 5051 recordTrack->mState = TrackBase::IDLE; 5052 mActiveTrack = recordTrack; 5053 mLock.unlock(); 5054 status_t status = AudioSystem::startInput(mId); 5055 mLock.lock(); 5056 if (status != NO_ERROR) { 5057 mActiveTrack.clear(); 5058 return status; 5059 } 5060 mRsmpInIndex = mFrameCount; 5061 mBytesRead = 0; 5062 if (mResampler != NULL) { 5063 mResampler->reset(); 5064 } 5065 mActiveTrack->mState = TrackBase::RESUMING; 5066 // signal thread to start 5067 ALOGV("Signal record thread"); 5068 mWaitWorkCV.signal(); 5069 // do not wait for mStartStopCond if exiting 5070 if (exitPending()) { 5071 mActiveTrack.clear(); 5072 status = INVALID_OPERATION; 5073 goto startError; 5074 } 5075 mStartStopCond.wait(mLock); 5076 if (mActiveTrack == 0) { 5077 ALOGV("Record failed to start"); 5078 status = BAD_VALUE; 5079 goto startError; 5080 } 5081 ALOGV("Record started OK"); 5082 return status; 5083 } 5084startError: 5085 AudioSystem::stopInput(mId); 5086 return status; 5087} 5088 5089void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5090 ALOGV("RecordThread::stop"); 5091 sp <ThreadBase> strongMe = this; 5092 { 5093 AutoMutex lock(mLock); 5094 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5095 mActiveTrack->mState = TrackBase::PAUSING; 5096 // do not wait for mStartStopCond if exiting 5097 if (exitPending()) { 5098 return; 5099 } 5100 mStartStopCond.wait(mLock); 5101 // if we have been restarted, recordTrack == mActiveTrack.get() here 5102 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5103 mLock.unlock(); 5104 AudioSystem::stopInput(mId); 5105 mLock.lock(); 5106 ALOGV("Record stopped OK"); 5107 } 5108 } 5109 } 5110} 5111 5112status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5113{ 5114 const size_t SIZE = 256; 5115 char buffer[SIZE]; 5116 String8 result; 5117 5118 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5119 result.append(buffer); 5120 5121 if (mActiveTrack != 0) { 5122 result.append("Active Track:\n"); 5123 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5124 mActiveTrack->dump(buffer, SIZE); 5125 result.append(buffer); 5126 5127 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5128 result.append(buffer); 5129 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5130 result.append(buffer); 5131 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5132 result.append(buffer); 5133 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5134 result.append(buffer); 5135 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5136 result.append(buffer); 5137 5138 5139 } else { 5140 result.append("No record client\n"); 5141 } 5142 write(fd, result.string(), result.size()); 5143 5144 dumpBase(fd, args); 5145 dumpEffectChains(fd, args); 5146 5147 return NO_ERROR; 5148} 5149 5150status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5151{ 5152 size_t framesReq = buffer->frameCount; 5153 size_t framesReady = mFrameCount - mRsmpInIndex; 5154 int channelCount; 5155 5156 if (framesReady == 0) { 5157 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5158 if (mBytesRead < 0) { 5159 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5160 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5161 // Force input into standby so that it tries to 5162 // recover at next read attempt 5163 mInput->stream->common.standby(&mInput->stream->common); 5164 usleep(kRecordThreadSleepUs); 5165 } 5166 buffer->raw = NULL; 5167 buffer->frameCount = 0; 5168 return NOT_ENOUGH_DATA; 5169 } 5170 mRsmpInIndex = 0; 5171 framesReady = mFrameCount; 5172 } 5173 5174 if (framesReq > framesReady) { 5175 framesReq = framesReady; 5176 } 5177 5178 if (mChannelCount == 1 && mReqChannelCount == 2) { 5179 channelCount = 1; 5180 } else { 5181 channelCount = 2; 5182 } 5183 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5184 buffer->frameCount = framesReq; 5185 return NO_ERROR; 5186} 5187 5188void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5189{ 5190 mRsmpInIndex += buffer->frameCount; 5191 buffer->frameCount = 0; 5192} 5193 5194bool AudioFlinger::RecordThread::checkForNewParameters_l() 5195{ 5196 bool reconfig = false; 5197 5198 while (!mNewParameters.isEmpty()) { 5199 status_t status = NO_ERROR; 5200 String8 keyValuePair = mNewParameters[0]; 5201 AudioParameter param = AudioParameter(keyValuePair); 5202 int value; 5203 audio_format_t reqFormat = mFormat; 5204 int reqSamplingRate = mReqSampleRate; 5205 int reqChannelCount = mReqChannelCount; 5206 5207 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5208 reqSamplingRate = value; 5209 reconfig = true; 5210 } 5211 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5212 reqFormat = (audio_format_t) value; 5213 reconfig = true; 5214 } 5215 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5216 reqChannelCount = popcount(value); 5217 reconfig = true; 5218 } 5219 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5220 // do not accept frame count changes if tracks are open as the track buffer 5221 // size depends on frame count and correct behavior would not be guaranteed 5222 // if frame count is changed after track creation 5223 if (mActiveTrack != 0) { 5224 status = INVALID_OPERATION; 5225 } else { 5226 reconfig = true; 5227 } 5228 } 5229 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5230 // forward device change to effects that have requested to be 5231 // aware of attached audio device. 5232 for (size_t i = 0; i < mEffectChains.size(); i++) { 5233 mEffectChains[i]->setDevice_l(value); 5234 } 5235 // store input device and output device but do not forward output device to audio HAL. 5236 // Note that status is ignored by the caller for output device 5237 // (see AudioFlinger::setParameters() 5238 if (value & AUDIO_DEVICE_OUT_ALL) { 5239 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5240 status = BAD_VALUE; 5241 } else { 5242 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5243 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5244 if (mTrack != NULL) { 5245 bool suspend = audio_is_bluetooth_sco_device( 5246 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5247 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5248 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5249 } 5250 } 5251 mDevice |= (uint32_t)value; 5252 } 5253 if (status == NO_ERROR) { 5254 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5255 if (status == INVALID_OPERATION) { 5256 mInput->stream->common.standby(&mInput->stream->common); 5257 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5258 } 5259 if (reconfig) { 5260 if (status == BAD_VALUE && 5261 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5262 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5263 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5264 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5265 (reqChannelCount < 3)) { 5266 status = NO_ERROR; 5267 } 5268 if (status == NO_ERROR) { 5269 readInputParameters(); 5270 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5271 } 5272 } 5273 } 5274 5275 mNewParameters.removeAt(0); 5276 5277 mParamStatus = status; 5278 mParamCond.signal(); 5279 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5280 // already timed out waiting for the status and will never signal the condition. 5281 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5282 } 5283 return reconfig; 5284} 5285 5286String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5287{ 5288 char *s; 5289 String8 out_s8 = String8(); 5290 5291 Mutex::Autolock _l(mLock); 5292 if (initCheck() != NO_ERROR) { 5293 return out_s8; 5294 } 5295 5296 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5297 out_s8 = String8(s); 5298 free(s); 5299 return out_s8; 5300} 5301 5302void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5303 AudioSystem::OutputDescriptor desc; 5304 void *param2 = NULL; 5305 5306 switch (event) { 5307 case AudioSystem::INPUT_OPENED: 5308 case AudioSystem::INPUT_CONFIG_CHANGED: 5309 desc.channels = mChannelMask; 5310 desc.samplingRate = mSampleRate; 5311 desc.format = mFormat; 5312 desc.frameCount = mFrameCount; 5313 desc.latency = 0; 5314 param2 = &desc; 5315 break; 5316 5317 case AudioSystem::INPUT_CLOSED: 5318 default: 5319 break; 5320 } 5321 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5322} 5323 5324void AudioFlinger::RecordThread::readInputParameters() 5325{ 5326 delete mRsmpInBuffer; 5327 // mRsmpInBuffer is always assigned a new[] below 5328 delete mRsmpOutBuffer; 5329 mRsmpOutBuffer = NULL; 5330 delete mResampler; 5331 mResampler = NULL; 5332 5333 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5334 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5335 mChannelCount = (uint16_t)popcount(mChannelMask); 5336 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5337 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5338 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5339 mFrameCount = mInputBytes / mFrameSize; 5340 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5341 5342 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5343 { 5344 int channelCount; 5345 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5346 // stereo to mono post process as the resampler always outputs stereo. 5347 if (mChannelCount == 1 && mReqChannelCount == 2) { 5348 channelCount = 1; 5349 } else { 5350 channelCount = 2; 5351 } 5352 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5353 mResampler->setSampleRate(mSampleRate); 5354 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5355 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5356 5357 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5358 if (mChannelCount == 1 && mReqChannelCount == 1) { 5359 mFrameCount >>= 1; 5360 } 5361 5362 } 5363 mRsmpInIndex = mFrameCount; 5364} 5365 5366unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5367{ 5368 Mutex::Autolock _l(mLock); 5369 if (initCheck() != NO_ERROR) { 5370 return 0; 5371 } 5372 5373 return mInput->stream->get_input_frames_lost(mInput->stream); 5374} 5375 5376uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5377{ 5378 Mutex::Autolock _l(mLock); 5379 uint32_t result = 0; 5380 if (getEffectChain_l(sessionId) != 0) { 5381 result = EFFECT_SESSION; 5382 } 5383 5384 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5385 result |= TRACK_SESSION; 5386 } 5387 5388 return result; 5389} 5390 5391AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5392{ 5393 Mutex::Autolock _l(mLock); 5394 return mTrack; 5395} 5396 5397AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5398{ 5399 Mutex::Autolock _l(mLock); 5400 return mInput; 5401} 5402 5403AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5404{ 5405 Mutex::Autolock _l(mLock); 5406 AudioStreamIn *input = mInput; 5407 mInput = NULL; 5408 return input; 5409} 5410 5411// this method must always be called either with ThreadBase mLock held or inside the thread loop 5412audio_stream_t* AudioFlinger::RecordThread::stream() 5413{ 5414 if (mInput == NULL) { 5415 return NULL; 5416 } 5417 return &mInput->stream->common; 5418} 5419 5420 5421// ---------------------------------------------------------------------------- 5422 5423audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5424 uint32_t *pSamplingRate, 5425 audio_format_t *pFormat, 5426 uint32_t *pChannels, 5427 uint32_t *pLatencyMs, 5428 uint32_t flags) 5429{ 5430 status_t status; 5431 PlaybackThread *thread = NULL; 5432 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5433 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5434 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5435 uint32_t channels = pChannels ? *pChannels : 0; 5436 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5437 audio_stream_out_t *outStream; 5438 audio_hw_device_t *outHwDev; 5439 5440 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5441 pDevices ? *pDevices : 0, 5442 samplingRate, 5443 format, 5444 channels, 5445 flags); 5446 5447 if (pDevices == NULL || *pDevices == 0) { 5448 return 0; 5449 } 5450 5451 Mutex::Autolock _l(mLock); 5452 5453 outHwDev = findSuitableHwDev_l(*pDevices); 5454 if (outHwDev == NULL) 5455 return 0; 5456 5457 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5458 &channels, &samplingRate, &outStream); 5459 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5460 outStream, 5461 samplingRate, 5462 format, 5463 channels, 5464 status); 5465 5466 mHardwareStatus = AUDIO_HW_IDLE; 5467 if (outStream != NULL) { 5468 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5469 audio_io_handle_t id = nextUniqueId(); 5470 5471 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5472 (format != AUDIO_FORMAT_PCM_16_BIT) || 5473 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5474 thread = new DirectOutputThread(this, output, id, *pDevices); 5475 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5476 } else { 5477 thread = new MixerThread(this, output, id, *pDevices); 5478 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5479 } 5480 mPlaybackThreads.add(id, thread); 5481 5482 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5483 if (pFormat != NULL) *pFormat = format; 5484 if (pChannels != NULL) *pChannels = channels; 5485 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5486 5487 // notify client processes of the new output creation 5488 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5489 return id; 5490 } 5491 5492 return 0; 5493} 5494 5495audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5496 audio_io_handle_t output2) 5497{ 5498 Mutex::Autolock _l(mLock); 5499 MixerThread *thread1 = checkMixerThread_l(output1); 5500 MixerThread *thread2 = checkMixerThread_l(output2); 5501 5502 if (thread1 == NULL || thread2 == NULL) { 5503 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5504 return 0; 5505 } 5506 5507 audio_io_handle_t id = nextUniqueId(); 5508 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5509 thread->addOutputTrack(thread2); 5510 mPlaybackThreads.add(id, thread); 5511 // notify client processes of the new output creation 5512 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5513 return id; 5514} 5515 5516status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5517{ 5518 // keep strong reference on the playback thread so that 5519 // it is not destroyed while exit() is executed 5520 sp <PlaybackThread> thread; 5521 { 5522 Mutex::Autolock _l(mLock); 5523 thread = checkPlaybackThread_l(output); 5524 if (thread == NULL) { 5525 return BAD_VALUE; 5526 } 5527 5528 ALOGV("closeOutput() %d", output); 5529 5530 if (thread->type() == ThreadBase::MIXER) { 5531 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5532 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5533 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5534 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5535 } 5536 } 5537 } 5538 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5539 mPlaybackThreads.removeItem(output); 5540 } 5541 thread->exit(); 5542 // The thread entity (active unit of execution) is no longer running here, 5543 // but the ThreadBase container still exists. 5544 5545 if (thread->type() != ThreadBase::DUPLICATING) { 5546 AudioStreamOut *out = thread->clearOutput(); 5547 assert(out != NULL); 5548 // from now on thread->mOutput is NULL 5549 out->hwDev->close_output_stream(out->hwDev, out->stream); 5550 delete out; 5551 } 5552 return NO_ERROR; 5553} 5554 5555status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5556{ 5557 Mutex::Autolock _l(mLock); 5558 PlaybackThread *thread = checkPlaybackThread_l(output); 5559 5560 if (thread == NULL) { 5561 return BAD_VALUE; 5562 } 5563 5564 ALOGV("suspendOutput() %d", output); 5565 thread->suspend(); 5566 5567 return NO_ERROR; 5568} 5569 5570status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5571{ 5572 Mutex::Autolock _l(mLock); 5573 PlaybackThread *thread = checkPlaybackThread_l(output); 5574 5575 if (thread == NULL) { 5576 return BAD_VALUE; 5577 } 5578 5579 ALOGV("restoreOutput() %d", output); 5580 5581 thread->restore(); 5582 5583 return NO_ERROR; 5584} 5585 5586audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5587 uint32_t *pSamplingRate, 5588 audio_format_t *pFormat, 5589 uint32_t *pChannels, 5590 audio_in_acoustics_t acoustics) 5591{ 5592 status_t status; 5593 RecordThread *thread = NULL; 5594 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5595 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5596 uint32_t channels = pChannels ? *pChannels : 0; 5597 uint32_t reqSamplingRate = samplingRate; 5598 audio_format_t reqFormat = format; 5599 uint32_t reqChannels = channels; 5600 audio_stream_in_t *inStream; 5601 audio_hw_device_t *inHwDev; 5602 5603 if (pDevices == NULL || *pDevices == 0) { 5604 return 0; 5605 } 5606 5607 Mutex::Autolock _l(mLock); 5608 5609 inHwDev = findSuitableHwDev_l(*pDevices); 5610 if (inHwDev == NULL) 5611 return 0; 5612 5613 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5614 &channels, &samplingRate, 5615 acoustics, 5616 &inStream); 5617 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5618 inStream, 5619 samplingRate, 5620 format, 5621 channels, 5622 acoustics, 5623 status); 5624 5625 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5626 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5627 // or stereo to mono conversions on 16 bit PCM inputs. 5628 if (inStream == NULL && status == BAD_VALUE && 5629 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5630 (samplingRate <= 2 * reqSamplingRate) && 5631 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5632 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5633 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5634 &channels, &samplingRate, 5635 acoustics, 5636 &inStream); 5637 } 5638 5639 if (inStream != NULL) { 5640 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5641 5642 audio_io_handle_t id = nextUniqueId(); 5643 // Start record thread 5644 // RecorThread require both input and output device indication to forward to audio 5645 // pre processing modules 5646 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5647 thread = new RecordThread(this, 5648 input, 5649 reqSamplingRate, 5650 reqChannels, 5651 id, 5652 device); 5653 mRecordThreads.add(id, thread); 5654 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5655 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5656 if (pFormat != NULL) *pFormat = format; 5657 if (pChannels != NULL) *pChannels = reqChannels; 5658 5659 input->stream->common.standby(&input->stream->common); 5660 5661 // notify client processes of the new input creation 5662 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5663 return id; 5664 } 5665 5666 return 0; 5667} 5668 5669status_t AudioFlinger::closeInput(audio_io_handle_t input) 5670{ 5671 // keep strong reference on the record thread so that 5672 // it is not destroyed while exit() is executed 5673 sp <RecordThread> thread; 5674 { 5675 Mutex::Autolock _l(mLock); 5676 thread = checkRecordThread_l(input); 5677 if (thread == NULL) { 5678 return BAD_VALUE; 5679 } 5680 5681 ALOGV("closeInput() %d", input); 5682 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5683 mRecordThreads.removeItem(input); 5684 } 5685 thread->exit(); 5686 // The thread entity (active unit of execution) is no longer running here, 5687 // but the ThreadBase container still exists. 5688 5689 AudioStreamIn *in = thread->clearInput(); 5690 assert(in != NULL); 5691 // from now on thread->mInput is NULL 5692 in->hwDev->close_input_stream(in->hwDev, in->stream); 5693 delete in; 5694 5695 return NO_ERROR; 5696} 5697 5698status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5699{ 5700 Mutex::Autolock _l(mLock); 5701 MixerThread *dstThread = checkMixerThread_l(output); 5702 if (dstThread == NULL) { 5703 ALOGW("setStreamOutput() bad output id %d", output); 5704 return BAD_VALUE; 5705 } 5706 5707 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5708 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5709 5710 dstThread->setStreamValid(stream, true); 5711 5712 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5713 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5714 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5715 MixerThread *srcThread = (MixerThread *)thread; 5716 srcThread->setStreamValid(stream, false); 5717 srcThread->invalidateTracks(stream); 5718 } 5719 } 5720 5721 return NO_ERROR; 5722} 5723 5724 5725int AudioFlinger::newAudioSessionId() 5726{ 5727 return nextUniqueId(); 5728} 5729 5730void AudioFlinger::acquireAudioSessionId(int audioSession) 5731{ 5732 Mutex::Autolock _l(mLock); 5733 pid_t caller = IPCThreadState::self()->getCallingPid(); 5734 ALOGV("acquiring %d from %d", audioSession, caller); 5735 size_t num = mAudioSessionRefs.size(); 5736 for (size_t i = 0; i< num; i++) { 5737 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5738 if (ref->sessionid == audioSession && ref->pid == caller) { 5739 ref->cnt++; 5740 ALOGV(" incremented refcount to %d", ref->cnt); 5741 return; 5742 } 5743 } 5744 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5745 ALOGV(" added new entry for %d", audioSession); 5746} 5747 5748void AudioFlinger::releaseAudioSessionId(int audioSession) 5749{ 5750 Mutex::Autolock _l(mLock); 5751 pid_t caller = IPCThreadState::self()->getCallingPid(); 5752 ALOGV("releasing %d from %d", audioSession, caller); 5753 size_t num = mAudioSessionRefs.size(); 5754 for (size_t i = 0; i< num; i++) { 5755 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5756 if (ref->sessionid == audioSession && ref->pid == caller) { 5757 ref->cnt--; 5758 ALOGV(" decremented refcount to %d", ref->cnt); 5759 if (ref->cnt == 0) { 5760 mAudioSessionRefs.removeAt(i); 5761 delete ref; 5762 purgeStaleEffects_l(); 5763 } 5764 return; 5765 } 5766 } 5767 ALOGW("session id %d not found for pid %d", audioSession, caller); 5768} 5769 5770void AudioFlinger::purgeStaleEffects_l() { 5771 5772 ALOGV("purging stale effects"); 5773 5774 Vector< sp<EffectChain> > chains; 5775 5776 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5777 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5778 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5779 sp<EffectChain> ec = t->mEffectChains[j]; 5780 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5781 chains.push(ec); 5782 } 5783 } 5784 } 5785 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5786 sp<RecordThread> t = mRecordThreads.valueAt(i); 5787 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5788 sp<EffectChain> ec = t->mEffectChains[j]; 5789 chains.push(ec); 5790 } 5791 } 5792 5793 for (size_t i = 0; i < chains.size(); i++) { 5794 sp<EffectChain> ec = chains[i]; 5795 int sessionid = ec->sessionId(); 5796 sp<ThreadBase> t = ec->mThread.promote(); 5797 if (t == 0) { 5798 continue; 5799 } 5800 size_t numsessionrefs = mAudioSessionRefs.size(); 5801 bool found = false; 5802 for (size_t k = 0; k < numsessionrefs; k++) { 5803 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5804 if (ref->sessionid == sessionid) { 5805 ALOGV(" session %d still exists for %d with %d refs", 5806 sessionid, ref->pid, ref->cnt); 5807 found = true; 5808 break; 5809 } 5810 } 5811 if (!found) { 5812 // remove all effects from the chain 5813 while (ec->mEffects.size()) { 5814 sp<EffectModule> effect = ec->mEffects[0]; 5815 effect->unPin(); 5816 Mutex::Autolock _l (t->mLock); 5817 t->removeEffect_l(effect); 5818 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5819 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5820 if (handle != 0) { 5821 handle->mEffect.clear(); 5822 if (handle->mHasControl && handle->mEnabled) { 5823 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5824 } 5825 } 5826 } 5827 AudioSystem::unregisterEffect(effect->id()); 5828 } 5829 } 5830 } 5831 return; 5832} 5833 5834// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5835AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5836{ 5837 return mPlaybackThreads.valueFor(output).get(); 5838} 5839 5840// checkMixerThread_l() must be called with AudioFlinger::mLock held 5841AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5842{ 5843 PlaybackThread *thread = checkPlaybackThread_l(output); 5844 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5845} 5846 5847// checkRecordThread_l() must be called with AudioFlinger::mLock held 5848AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5849{ 5850 return mRecordThreads.valueFor(input).get(); 5851} 5852 5853uint32_t AudioFlinger::nextUniqueId() 5854{ 5855 return android_atomic_inc(&mNextUniqueId); 5856} 5857 5858AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5859{ 5860 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5861 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5862 AudioStreamOut *output = thread->getOutput(); 5863 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5864 return thread; 5865 } 5866 } 5867 return NULL; 5868} 5869 5870uint32_t AudioFlinger::primaryOutputDevice_l() 5871{ 5872 PlaybackThread *thread = primaryPlaybackThread_l(); 5873 5874 if (thread == NULL) { 5875 return 0; 5876 } 5877 5878 return thread->device(); 5879} 5880 5881 5882// ---------------------------------------------------------------------------- 5883// Effect management 5884// ---------------------------------------------------------------------------- 5885 5886 5887status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5888{ 5889 Mutex::Autolock _l(mLock); 5890 return EffectQueryNumberEffects(numEffects); 5891} 5892 5893status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5894{ 5895 Mutex::Autolock _l(mLock); 5896 return EffectQueryEffect(index, descriptor); 5897} 5898 5899status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5900 effect_descriptor_t *descriptor) const 5901{ 5902 Mutex::Autolock _l(mLock); 5903 return EffectGetDescriptor(pUuid, descriptor); 5904} 5905 5906 5907sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5908 effect_descriptor_t *pDesc, 5909 const sp<IEffectClient>& effectClient, 5910 int32_t priority, 5911 audio_io_handle_t io, 5912 int sessionId, 5913 status_t *status, 5914 int *id, 5915 int *enabled) 5916{ 5917 status_t lStatus = NO_ERROR; 5918 sp<EffectHandle> handle; 5919 effect_descriptor_t desc; 5920 5921 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5922 pid, effectClient.get(), priority, sessionId, io); 5923 5924 if (pDesc == NULL) { 5925 lStatus = BAD_VALUE; 5926 goto Exit; 5927 } 5928 5929 // check audio settings permission for global effects 5930 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5931 lStatus = PERMISSION_DENIED; 5932 goto Exit; 5933 } 5934 5935 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5936 // that can only be created by audio policy manager (running in same process) 5937 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5938 lStatus = PERMISSION_DENIED; 5939 goto Exit; 5940 } 5941 5942 if (io == 0) { 5943 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5944 // output must be specified by AudioPolicyManager when using session 5945 // AUDIO_SESSION_OUTPUT_STAGE 5946 lStatus = BAD_VALUE; 5947 goto Exit; 5948 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5949 // if the output returned by getOutputForEffect() is removed before we lock the 5950 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5951 // and we will exit safely 5952 io = AudioSystem::getOutputForEffect(&desc); 5953 } 5954 } 5955 5956 { 5957 Mutex::Autolock _l(mLock); 5958 5959 5960 if (!EffectIsNullUuid(&pDesc->uuid)) { 5961 // if uuid is specified, request effect descriptor 5962 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5963 if (lStatus < 0) { 5964 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5965 goto Exit; 5966 } 5967 } else { 5968 // if uuid is not specified, look for an available implementation 5969 // of the required type in effect factory 5970 if (EffectIsNullUuid(&pDesc->type)) { 5971 ALOGW("createEffect() no effect type"); 5972 lStatus = BAD_VALUE; 5973 goto Exit; 5974 } 5975 uint32_t numEffects = 0; 5976 effect_descriptor_t d; 5977 d.flags = 0; // prevent compiler warning 5978 bool found = false; 5979 5980 lStatus = EffectQueryNumberEffects(&numEffects); 5981 if (lStatus < 0) { 5982 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5983 goto Exit; 5984 } 5985 for (uint32_t i = 0; i < numEffects; i++) { 5986 lStatus = EffectQueryEffect(i, &desc); 5987 if (lStatus < 0) { 5988 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5989 continue; 5990 } 5991 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5992 // If matching type found save effect descriptor. If the session is 5993 // 0 and the effect is not auxiliary, continue enumeration in case 5994 // an auxiliary version of this effect type is available 5995 found = true; 5996 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5997 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5998 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5999 break; 6000 } 6001 } 6002 } 6003 if (!found) { 6004 lStatus = BAD_VALUE; 6005 ALOGW("createEffect() effect not found"); 6006 goto Exit; 6007 } 6008 // For same effect type, chose auxiliary version over insert version if 6009 // connect to output mix (Compliance to OpenSL ES) 6010 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6011 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6012 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6013 } 6014 } 6015 6016 // Do not allow auxiliary effects on a session different from 0 (output mix) 6017 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6018 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6019 lStatus = INVALID_OPERATION; 6020 goto Exit; 6021 } 6022 6023 // check recording permission for visualizer 6024 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6025 !recordingAllowed()) { 6026 lStatus = PERMISSION_DENIED; 6027 goto Exit; 6028 } 6029 6030 // return effect descriptor 6031 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6032 6033 // If output is not specified try to find a matching audio session ID in one of the 6034 // output threads. 6035 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6036 // because of code checking output when entering the function. 6037 // Note: io is never 0 when creating an effect on an input 6038 if (io == 0) { 6039 // look for the thread where the specified audio session is present 6040 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6041 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6042 io = mPlaybackThreads.keyAt(i); 6043 break; 6044 } 6045 } 6046 if (io == 0) { 6047 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6048 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6049 io = mRecordThreads.keyAt(i); 6050 break; 6051 } 6052 } 6053 } 6054 // If no output thread contains the requested session ID, default to 6055 // first output. The effect chain will be moved to the correct output 6056 // thread when a track with the same session ID is created 6057 if (io == 0 && mPlaybackThreads.size()) { 6058 io = mPlaybackThreads.keyAt(0); 6059 } 6060 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6061 } 6062 ThreadBase *thread = checkRecordThread_l(io); 6063 if (thread == NULL) { 6064 thread = checkPlaybackThread_l(io); 6065 if (thread == NULL) { 6066 ALOGE("createEffect() unknown output thread"); 6067 lStatus = BAD_VALUE; 6068 goto Exit; 6069 } 6070 } 6071 6072 sp<Client> client = registerPid_l(pid); 6073 6074 // create effect on selected output thread 6075 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6076 &desc, enabled, &lStatus); 6077 if (handle != 0 && id != NULL) { 6078 *id = handle->id(); 6079 } 6080 } 6081 6082Exit: 6083 if(status) { 6084 *status = lStatus; 6085 } 6086 return handle; 6087} 6088 6089status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6090 audio_io_handle_t dstOutput) 6091{ 6092 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6093 sessionId, srcOutput, dstOutput); 6094 Mutex::Autolock _l(mLock); 6095 if (srcOutput == dstOutput) { 6096 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6097 return NO_ERROR; 6098 } 6099 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6100 if (srcThread == NULL) { 6101 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6102 return BAD_VALUE; 6103 } 6104 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6105 if (dstThread == NULL) { 6106 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6107 return BAD_VALUE; 6108 } 6109 6110 Mutex::Autolock _dl(dstThread->mLock); 6111 Mutex::Autolock _sl(srcThread->mLock); 6112 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6113 6114 return NO_ERROR; 6115} 6116 6117// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6118status_t AudioFlinger::moveEffectChain_l(int sessionId, 6119 AudioFlinger::PlaybackThread *srcThread, 6120 AudioFlinger::PlaybackThread *dstThread, 6121 bool reRegister) 6122{ 6123 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6124 sessionId, srcThread, dstThread); 6125 6126 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6127 if (chain == 0) { 6128 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6129 sessionId, srcThread); 6130 return INVALID_OPERATION; 6131 } 6132 6133 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6134 // so that a new chain is created with correct parameters when first effect is added. This is 6135 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6136 // removed. 6137 srcThread->removeEffectChain_l(chain); 6138 6139 // transfer all effects one by one so that new effect chain is created on new thread with 6140 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6141 audio_io_handle_t dstOutput = dstThread->id(); 6142 sp<EffectChain> dstChain; 6143 uint32_t strategy = 0; // prevent compiler warning 6144 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6145 while (effect != 0) { 6146 srcThread->removeEffect_l(effect); 6147 dstThread->addEffect_l(effect); 6148 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6149 if (effect->state() == EffectModule::ACTIVE || 6150 effect->state() == EffectModule::STOPPING) { 6151 effect->start(); 6152 } 6153 // if the move request is not received from audio policy manager, the effect must be 6154 // re-registered with the new strategy and output 6155 if (dstChain == 0) { 6156 dstChain = effect->chain().promote(); 6157 if (dstChain == 0) { 6158 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6159 srcThread->addEffect_l(effect); 6160 return NO_INIT; 6161 } 6162 strategy = dstChain->strategy(); 6163 } 6164 if (reRegister) { 6165 AudioSystem::unregisterEffect(effect->id()); 6166 AudioSystem::registerEffect(&effect->desc(), 6167 dstOutput, 6168 strategy, 6169 sessionId, 6170 effect->id()); 6171 } 6172 effect = chain->getEffectFromId_l(0); 6173 } 6174 6175 return NO_ERROR; 6176} 6177 6178 6179// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6180sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6181 const sp<AudioFlinger::Client>& client, 6182 const sp<IEffectClient>& effectClient, 6183 int32_t priority, 6184 int sessionId, 6185 effect_descriptor_t *desc, 6186 int *enabled, 6187 status_t *status 6188 ) 6189{ 6190 sp<EffectModule> effect; 6191 sp<EffectHandle> handle; 6192 status_t lStatus; 6193 sp<EffectChain> chain; 6194 bool chainCreated = false; 6195 bool effectCreated = false; 6196 bool effectRegistered = false; 6197 6198 lStatus = initCheck(); 6199 if (lStatus != NO_ERROR) { 6200 ALOGW("createEffect_l() Audio driver not initialized."); 6201 goto Exit; 6202 } 6203 6204 // Do not allow effects with session ID 0 on direct output or duplicating threads 6205 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6206 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6207 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6208 desc->name, sessionId); 6209 lStatus = BAD_VALUE; 6210 goto Exit; 6211 } 6212 // Only Pre processor effects are allowed on input threads and only on input threads 6213 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6214 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6215 desc->name, desc->flags, mType); 6216 lStatus = BAD_VALUE; 6217 goto Exit; 6218 } 6219 6220 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6221 6222 { // scope for mLock 6223 Mutex::Autolock _l(mLock); 6224 6225 // check for existing effect chain with the requested audio session 6226 chain = getEffectChain_l(sessionId); 6227 if (chain == 0) { 6228 // create a new chain for this session 6229 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6230 chain = new EffectChain(this, sessionId); 6231 addEffectChain_l(chain); 6232 chain->setStrategy(getStrategyForSession_l(sessionId)); 6233 chainCreated = true; 6234 } else { 6235 effect = chain->getEffectFromDesc_l(desc); 6236 } 6237 6238 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6239 6240 if (effect == 0) { 6241 int id = mAudioFlinger->nextUniqueId(); 6242 // Check CPU and memory usage 6243 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6244 if (lStatus != NO_ERROR) { 6245 goto Exit; 6246 } 6247 effectRegistered = true; 6248 // create a new effect module if none present in the chain 6249 effect = new EffectModule(this, chain, desc, id, sessionId); 6250 lStatus = effect->status(); 6251 if (lStatus != NO_ERROR) { 6252 goto Exit; 6253 } 6254 lStatus = chain->addEffect_l(effect); 6255 if (lStatus != NO_ERROR) { 6256 goto Exit; 6257 } 6258 effectCreated = true; 6259 6260 effect->setDevice(mDevice); 6261 effect->setMode(mAudioFlinger->getMode()); 6262 } 6263 // create effect handle and connect it to effect module 6264 handle = new EffectHandle(effect, client, effectClient, priority); 6265 lStatus = effect->addHandle(handle); 6266 if (enabled != NULL) { 6267 *enabled = (int)effect->isEnabled(); 6268 } 6269 } 6270 6271Exit: 6272 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6273 Mutex::Autolock _l(mLock); 6274 if (effectCreated) { 6275 chain->removeEffect_l(effect); 6276 } 6277 if (effectRegistered) { 6278 AudioSystem::unregisterEffect(effect->id()); 6279 } 6280 if (chainCreated) { 6281 removeEffectChain_l(chain); 6282 } 6283 handle.clear(); 6284 } 6285 6286 if(status) { 6287 *status = lStatus; 6288 } 6289 return handle; 6290} 6291 6292sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6293{ 6294 sp<EffectChain> chain = getEffectChain_l(sessionId); 6295 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6296} 6297 6298// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6299// PlaybackThread::mLock held 6300status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6301{ 6302 // check for existing effect chain with the requested audio session 6303 int sessionId = effect->sessionId(); 6304 sp<EffectChain> chain = getEffectChain_l(sessionId); 6305 bool chainCreated = false; 6306 6307 if (chain == 0) { 6308 // create a new chain for this session 6309 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6310 chain = new EffectChain(this, sessionId); 6311 addEffectChain_l(chain); 6312 chain->setStrategy(getStrategyForSession_l(sessionId)); 6313 chainCreated = true; 6314 } 6315 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6316 6317 if (chain->getEffectFromId_l(effect->id()) != 0) { 6318 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6319 this, effect->desc().name, chain.get()); 6320 return BAD_VALUE; 6321 } 6322 6323 status_t status = chain->addEffect_l(effect); 6324 if (status != NO_ERROR) { 6325 if (chainCreated) { 6326 removeEffectChain_l(chain); 6327 } 6328 return status; 6329 } 6330 6331 effect->setDevice(mDevice); 6332 effect->setMode(mAudioFlinger->getMode()); 6333 return NO_ERROR; 6334} 6335 6336void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6337 6338 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6339 effect_descriptor_t desc = effect->desc(); 6340 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6341 detachAuxEffect_l(effect->id()); 6342 } 6343 6344 sp<EffectChain> chain = effect->chain().promote(); 6345 if (chain != 0) { 6346 // remove effect chain if removing last effect 6347 if (chain->removeEffect_l(effect) == 0) { 6348 removeEffectChain_l(chain); 6349 } 6350 } else { 6351 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6352 } 6353} 6354 6355void AudioFlinger::ThreadBase::lockEffectChains_l( 6356 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6357{ 6358 effectChains = mEffectChains; 6359 for (size_t i = 0; i < mEffectChains.size(); i++) { 6360 mEffectChains[i]->lock(); 6361 } 6362} 6363 6364void AudioFlinger::ThreadBase::unlockEffectChains( 6365 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6366{ 6367 for (size_t i = 0; i < effectChains.size(); i++) { 6368 effectChains[i]->unlock(); 6369 } 6370} 6371 6372sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6373{ 6374 Mutex::Autolock _l(mLock); 6375 return getEffectChain_l(sessionId); 6376} 6377 6378sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6379{ 6380 size_t size = mEffectChains.size(); 6381 for (size_t i = 0; i < size; i++) { 6382 if (mEffectChains[i]->sessionId() == sessionId) { 6383 return mEffectChains[i]; 6384 } 6385 } 6386 return 0; 6387} 6388 6389void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6390{ 6391 Mutex::Autolock _l(mLock); 6392 size_t size = mEffectChains.size(); 6393 for (size_t i = 0; i < size; i++) { 6394 mEffectChains[i]->setMode_l(mode); 6395 } 6396} 6397 6398void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6399 const wp<EffectHandle>& handle, 6400 bool unpinIfLast) { 6401 6402 Mutex::Autolock _l(mLock); 6403 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6404 // delete the effect module if removing last handle on it 6405 if (effect->removeHandle(handle) == 0) { 6406 if (!effect->isPinned() || unpinIfLast) { 6407 removeEffect_l(effect); 6408 AudioSystem::unregisterEffect(effect->id()); 6409 } 6410 } 6411} 6412 6413status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6414{ 6415 int session = chain->sessionId(); 6416 int16_t *buffer = mMixBuffer; 6417 bool ownsBuffer = false; 6418 6419 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6420 if (session > 0) { 6421 // Only one effect chain can be present in direct output thread and it uses 6422 // the mix buffer as input 6423 if (mType != DIRECT) { 6424 size_t numSamples = mFrameCount * mChannelCount; 6425 buffer = new int16_t[numSamples]; 6426 memset(buffer, 0, numSamples * sizeof(int16_t)); 6427 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6428 ownsBuffer = true; 6429 } 6430 6431 // Attach all tracks with same session ID to this chain. 6432 for (size_t i = 0; i < mTracks.size(); ++i) { 6433 sp<Track> track = mTracks[i]; 6434 if (session == track->sessionId()) { 6435 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6436 track->setMainBuffer(buffer); 6437 chain->incTrackCnt(); 6438 } 6439 } 6440 6441 // indicate all active tracks in the chain 6442 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6443 sp<Track> track = mActiveTracks[i].promote(); 6444 if (track == 0) continue; 6445 if (session == track->sessionId()) { 6446 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6447 chain->incActiveTrackCnt(); 6448 } 6449 } 6450 } 6451 6452 chain->setInBuffer(buffer, ownsBuffer); 6453 chain->setOutBuffer(mMixBuffer); 6454 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6455 // chains list in order to be processed last as it contains output stage effects 6456 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6457 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6458 // after track specific effects and before output stage 6459 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6460 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6461 // Effect chain for other sessions are inserted at beginning of effect 6462 // chains list to be processed before output mix effects. Relative order between other 6463 // sessions is not important 6464 size_t size = mEffectChains.size(); 6465 size_t i = 0; 6466 for (i = 0; i < size; i++) { 6467 if (mEffectChains[i]->sessionId() < session) break; 6468 } 6469 mEffectChains.insertAt(chain, i); 6470 checkSuspendOnAddEffectChain_l(chain); 6471 6472 return NO_ERROR; 6473} 6474 6475size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6476{ 6477 int session = chain->sessionId(); 6478 6479 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6480 6481 for (size_t i = 0; i < mEffectChains.size(); i++) { 6482 if (chain == mEffectChains[i]) { 6483 mEffectChains.removeAt(i); 6484 // detach all active tracks from the chain 6485 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6486 sp<Track> track = mActiveTracks[i].promote(); 6487 if (track == 0) continue; 6488 if (session == track->sessionId()) { 6489 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6490 chain.get(), session); 6491 chain->decActiveTrackCnt(); 6492 } 6493 } 6494 6495 // detach all tracks with same session ID from this chain 6496 for (size_t i = 0; i < mTracks.size(); ++i) { 6497 sp<Track> track = mTracks[i]; 6498 if (session == track->sessionId()) { 6499 track->setMainBuffer(mMixBuffer); 6500 chain->decTrackCnt(); 6501 } 6502 } 6503 break; 6504 } 6505 } 6506 return mEffectChains.size(); 6507} 6508 6509status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6510 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6511{ 6512 Mutex::Autolock _l(mLock); 6513 return attachAuxEffect_l(track, EffectId); 6514} 6515 6516status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6517 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6518{ 6519 status_t status = NO_ERROR; 6520 6521 if (EffectId == 0) { 6522 track->setAuxBuffer(0, NULL); 6523 } else { 6524 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6525 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6526 if (effect != 0) { 6527 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6528 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6529 } else { 6530 status = INVALID_OPERATION; 6531 } 6532 } else { 6533 status = BAD_VALUE; 6534 } 6535 } 6536 return status; 6537} 6538 6539void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6540{ 6541 for (size_t i = 0; i < mTracks.size(); ++i) { 6542 sp<Track> track = mTracks[i]; 6543 if (track->auxEffectId() == effectId) { 6544 attachAuxEffect_l(track, 0); 6545 } 6546 } 6547} 6548 6549status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6550{ 6551 // only one chain per input thread 6552 if (mEffectChains.size() != 0) { 6553 return INVALID_OPERATION; 6554 } 6555 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6556 6557 chain->setInBuffer(NULL); 6558 chain->setOutBuffer(NULL); 6559 6560 checkSuspendOnAddEffectChain_l(chain); 6561 6562 mEffectChains.add(chain); 6563 6564 return NO_ERROR; 6565} 6566 6567size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6568{ 6569 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6570 ALOGW_IF(mEffectChains.size() != 1, 6571 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6572 chain.get(), mEffectChains.size(), this); 6573 if (mEffectChains.size() == 1) { 6574 mEffectChains.removeAt(0); 6575 } 6576 return 0; 6577} 6578 6579// ---------------------------------------------------------------------------- 6580// EffectModule implementation 6581// ---------------------------------------------------------------------------- 6582 6583#undef LOG_TAG 6584#define LOG_TAG "AudioFlinger::EffectModule" 6585 6586AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6587 const wp<AudioFlinger::EffectChain>& chain, 6588 effect_descriptor_t *desc, 6589 int id, 6590 int sessionId) 6591 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6592 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6593{ 6594 ALOGV("Constructor %p", this); 6595 int lStatus; 6596 if (thread == NULL) { 6597 return; 6598 } 6599 6600 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6601 6602 // create effect engine from effect factory 6603 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6604 6605 if (mStatus != NO_ERROR) { 6606 return; 6607 } 6608 lStatus = init(); 6609 if (lStatus < 0) { 6610 mStatus = lStatus; 6611 goto Error; 6612 } 6613 6614 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6615 mPinned = true; 6616 } 6617 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6618 return; 6619Error: 6620 EffectRelease(mEffectInterface); 6621 mEffectInterface = NULL; 6622 ALOGV("Constructor Error %d", mStatus); 6623} 6624 6625AudioFlinger::EffectModule::~EffectModule() 6626{ 6627 ALOGV("Destructor %p", this); 6628 if (mEffectInterface != NULL) { 6629 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6630 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6631 sp<ThreadBase> thread = mThread.promote(); 6632 if (thread != 0) { 6633 audio_stream_t *stream = thread->stream(); 6634 if (stream != NULL) { 6635 stream->remove_audio_effect(stream, mEffectInterface); 6636 } 6637 } 6638 } 6639 // release effect engine 6640 EffectRelease(mEffectInterface); 6641 } 6642} 6643 6644status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6645{ 6646 status_t status; 6647 6648 Mutex::Autolock _l(mLock); 6649 int priority = handle->priority(); 6650 size_t size = mHandles.size(); 6651 sp<EffectHandle> h; 6652 size_t i; 6653 for (i = 0; i < size; i++) { 6654 h = mHandles[i].promote(); 6655 if (h == 0) continue; 6656 if (h->priority() <= priority) break; 6657 } 6658 // if inserted in first place, move effect control from previous owner to this handle 6659 if (i == 0) { 6660 bool enabled = false; 6661 if (h != 0) { 6662 enabled = h->enabled(); 6663 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6664 } 6665 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6666 status = NO_ERROR; 6667 } else { 6668 status = ALREADY_EXISTS; 6669 } 6670 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6671 mHandles.insertAt(handle, i); 6672 return status; 6673} 6674 6675size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6676{ 6677 Mutex::Autolock _l(mLock); 6678 size_t size = mHandles.size(); 6679 size_t i; 6680 for (i = 0; i < size; i++) { 6681 if (mHandles[i] == handle) break; 6682 } 6683 if (i == size) { 6684 return size; 6685 } 6686 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6687 6688 bool enabled = false; 6689 EffectHandle *hdl = handle.unsafe_get(); 6690 if (hdl != NULL) { 6691 ALOGV("removeHandle() unsafe_get OK"); 6692 enabled = hdl->enabled(); 6693 } 6694 mHandles.removeAt(i); 6695 size = mHandles.size(); 6696 // if removed from first place, move effect control from this handle to next in line 6697 if (i == 0 && size != 0) { 6698 sp<EffectHandle> h = mHandles[0].promote(); 6699 if (h != 0) { 6700 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6701 } 6702 } 6703 6704 // Prevent calls to process() and other functions on effect interface from now on. 6705 // The effect engine will be released by the destructor when the last strong reference on 6706 // this object is released which can happen after next process is called. 6707 if (size == 0 && !mPinned) { 6708 mState = DESTROYED; 6709 } 6710 6711 return size; 6712} 6713 6714sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6715{ 6716 Mutex::Autolock _l(mLock); 6717 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6718} 6719 6720void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6721{ 6722 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6723 // keep a strong reference on this EffectModule to avoid calling the 6724 // destructor before we exit 6725 sp<EffectModule> keep(this); 6726 { 6727 sp<ThreadBase> thread = mThread.promote(); 6728 if (thread != 0) { 6729 thread->disconnectEffect(keep, handle, unpinIfLast); 6730 } 6731 } 6732} 6733 6734void AudioFlinger::EffectModule::updateState() { 6735 Mutex::Autolock _l(mLock); 6736 6737 switch (mState) { 6738 case RESTART: 6739 reset_l(); 6740 // FALL THROUGH 6741 6742 case STARTING: 6743 // clear auxiliary effect input buffer for next accumulation 6744 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6745 memset(mConfig.inputCfg.buffer.raw, 6746 0, 6747 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6748 } 6749 start_l(); 6750 mState = ACTIVE; 6751 break; 6752 case STOPPING: 6753 stop_l(); 6754 mDisableWaitCnt = mMaxDisableWaitCnt; 6755 mState = STOPPED; 6756 break; 6757 case STOPPED: 6758 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6759 // turn off sequence. 6760 if (--mDisableWaitCnt == 0) { 6761 reset_l(); 6762 mState = IDLE; 6763 } 6764 break; 6765 default: //IDLE , ACTIVE, DESTROYED 6766 break; 6767 } 6768} 6769 6770void AudioFlinger::EffectModule::process() 6771{ 6772 Mutex::Autolock _l(mLock); 6773 6774 if (mState == DESTROYED || mEffectInterface == NULL || 6775 mConfig.inputCfg.buffer.raw == NULL || 6776 mConfig.outputCfg.buffer.raw == NULL) { 6777 return; 6778 } 6779 6780 if (isProcessEnabled()) { 6781 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6782 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6783 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6784 mConfig.inputCfg.buffer.s32, 6785 mConfig.inputCfg.buffer.frameCount/2); 6786 } 6787 6788 // do the actual processing in the effect engine 6789 int ret = (*mEffectInterface)->process(mEffectInterface, 6790 &mConfig.inputCfg.buffer, 6791 &mConfig.outputCfg.buffer); 6792 6793 // force transition to IDLE state when engine is ready 6794 if (mState == STOPPED && ret == -ENODATA) { 6795 mDisableWaitCnt = 1; 6796 } 6797 6798 // clear auxiliary effect input buffer for next accumulation 6799 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6800 memset(mConfig.inputCfg.buffer.raw, 0, 6801 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6802 } 6803 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6804 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6805 // If an insert effect is idle and input buffer is different from output buffer, 6806 // accumulate input onto output 6807 sp<EffectChain> chain = mChain.promote(); 6808 if (chain != 0 && chain->activeTrackCnt() != 0) { 6809 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6810 int16_t *in = mConfig.inputCfg.buffer.s16; 6811 int16_t *out = mConfig.outputCfg.buffer.s16; 6812 for (size_t i = 0; i < frameCnt; i++) { 6813 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6814 } 6815 } 6816 } 6817} 6818 6819void AudioFlinger::EffectModule::reset_l() 6820{ 6821 if (mEffectInterface == NULL) { 6822 return; 6823 } 6824 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6825} 6826 6827status_t AudioFlinger::EffectModule::configure() 6828{ 6829 uint32_t channels; 6830 if (mEffectInterface == NULL) { 6831 return NO_INIT; 6832 } 6833 6834 sp<ThreadBase> thread = mThread.promote(); 6835 if (thread == 0) { 6836 return DEAD_OBJECT; 6837 } 6838 6839 // TODO: handle configuration of effects replacing track process 6840 if (thread->channelCount() == 1) { 6841 channels = AUDIO_CHANNEL_OUT_MONO; 6842 } else { 6843 channels = AUDIO_CHANNEL_OUT_STEREO; 6844 } 6845 6846 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6847 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6848 } else { 6849 mConfig.inputCfg.channels = channels; 6850 } 6851 mConfig.outputCfg.channels = channels; 6852 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6853 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6854 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6855 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6856 mConfig.inputCfg.bufferProvider.cookie = NULL; 6857 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6858 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6859 mConfig.outputCfg.bufferProvider.cookie = NULL; 6860 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6861 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6862 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6863 // Insert effect: 6864 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6865 // always overwrites output buffer: input buffer == output buffer 6866 // - in other sessions: 6867 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6868 // other effect: overwrites output buffer: input buffer == output buffer 6869 // Auxiliary effect: 6870 // accumulates in output buffer: input buffer != output buffer 6871 // Therefore: accumulate <=> input buffer != output buffer 6872 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6873 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6874 } else { 6875 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6876 } 6877 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6878 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6879 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6880 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6881 6882 ALOGV("configure() %p thread %p buffer %p framecount %d", 6883 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6884 6885 status_t cmdStatus; 6886 uint32_t size = sizeof(int); 6887 status_t status = (*mEffectInterface)->command(mEffectInterface, 6888 EFFECT_CMD_SET_CONFIG, 6889 sizeof(effect_config_t), 6890 &mConfig, 6891 &size, 6892 &cmdStatus); 6893 if (status == 0) { 6894 status = cmdStatus; 6895 } 6896 6897 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6898 (1000 * mConfig.outputCfg.buffer.frameCount); 6899 6900 return status; 6901} 6902 6903status_t AudioFlinger::EffectModule::init() 6904{ 6905 Mutex::Autolock _l(mLock); 6906 if (mEffectInterface == NULL) { 6907 return NO_INIT; 6908 } 6909 status_t cmdStatus; 6910 uint32_t size = sizeof(status_t); 6911 status_t status = (*mEffectInterface)->command(mEffectInterface, 6912 EFFECT_CMD_INIT, 6913 0, 6914 NULL, 6915 &size, 6916 &cmdStatus); 6917 if (status == 0) { 6918 status = cmdStatus; 6919 } 6920 return status; 6921} 6922 6923status_t AudioFlinger::EffectModule::start() 6924{ 6925 Mutex::Autolock _l(mLock); 6926 return start_l(); 6927} 6928 6929status_t AudioFlinger::EffectModule::start_l() 6930{ 6931 if (mEffectInterface == NULL) { 6932 return NO_INIT; 6933 } 6934 status_t cmdStatus; 6935 uint32_t size = sizeof(status_t); 6936 status_t status = (*mEffectInterface)->command(mEffectInterface, 6937 EFFECT_CMD_ENABLE, 6938 0, 6939 NULL, 6940 &size, 6941 &cmdStatus); 6942 if (status == 0) { 6943 status = cmdStatus; 6944 } 6945 if (status == 0 && 6946 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6947 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6948 sp<ThreadBase> thread = mThread.promote(); 6949 if (thread != 0) { 6950 audio_stream_t *stream = thread->stream(); 6951 if (stream != NULL) { 6952 stream->add_audio_effect(stream, mEffectInterface); 6953 } 6954 } 6955 } 6956 return status; 6957} 6958 6959status_t AudioFlinger::EffectModule::stop() 6960{ 6961 Mutex::Autolock _l(mLock); 6962 return stop_l(); 6963} 6964 6965status_t AudioFlinger::EffectModule::stop_l() 6966{ 6967 if (mEffectInterface == NULL) { 6968 return NO_INIT; 6969 } 6970 status_t cmdStatus; 6971 uint32_t size = sizeof(status_t); 6972 status_t status = (*mEffectInterface)->command(mEffectInterface, 6973 EFFECT_CMD_DISABLE, 6974 0, 6975 NULL, 6976 &size, 6977 &cmdStatus); 6978 if (status == 0) { 6979 status = cmdStatus; 6980 } 6981 if (status == 0 && 6982 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6983 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6984 sp<ThreadBase> thread = mThread.promote(); 6985 if (thread != 0) { 6986 audio_stream_t *stream = thread->stream(); 6987 if (stream != NULL) { 6988 stream->remove_audio_effect(stream, mEffectInterface); 6989 } 6990 } 6991 } 6992 return status; 6993} 6994 6995status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6996 uint32_t cmdSize, 6997 void *pCmdData, 6998 uint32_t *replySize, 6999 void *pReplyData) 7000{ 7001 Mutex::Autolock _l(mLock); 7002// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7003 7004 if (mState == DESTROYED || mEffectInterface == NULL) { 7005 return NO_INIT; 7006 } 7007 status_t status = (*mEffectInterface)->command(mEffectInterface, 7008 cmdCode, 7009 cmdSize, 7010 pCmdData, 7011 replySize, 7012 pReplyData); 7013 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7014 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7015 for (size_t i = 1; i < mHandles.size(); i++) { 7016 sp<EffectHandle> h = mHandles[i].promote(); 7017 if (h != 0) { 7018 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7019 } 7020 } 7021 } 7022 return status; 7023} 7024 7025status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7026{ 7027 7028 Mutex::Autolock _l(mLock); 7029 ALOGV("setEnabled %p enabled %d", this, enabled); 7030 7031 if (enabled != isEnabled()) { 7032 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7033 if (enabled && status != NO_ERROR) { 7034 return status; 7035 } 7036 7037 switch (mState) { 7038 // going from disabled to enabled 7039 case IDLE: 7040 mState = STARTING; 7041 break; 7042 case STOPPED: 7043 mState = RESTART; 7044 break; 7045 case STOPPING: 7046 mState = ACTIVE; 7047 break; 7048 7049 // going from enabled to disabled 7050 case RESTART: 7051 mState = STOPPED; 7052 break; 7053 case STARTING: 7054 mState = IDLE; 7055 break; 7056 case ACTIVE: 7057 mState = STOPPING; 7058 break; 7059 case DESTROYED: 7060 return NO_ERROR; // simply ignore as we are being destroyed 7061 } 7062 for (size_t i = 1; i < mHandles.size(); i++) { 7063 sp<EffectHandle> h = mHandles[i].promote(); 7064 if (h != 0) { 7065 h->setEnabled(enabled); 7066 } 7067 } 7068 } 7069 return NO_ERROR; 7070} 7071 7072bool AudioFlinger::EffectModule::isEnabled() const 7073{ 7074 switch (mState) { 7075 case RESTART: 7076 case STARTING: 7077 case ACTIVE: 7078 return true; 7079 case IDLE: 7080 case STOPPING: 7081 case STOPPED: 7082 case DESTROYED: 7083 default: 7084 return false; 7085 } 7086} 7087 7088bool AudioFlinger::EffectModule::isProcessEnabled() const 7089{ 7090 switch (mState) { 7091 case RESTART: 7092 case ACTIVE: 7093 case STOPPING: 7094 case STOPPED: 7095 return true; 7096 case IDLE: 7097 case STARTING: 7098 case DESTROYED: 7099 default: 7100 return false; 7101 } 7102} 7103 7104status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7105{ 7106 Mutex::Autolock _l(mLock); 7107 status_t status = NO_ERROR; 7108 7109 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7110 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7111 if (isProcessEnabled() && 7112 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7113 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7114 status_t cmdStatus; 7115 uint32_t volume[2]; 7116 uint32_t *pVolume = NULL; 7117 uint32_t size = sizeof(volume); 7118 volume[0] = *left; 7119 volume[1] = *right; 7120 if (controller) { 7121 pVolume = volume; 7122 } 7123 status = (*mEffectInterface)->command(mEffectInterface, 7124 EFFECT_CMD_SET_VOLUME, 7125 size, 7126 volume, 7127 &size, 7128 pVolume); 7129 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7130 *left = volume[0]; 7131 *right = volume[1]; 7132 } 7133 } 7134 return status; 7135} 7136 7137status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7138{ 7139 Mutex::Autolock _l(mLock); 7140 status_t status = NO_ERROR; 7141 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7142 // audio pre processing modules on RecordThread can receive both output and 7143 // input device indication in the same call 7144 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7145 if (dev) { 7146 status_t cmdStatus; 7147 uint32_t size = sizeof(status_t); 7148 7149 status = (*mEffectInterface)->command(mEffectInterface, 7150 EFFECT_CMD_SET_DEVICE, 7151 sizeof(uint32_t), 7152 &dev, 7153 &size, 7154 &cmdStatus); 7155 if (status == NO_ERROR) { 7156 status = cmdStatus; 7157 } 7158 } 7159 dev = device & AUDIO_DEVICE_IN_ALL; 7160 if (dev) { 7161 status_t cmdStatus; 7162 uint32_t size = sizeof(status_t); 7163 7164 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7165 EFFECT_CMD_SET_INPUT_DEVICE, 7166 sizeof(uint32_t), 7167 &dev, 7168 &size, 7169 &cmdStatus); 7170 if (status2 == NO_ERROR) { 7171 status2 = cmdStatus; 7172 } 7173 if (status == NO_ERROR) { 7174 status = status2; 7175 } 7176 } 7177 } 7178 return status; 7179} 7180 7181status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7182{ 7183 Mutex::Autolock _l(mLock); 7184 status_t status = NO_ERROR; 7185 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7186 status_t cmdStatus; 7187 uint32_t size = sizeof(status_t); 7188 status = (*mEffectInterface)->command(mEffectInterface, 7189 EFFECT_CMD_SET_AUDIO_MODE, 7190 sizeof(audio_mode_t), 7191 &mode, 7192 &size, 7193 &cmdStatus); 7194 if (status == NO_ERROR) { 7195 status = cmdStatus; 7196 } 7197 } 7198 return status; 7199} 7200 7201void AudioFlinger::EffectModule::setSuspended(bool suspended) 7202{ 7203 Mutex::Autolock _l(mLock); 7204 mSuspended = suspended; 7205} 7206 7207bool AudioFlinger::EffectModule::suspended() const 7208{ 7209 Mutex::Autolock _l(mLock); 7210 return mSuspended; 7211} 7212 7213status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7214{ 7215 const size_t SIZE = 256; 7216 char buffer[SIZE]; 7217 String8 result; 7218 7219 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7220 result.append(buffer); 7221 7222 bool locked = tryLock(mLock); 7223 // failed to lock - AudioFlinger is probably deadlocked 7224 if (!locked) { 7225 result.append("\t\tCould not lock Fx mutex:\n"); 7226 } 7227 7228 result.append("\t\tSession Status State Engine:\n"); 7229 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7230 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7231 result.append(buffer); 7232 7233 result.append("\t\tDescriptor:\n"); 7234 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7235 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7236 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7237 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7238 result.append(buffer); 7239 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7240 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7241 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7242 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7243 result.append(buffer); 7244 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7245 mDescriptor.apiVersion, 7246 mDescriptor.flags); 7247 result.append(buffer); 7248 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7249 mDescriptor.name); 7250 result.append(buffer); 7251 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7252 mDescriptor.implementor); 7253 result.append(buffer); 7254 7255 result.append("\t\t- Input configuration:\n"); 7256 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7257 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7258 (uint32_t)mConfig.inputCfg.buffer.raw, 7259 mConfig.inputCfg.buffer.frameCount, 7260 mConfig.inputCfg.samplingRate, 7261 mConfig.inputCfg.channels, 7262 mConfig.inputCfg.format); 7263 result.append(buffer); 7264 7265 result.append("\t\t- Output configuration:\n"); 7266 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7267 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7268 (uint32_t)mConfig.outputCfg.buffer.raw, 7269 mConfig.outputCfg.buffer.frameCount, 7270 mConfig.outputCfg.samplingRate, 7271 mConfig.outputCfg.channels, 7272 mConfig.outputCfg.format); 7273 result.append(buffer); 7274 7275 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7276 result.append(buffer); 7277 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7278 for (size_t i = 0; i < mHandles.size(); ++i) { 7279 sp<EffectHandle> handle = mHandles[i].promote(); 7280 if (handle != 0) { 7281 handle->dump(buffer, SIZE); 7282 result.append(buffer); 7283 } 7284 } 7285 7286 result.append("\n"); 7287 7288 write(fd, result.string(), result.length()); 7289 7290 if (locked) { 7291 mLock.unlock(); 7292 } 7293 7294 return NO_ERROR; 7295} 7296 7297// ---------------------------------------------------------------------------- 7298// EffectHandle implementation 7299// ---------------------------------------------------------------------------- 7300 7301#undef LOG_TAG 7302#define LOG_TAG "AudioFlinger::EffectHandle" 7303 7304AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7305 const sp<AudioFlinger::Client>& client, 7306 const sp<IEffectClient>& effectClient, 7307 int32_t priority) 7308 : BnEffect(), 7309 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7310 mPriority(priority), mHasControl(false), mEnabled(false) 7311{ 7312 ALOGV("constructor %p", this); 7313 7314 if (client == 0) { 7315 return; 7316 } 7317 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7318 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7319 if (mCblkMemory != 0) { 7320 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7321 7322 if (mCblk != NULL) { 7323 new(mCblk) effect_param_cblk_t(); 7324 mBuffer = (uint8_t *)mCblk + bufOffset; 7325 } 7326 } else { 7327 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7328 return; 7329 } 7330} 7331 7332AudioFlinger::EffectHandle::~EffectHandle() 7333{ 7334 ALOGV("Destructor %p", this); 7335 disconnect(false); 7336 ALOGV("Destructor DONE %p", this); 7337} 7338 7339status_t AudioFlinger::EffectHandle::enable() 7340{ 7341 ALOGV("enable %p", this); 7342 if (!mHasControl) return INVALID_OPERATION; 7343 if (mEffect == 0) return DEAD_OBJECT; 7344 7345 if (mEnabled) { 7346 return NO_ERROR; 7347 } 7348 7349 mEnabled = true; 7350 7351 sp<ThreadBase> thread = mEffect->thread().promote(); 7352 if (thread != 0) { 7353 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7354 } 7355 7356 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7357 if (mEffect->suspended()) { 7358 return NO_ERROR; 7359 } 7360 7361 status_t status = mEffect->setEnabled(true); 7362 if (status != NO_ERROR) { 7363 if (thread != 0) { 7364 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7365 } 7366 mEnabled = false; 7367 } 7368 return status; 7369} 7370 7371status_t AudioFlinger::EffectHandle::disable() 7372{ 7373 ALOGV("disable %p", this); 7374 if (!mHasControl) return INVALID_OPERATION; 7375 if (mEffect == 0) return DEAD_OBJECT; 7376 7377 if (!mEnabled) { 7378 return NO_ERROR; 7379 } 7380 mEnabled = false; 7381 7382 if (mEffect->suspended()) { 7383 return NO_ERROR; 7384 } 7385 7386 status_t status = mEffect->setEnabled(false); 7387 7388 sp<ThreadBase> thread = mEffect->thread().promote(); 7389 if (thread != 0) { 7390 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7391 } 7392 7393 return status; 7394} 7395 7396void AudioFlinger::EffectHandle::disconnect() 7397{ 7398 disconnect(true); 7399} 7400 7401void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7402{ 7403 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7404 if (mEffect == 0) { 7405 return; 7406 } 7407 mEffect->disconnect(this, unpinIfLast); 7408 7409 if (mHasControl && mEnabled) { 7410 sp<ThreadBase> thread = mEffect->thread().promote(); 7411 if (thread != 0) { 7412 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7413 } 7414 } 7415 7416 // release sp on module => module destructor can be called now 7417 mEffect.clear(); 7418 if (mClient != 0) { 7419 if (mCblk != NULL) { 7420 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7421 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7422 } 7423 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7424 // Client destructor must run with AudioFlinger mutex locked 7425 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7426 mClient.clear(); 7427 } 7428} 7429 7430status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7431 uint32_t cmdSize, 7432 void *pCmdData, 7433 uint32_t *replySize, 7434 void *pReplyData) 7435{ 7436// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7437// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7438 7439 // only get parameter command is permitted for applications not controlling the effect 7440 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7441 return INVALID_OPERATION; 7442 } 7443 if (mEffect == 0) return DEAD_OBJECT; 7444 if (mClient == 0) return INVALID_OPERATION; 7445 7446 // handle commands that are not forwarded transparently to effect engine 7447 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7448 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7449 // no risk to block the whole media server process or mixer threads is we are stuck here 7450 Mutex::Autolock _l(mCblk->lock); 7451 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7452 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7453 mCblk->serverIndex = 0; 7454 mCblk->clientIndex = 0; 7455 return BAD_VALUE; 7456 } 7457 status_t status = NO_ERROR; 7458 while (mCblk->serverIndex < mCblk->clientIndex) { 7459 int reply; 7460 uint32_t rsize = sizeof(int); 7461 int *p = (int *)(mBuffer + mCblk->serverIndex); 7462 int size = *p++; 7463 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7464 ALOGW("command(): invalid parameter block size"); 7465 break; 7466 } 7467 effect_param_t *param = (effect_param_t *)p; 7468 if (param->psize == 0 || param->vsize == 0) { 7469 ALOGW("command(): null parameter or value size"); 7470 mCblk->serverIndex += size; 7471 continue; 7472 } 7473 uint32_t psize = sizeof(effect_param_t) + 7474 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7475 param->vsize; 7476 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7477 psize, 7478 p, 7479 &rsize, 7480 &reply); 7481 // stop at first error encountered 7482 if (ret != NO_ERROR) { 7483 status = ret; 7484 *(int *)pReplyData = reply; 7485 break; 7486 } else if (reply != NO_ERROR) { 7487 *(int *)pReplyData = reply; 7488 break; 7489 } 7490 mCblk->serverIndex += size; 7491 } 7492 mCblk->serverIndex = 0; 7493 mCblk->clientIndex = 0; 7494 return status; 7495 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7496 *(int *)pReplyData = NO_ERROR; 7497 return enable(); 7498 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7499 *(int *)pReplyData = NO_ERROR; 7500 return disable(); 7501 } 7502 7503 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7504} 7505 7506void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7507{ 7508 ALOGV("setControl %p control %d", this, hasControl); 7509 7510 mHasControl = hasControl; 7511 mEnabled = enabled; 7512 7513 if (signal && mEffectClient != 0) { 7514 mEffectClient->controlStatusChanged(hasControl); 7515 } 7516} 7517 7518void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7519 uint32_t cmdSize, 7520 void *pCmdData, 7521 uint32_t replySize, 7522 void *pReplyData) 7523{ 7524 if (mEffectClient != 0) { 7525 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7526 } 7527} 7528 7529 7530 7531void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7532{ 7533 if (mEffectClient != 0) { 7534 mEffectClient->enableStatusChanged(enabled); 7535 } 7536} 7537 7538status_t AudioFlinger::EffectHandle::onTransact( 7539 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7540{ 7541 return BnEffect::onTransact(code, data, reply, flags); 7542} 7543 7544 7545void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7546{ 7547 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7548 7549 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7550 (mClient == 0) ? getpid_cached : mClient->pid(), 7551 mPriority, 7552 mHasControl, 7553 !locked, 7554 mCblk ? mCblk->clientIndex : 0, 7555 mCblk ? mCblk->serverIndex : 0 7556 ); 7557 7558 if (locked) { 7559 mCblk->lock.unlock(); 7560 } 7561} 7562 7563#undef LOG_TAG 7564#define LOG_TAG "AudioFlinger::EffectChain" 7565 7566AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7567 int sessionId) 7568 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7569 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7570 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7571{ 7572 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7573 if (thread == NULL) { 7574 return; 7575 } 7576 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7577 thread->frameCount(); 7578} 7579 7580AudioFlinger::EffectChain::~EffectChain() 7581{ 7582 if (mOwnInBuffer) { 7583 delete mInBuffer; 7584 } 7585 7586} 7587 7588// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7589sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7590{ 7591 size_t size = mEffects.size(); 7592 7593 for (size_t i = 0; i < size; i++) { 7594 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7595 return mEffects[i]; 7596 } 7597 } 7598 return 0; 7599} 7600 7601// getEffectFromId_l() must be called with ThreadBase::mLock held 7602sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7603{ 7604 size_t size = mEffects.size(); 7605 7606 for (size_t i = 0; i < size; i++) { 7607 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7608 if (id == 0 || mEffects[i]->id() == id) { 7609 return mEffects[i]; 7610 } 7611 } 7612 return 0; 7613} 7614 7615// getEffectFromType_l() must be called with ThreadBase::mLock held 7616sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7617 const effect_uuid_t *type) 7618{ 7619 size_t size = mEffects.size(); 7620 7621 for (size_t i = 0; i < size; i++) { 7622 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7623 return mEffects[i]; 7624 } 7625 } 7626 return 0; 7627} 7628 7629// Must be called with EffectChain::mLock locked 7630void AudioFlinger::EffectChain::process_l() 7631{ 7632 sp<ThreadBase> thread = mThread.promote(); 7633 if (thread == 0) { 7634 ALOGW("process_l(): cannot promote mixer thread"); 7635 return; 7636 } 7637 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7638 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7639 // always process effects unless no more tracks are on the session and the effect tail 7640 // has been rendered 7641 bool doProcess = true; 7642 if (!isGlobalSession) { 7643 bool tracksOnSession = (trackCnt() != 0); 7644 7645 if (!tracksOnSession && mTailBufferCount == 0) { 7646 doProcess = false; 7647 } 7648 7649 if (activeTrackCnt() == 0) { 7650 // if no track is active and the effect tail has not been rendered, 7651 // the input buffer must be cleared here as the mixer process will not do it 7652 if (tracksOnSession || mTailBufferCount > 0) { 7653 size_t numSamples = thread->frameCount() * thread->channelCount(); 7654 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7655 if (mTailBufferCount > 0) { 7656 mTailBufferCount--; 7657 } 7658 } 7659 } 7660 } 7661 7662 size_t size = mEffects.size(); 7663 if (doProcess) { 7664 for (size_t i = 0; i < size; i++) { 7665 mEffects[i]->process(); 7666 } 7667 } 7668 for (size_t i = 0; i < size; i++) { 7669 mEffects[i]->updateState(); 7670 } 7671} 7672 7673// addEffect_l() must be called with PlaybackThread::mLock held 7674status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7675{ 7676 effect_descriptor_t desc = effect->desc(); 7677 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7678 7679 Mutex::Autolock _l(mLock); 7680 effect->setChain(this); 7681 sp<ThreadBase> thread = mThread.promote(); 7682 if (thread == 0) { 7683 return NO_INIT; 7684 } 7685 effect->setThread(thread); 7686 7687 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7688 // Auxiliary effects are inserted at the beginning of mEffects vector as 7689 // they are processed first and accumulated in chain input buffer 7690 mEffects.insertAt(effect, 0); 7691 7692 // the input buffer for auxiliary effect contains mono samples in 7693 // 32 bit format. This is to avoid saturation in AudoMixer 7694 // accumulation stage. Saturation is done in EffectModule::process() before 7695 // calling the process in effect engine 7696 size_t numSamples = thread->frameCount(); 7697 int32_t *buffer = new int32_t[numSamples]; 7698 memset(buffer, 0, numSamples * sizeof(int32_t)); 7699 effect->setInBuffer((int16_t *)buffer); 7700 // auxiliary effects output samples to chain input buffer for further processing 7701 // by insert effects 7702 effect->setOutBuffer(mInBuffer); 7703 } else { 7704 // Insert effects are inserted at the end of mEffects vector as they are processed 7705 // after track and auxiliary effects. 7706 // Insert effect order as a function of indicated preference: 7707 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7708 // another effect is present 7709 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7710 // last effect claiming first position 7711 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7712 // first effect claiming last position 7713 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7714 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7715 // already present 7716 7717 size_t size = mEffects.size(); 7718 size_t idx_insert = size; 7719 ssize_t idx_insert_first = -1; 7720 ssize_t idx_insert_last = -1; 7721 7722 for (size_t i = 0; i < size; i++) { 7723 effect_descriptor_t d = mEffects[i]->desc(); 7724 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7725 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7726 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7727 // check invalid effect chaining combinations 7728 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7729 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7730 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7731 return INVALID_OPERATION; 7732 } 7733 // remember position of first insert effect and by default 7734 // select this as insert position for new effect 7735 if (idx_insert == size) { 7736 idx_insert = i; 7737 } 7738 // remember position of last insert effect claiming 7739 // first position 7740 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7741 idx_insert_first = i; 7742 } 7743 // remember position of first insert effect claiming 7744 // last position 7745 if (iPref == EFFECT_FLAG_INSERT_LAST && 7746 idx_insert_last == -1) { 7747 idx_insert_last = i; 7748 } 7749 } 7750 } 7751 7752 // modify idx_insert from first position if needed 7753 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7754 if (idx_insert_last != -1) { 7755 idx_insert = idx_insert_last; 7756 } else { 7757 idx_insert = size; 7758 } 7759 } else { 7760 if (idx_insert_first != -1) { 7761 idx_insert = idx_insert_first + 1; 7762 } 7763 } 7764 7765 // always read samples from chain input buffer 7766 effect->setInBuffer(mInBuffer); 7767 7768 // if last effect in the chain, output samples to chain 7769 // output buffer, otherwise to chain input buffer 7770 if (idx_insert == size) { 7771 if (idx_insert != 0) { 7772 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7773 mEffects[idx_insert-1]->configure(); 7774 } 7775 effect->setOutBuffer(mOutBuffer); 7776 } else { 7777 effect->setOutBuffer(mInBuffer); 7778 } 7779 mEffects.insertAt(effect, idx_insert); 7780 7781 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7782 } 7783 effect->configure(); 7784 return NO_ERROR; 7785} 7786 7787// removeEffect_l() must be called with PlaybackThread::mLock held 7788size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7789{ 7790 Mutex::Autolock _l(mLock); 7791 size_t size = mEffects.size(); 7792 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7793 7794 for (size_t i = 0; i < size; i++) { 7795 if (effect == mEffects[i]) { 7796 // calling stop here will remove pre-processing effect from the audio HAL. 7797 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7798 // the middle of a read from audio HAL 7799 if (mEffects[i]->state() == EffectModule::ACTIVE || 7800 mEffects[i]->state() == EffectModule::STOPPING) { 7801 mEffects[i]->stop(); 7802 } 7803 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7804 delete[] effect->inBuffer(); 7805 } else { 7806 if (i == size - 1 && i != 0) { 7807 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7808 mEffects[i - 1]->configure(); 7809 } 7810 } 7811 mEffects.removeAt(i); 7812 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7813 break; 7814 } 7815 } 7816 7817 return mEffects.size(); 7818} 7819 7820// setDevice_l() must be called with PlaybackThread::mLock held 7821void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7822{ 7823 size_t size = mEffects.size(); 7824 for (size_t i = 0; i < size; i++) { 7825 mEffects[i]->setDevice(device); 7826 } 7827} 7828 7829// setMode_l() must be called with PlaybackThread::mLock held 7830void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7831{ 7832 size_t size = mEffects.size(); 7833 for (size_t i = 0; i < size; i++) { 7834 mEffects[i]->setMode(mode); 7835 } 7836} 7837 7838// setVolume_l() must be called with PlaybackThread::mLock held 7839bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7840{ 7841 uint32_t newLeft = *left; 7842 uint32_t newRight = *right; 7843 bool hasControl = false; 7844 int ctrlIdx = -1; 7845 size_t size = mEffects.size(); 7846 7847 // first update volume controller 7848 for (size_t i = size; i > 0; i--) { 7849 if (mEffects[i - 1]->isProcessEnabled() && 7850 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7851 ctrlIdx = i - 1; 7852 hasControl = true; 7853 break; 7854 } 7855 } 7856 7857 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7858 if (hasControl) { 7859 *left = mNewLeftVolume; 7860 *right = mNewRightVolume; 7861 } 7862 return hasControl; 7863 } 7864 7865 mVolumeCtrlIdx = ctrlIdx; 7866 mLeftVolume = newLeft; 7867 mRightVolume = newRight; 7868 7869 // second get volume update from volume controller 7870 if (ctrlIdx >= 0) { 7871 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7872 mNewLeftVolume = newLeft; 7873 mNewRightVolume = newRight; 7874 } 7875 // then indicate volume to all other effects in chain. 7876 // Pass altered volume to effects before volume controller 7877 // and requested volume to effects after controller 7878 uint32_t lVol = newLeft; 7879 uint32_t rVol = newRight; 7880 7881 for (size_t i = 0; i < size; i++) { 7882 if ((int)i == ctrlIdx) continue; 7883 // this also works for ctrlIdx == -1 when there is no volume controller 7884 if ((int)i > ctrlIdx) { 7885 lVol = *left; 7886 rVol = *right; 7887 } 7888 mEffects[i]->setVolume(&lVol, &rVol, false); 7889 } 7890 *left = newLeft; 7891 *right = newRight; 7892 7893 return hasControl; 7894} 7895 7896status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7897{ 7898 const size_t SIZE = 256; 7899 char buffer[SIZE]; 7900 String8 result; 7901 7902 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7903 result.append(buffer); 7904 7905 bool locked = tryLock(mLock); 7906 // failed to lock - AudioFlinger is probably deadlocked 7907 if (!locked) { 7908 result.append("\tCould not lock mutex:\n"); 7909 } 7910 7911 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7912 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7913 mEffects.size(), 7914 (uint32_t)mInBuffer, 7915 (uint32_t)mOutBuffer, 7916 mActiveTrackCnt); 7917 result.append(buffer); 7918 write(fd, result.string(), result.size()); 7919 7920 for (size_t i = 0; i < mEffects.size(); ++i) { 7921 sp<EffectModule> effect = mEffects[i]; 7922 if (effect != 0) { 7923 effect->dump(fd, args); 7924 } 7925 } 7926 7927 if (locked) { 7928 mLock.unlock(); 7929 } 7930 7931 return NO_ERROR; 7932} 7933 7934// must be called with ThreadBase::mLock held 7935void AudioFlinger::EffectChain::setEffectSuspended_l( 7936 const effect_uuid_t *type, bool suspend) 7937{ 7938 sp<SuspendedEffectDesc> desc; 7939 // use effect type UUID timelow as key as there is no real risk of identical 7940 // timeLow fields among effect type UUIDs. 7941 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7942 if (suspend) { 7943 if (index >= 0) { 7944 desc = mSuspendedEffects.valueAt(index); 7945 } else { 7946 desc = new SuspendedEffectDesc(); 7947 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7948 mSuspendedEffects.add(type->timeLow, desc); 7949 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7950 } 7951 if (desc->mRefCount++ == 0) { 7952 sp<EffectModule> effect = getEffectIfEnabled(type); 7953 if (effect != 0) { 7954 desc->mEffect = effect; 7955 effect->setSuspended(true); 7956 effect->setEnabled(false); 7957 } 7958 } 7959 } else { 7960 if (index < 0) { 7961 return; 7962 } 7963 desc = mSuspendedEffects.valueAt(index); 7964 if (desc->mRefCount <= 0) { 7965 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7966 desc->mRefCount = 1; 7967 } 7968 if (--desc->mRefCount == 0) { 7969 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7970 if (desc->mEffect != 0) { 7971 sp<EffectModule> effect = desc->mEffect.promote(); 7972 if (effect != 0) { 7973 effect->setSuspended(false); 7974 sp<EffectHandle> handle = effect->controlHandle(); 7975 if (handle != 0) { 7976 effect->setEnabled(handle->enabled()); 7977 } 7978 } 7979 desc->mEffect.clear(); 7980 } 7981 mSuspendedEffects.removeItemsAt(index); 7982 } 7983 } 7984} 7985 7986// must be called with ThreadBase::mLock held 7987void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7988{ 7989 sp<SuspendedEffectDesc> desc; 7990 7991 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7992 if (suspend) { 7993 if (index >= 0) { 7994 desc = mSuspendedEffects.valueAt(index); 7995 } else { 7996 desc = new SuspendedEffectDesc(); 7997 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7998 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7999 } 8000 if (desc->mRefCount++ == 0) { 8001 Vector< sp<EffectModule> > effects; 8002 getSuspendEligibleEffects(effects); 8003 for (size_t i = 0; i < effects.size(); i++) { 8004 setEffectSuspended_l(&effects[i]->desc().type, true); 8005 } 8006 } 8007 } else { 8008 if (index < 0) { 8009 return; 8010 } 8011 desc = mSuspendedEffects.valueAt(index); 8012 if (desc->mRefCount <= 0) { 8013 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8014 desc->mRefCount = 1; 8015 } 8016 if (--desc->mRefCount == 0) { 8017 Vector<const effect_uuid_t *> types; 8018 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8019 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8020 continue; 8021 } 8022 types.add(&mSuspendedEffects.valueAt(i)->mType); 8023 } 8024 for (size_t i = 0; i < types.size(); i++) { 8025 setEffectSuspended_l(types[i], false); 8026 } 8027 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8028 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8029 } 8030 } 8031} 8032 8033 8034// The volume effect is used for automated tests only 8035#ifndef OPENSL_ES_H_ 8036static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8037 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8038const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8039#endif //OPENSL_ES_H_ 8040 8041bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8042{ 8043 // auxiliary effects and visualizer are never suspended on output mix 8044 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8045 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8046 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8047 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8048 return false; 8049 } 8050 return true; 8051} 8052 8053void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8054{ 8055 effects.clear(); 8056 for (size_t i = 0; i < mEffects.size(); i++) { 8057 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8058 effects.add(mEffects[i]); 8059 } 8060 } 8061} 8062 8063sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8064 const effect_uuid_t *type) 8065{ 8066 sp<EffectModule> effect = getEffectFromType_l(type); 8067 return effect != 0 && effect->isEnabled() ? effect : 0; 8068} 8069 8070void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8071 bool enabled) 8072{ 8073 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8074 if (enabled) { 8075 if (index < 0) { 8076 // if the effect is not suspend check if all effects are suspended 8077 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8078 if (index < 0) { 8079 return; 8080 } 8081 if (!isEffectEligibleForSuspend(effect->desc())) { 8082 return; 8083 } 8084 setEffectSuspended_l(&effect->desc().type, enabled); 8085 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8086 if (index < 0) { 8087 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8088 return; 8089 } 8090 } 8091 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8092 effect->desc().type.timeLow); 8093 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8094 // if effect is requested to suspended but was not yet enabled, supend it now. 8095 if (desc->mEffect == 0) { 8096 desc->mEffect = effect; 8097 effect->setEnabled(false); 8098 effect->setSuspended(true); 8099 } 8100 } else { 8101 if (index < 0) { 8102 return; 8103 } 8104 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8105 effect->desc().type.timeLow); 8106 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8107 desc->mEffect.clear(); 8108 effect->setSuspended(false); 8109 } 8110} 8111 8112#undef LOG_TAG 8113#define LOG_TAG "AudioFlinger" 8114 8115// ---------------------------------------------------------------------------- 8116 8117status_t AudioFlinger::onTransact( 8118 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8119{ 8120 return BnAudioFlinger::onTransact(code, data, reply, flags); 8121} 8122 8123}; // namespace android 8124