AudioFlinger.cpp revision 1b094ee8f7fe7eca65bf3d2f983ba95eef6db93d
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 int16_t left, right, aux; 2320 // This additional clamping is needed in case chain->setVolume_l() overshot 2321 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2322 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2323 left = int16_t(v_clamped); 2324 v_clamped = (vr + (1 << 11)) >> 12; 2325 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2326 right = int16_t(v_clamped); 2327 2328 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2329 aux = int16_t(va); 2330 2331 // XXX: these things DON'T need to be done each time 2332 mAudioMixer->setBufferProvider(name, track); 2333 mAudioMixer->enable(name); 2334 2335 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2336 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2337 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2338 mAudioMixer->setParameter( 2339 name, 2340 AudioMixer::TRACK, 2341 AudioMixer::FORMAT, (void *)track->format()); 2342 mAudioMixer->setParameter( 2343 name, 2344 AudioMixer::TRACK, 2345 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2346 mAudioMixer->setParameter( 2347 name, 2348 AudioMixer::RESAMPLE, 2349 AudioMixer::SAMPLE_RATE, 2350 (void *)(cblk->sampleRate)); 2351 mAudioMixer->setParameter( 2352 name, 2353 AudioMixer::TRACK, 2354 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2355 mAudioMixer->setParameter( 2356 name, 2357 AudioMixer::TRACK, 2358 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2359 2360 // reset retry count 2361 track->mRetryCount = kMaxTrackRetries; 2362 // If one track is ready, set the mixer ready if: 2363 // - the mixer was not ready during previous round OR 2364 // - no other track is not ready 2365 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2366 mixerStatus != MIXER_TRACKS_ENABLED) { 2367 mixerStatus = MIXER_TRACKS_READY; 2368 } 2369 } else { 2370 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2371 if (track->isStopped()) { 2372 track->reset(); 2373 } 2374 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2375 // We have consumed all the buffers of this track. 2376 // Remove it from the list of active tracks. 2377 tracksToRemove->add(track); 2378 } else { 2379 // No buffers for this track. Give it a few chances to 2380 // fill a buffer, then remove it from active list. 2381 if (--(track->mRetryCount) <= 0) { 2382 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2383 tracksToRemove->add(track); 2384 // indicate to client process that the track was disabled because of underrun 2385 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2386 // If one track is not ready, mark the mixer also not ready if: 2387 // - the mixer was ready during previous round OR 2388 // - no other track is ready 2389 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2390 mixerStatus != MIXER_TRACKS_READY) { 2391 mixerStatus = MIXER_TRACKS_ENABLED; 2392 } 2393 } 2394 mAudioMixer->disable(name); 2395 } 2396 } 2397 2398 // remove all the tracks that need to be... 2399 count = tracksToRemove->size(); 2400 if (CC_UNLIKELY(count)) { 2401 for (size_t i=0 ; i<count ; i++) { 2402 const sp<Track>& track = tracksToRemove->itemAt(i); 2403 mActiveTracks.remove(track); 2404 if (track->mainBuffer() != mMixBuffer) { 2405 chain = getEffectChain_l(track->sessionId()); 2406 if (chain != 0) { 2407 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2408 chain->decActiveTrackCnt(); 2409 } 2410 } 2411 if (track->isTerminated()) { 2412 removeTrack_l(track); 2413 } 2414 } 2415 } 2416 2417 // mix buffer must be cleared if all tracks are connected to an 2418 // effect chain as in this case the mixer will not write to 2419 // mix buffer and track effects will accumulate into it 2420 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2421 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2422 } 2423 2424 mPrevMixerStatus = mixerStatus; 2425 return mixerStatus; 2426} 2427 2428void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2429{ 2430 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2431 this, streamType, mTracks.size()); 2432 Mutex::Autolock _l(mLock); 2433 2434 size_t size = mTracks.size(); 2435 for (size_t i = 0; i < size; i++) { 2436 sp<Track> t = mTracks[i]; 2437 if (t->streamType() == streamType) { 2438 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2439 t->mCblk->cv.signal(); 2440 } 2441 } 2442} 2443 2444void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2445{ 2446 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2447 this, streamType, valid); 2448 Mutex::Autolock _l(mLock); 2449 2450 mStreamTypes[streamType].valid = valid; 2451} 2452 2453// getTrackName_l() must be called with ThreadBase::mLock held 2454int AudioFlinger::MixerThread::getTrackName_l() 2455{ 2456 return mAudioMixer->getTrackName(); 2457} 2458 2459// deleteTrackName_l() must be called with ThreadBase::mLock held 2460void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2461{ 2462 ALOGV("remove track (%d) and delete from mixer", name); 2463 mAudioMixer->deleteTrackName(name); 2464} 2465 2466// checkForNewParameters_l() must be called with ThreadBase::mLock held 2467bool AudioFlinger::MixerThread::checkForNewParameters_l() 2468{ 2469 bool reconfig = false; 2470 2471 while (!mNewParameters.isEmpty()) { 2472 status_t status = NO_ERROR; 2473 String8 keyValuePair = mNewParameters[0]; 2474 AudioParameter param = AudioParameter(keyValuePair); 2475 int value; 2476 2477 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2478 reconfig = true; 2479 } 2480 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2481 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2482 status = BAD_VALUE; 2483 } else { 2484 reconfig = true; 2485 } 2486 } 2487 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2488 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2489 status = BAD_VALUE; 2490 } else { 2491 reconfig = true; 2492 } 2493 } 2494 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2495 // do not accept frame count changes if tracks are open as the track buffer 2496 // size depends on frame count and correct behavior would not be guaranteed 2497 // if frame count is changed after track creation 2498 if (!mTracks.isEmpty()) { 2499 status = INVALID_OPERATION; 2500 } else { 2501 reconfig = true; 2502 } 2503 } 2504 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2505 // when changing the audio output device, call addBatteryData to notify 2506 // the change 2507 if ((int)mDevice != value) { 2508 uint32_t params = 0; 2509 // check whether speaker is on 2510 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2511 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2512 } 2513 2514 int deviceWithoutSpeaker 2515 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2516 // check if any other device (except speaker) is on 2517 if (value & deviceWithoutSpeaker ) { 2518 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2519 } 2520 2521 if (params != 0) { 2522 addBatteryData(params); 2523 } 2524 } 2525 2526 // forward device change to effects that have requested to be 2527 // aware of attached audio device. 2528 mDevice = (uint32_t)value; 2529 for (size_t i = 0; i < mEffectChains.size(); i++) { 2530 mEffectChains[i]->setDevice_l(mDevice); 2531 } 2532 } 2533 2534 if (status == NO_ERROR) { 2535 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2536 keyValuePair.string()); 2537 if (!mStandby && status == INVALID_OPERATION) { 2538 mOutput->stream->common.standby(&mOutput->stream->common); 2539 mStandby = true; 2540 mBytesWritten = 0; 2541 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2542 keyValuePair.string()); 2543 } 2544 if (status == NO_ERROR && reconfig) { 2545 delete mAudioMixer; 2546 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2547 mAudioMixer = NULL; 2548 readOutputParameters(); 2549 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2550 for (size_t i = 0; i < mTracks.size() ; i++) { 2551 int name = getTrackName_l(); 2552 if (name < 0) break; 2553 mTracks[i]->mName = name; 2554 // limit track sample rate to 2 x new output sample rate 2555 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2556 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2557 } 2558 } 2559 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2560 } 2561 } 2562 2563 mNewParameters.removeAt(0); 2564 2565 mParamStatus = status; 2566 mParamCond.signal(); 2567 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2568 // already timed out waiting for the status and will never signal the condition. 2569 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2570 } 2571 return reconfig; 2572} 2573 2574status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2575{ 2576 const size_t SIZE = 256; 2577 char buffer[SIZE]; 2578 String8 result; 2579 2580 PlaybackThread::dumpInternals(fd, args); 2581 2582 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2583 result.append(buffer); 2584 write(fd, result.string(), result.size()); 2585 return NO_ERROR; 2586} 2587 2588uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2589{ 2590 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2591} 2592 2593uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2594{ 2595 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2596} 2597 2598// ---------------------------------------------------------------------------- 2599AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2600 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2601 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2602 // mLeftVolFloat, mRightVolFloat 2603 // mLeftVolShort, mRightVolShort 2604{ 2605} 2606 2607AudioFlinger::DirectOutputThread::~DirectOutputThread() 2608{ 2609} 2610 2611void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2612{ 2613 // Do not apply volume on compressed audio 2614 if (!audio_is_linear_pcm(mFormat)) { 2615 return; 2616 } 2617 2618 // convert to signed 16 bit before volume calculation 2619 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2620 size_t count = mFrameCount * mChannelCount; 2621 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2622 int16_t *dst = mMixBuffer + count-1; 2623 while(count--) { 2624 *dst-- = (int16_t)(*src--^0x80) << 8; 2625 } 2626 } 2627 2628 size_t frameCount = mFrameCount; 2629 int16_t *out = mMixBuffer; 2630 if (ramp) { 2631 if (mChannelCount == 1) { 2632 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2633 int32_t vlInc = d / (int32_t)frameCount; 2634 int32_t vl = ((int32_t)mLeftVolShort << 16); 2635 do { 2636 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2637 out++; 2638 vl += vlInc; 2639 } while (--frameCount); 2640 2641 } else { 2642 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2643 int32_t vlInc = d / (int32_t)frameCount; 2644 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2645 int32_t vrInc = d / (int32_t)frameCount; 2646 int32_t vl = ((int32_t)mLeftVolShort << 16); 2647 int32_t vr = ((int32_t)mRightVolShort << 16); 2648 do { 2649 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2650 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2651 out += 2; 2652 vl += vlInc; 2653 vr += vrInc; 2654 } while (--frameCount); 2655 } 2656 } else { 2657 if (mChannelCount == 1) { 2658 do { 2659 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2660 out++; 2661 } while (--frameCount); 2662 } else { 2663 do { 2664 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2665 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2666 out += 2; 2667 } while (--frameCount); 2668 } 2669 } 2670 2671 // convert back to unsigned 8 bit after volume calculation 2672 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2673 size_t count = mFrameCount * mChannelCount; 2674 int16_t *src = mMixBuffer; 2675 uint8_t *dst = (uint8_t *)mMixBuffer; 2676 while(count--) { 2677 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2678 } 2679 } 2680 2681 mLeftVolShort = leftVol; 2682 mRightVolShort = rightVol; 2683} 2684 2685bool AudioFlinger::DirectOutputThread::threadLoop() 2686{ 2687 mixer_state mixerStatus = MIXER_IDLE; 2688 sp<Track> trackToRemove; 2689 sp<Track> activeTrack; 2690 nsecs_t standbyTime = systemTime(); 2691 size_t mixBufferSize = mFrameCount*mFrameSize; 2692 uint32_t activeSleepTime = activeSleepTimeUs(); 2693 uint32_t idleSleepTime = idleSleepTimeUs(); 2694 uint32_t sleepTime = idleSleepTime; 2695 // use shorter standby delay as on normal output to release 2696 // hardware resources as soon as possible 2697 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2698 2699 acquireWakeLock(); 2700 2701 while (!exitPending()) 2702 { 2703 bool rampVolume; 2704 uint16_t leftVol; 2705 uint16_t rightVol; 2706 Vector< sp<EffectChain> > effectChains; 2707 2708 processConfigEvents(); 2709 2710 mixerStatus = MIXER_IDLE; 2711 2712 { // scope for the mLock 2713 2714 Mutex::Autolock _l(mLock); 2715 2716 if (checkForNewParameters_l()) { 2717 mixBufferSize = mFrameCount*mFrameSize; 2718 activeSleepTime = activeSleepTimeUs(); 2719 idleSleepTime = idleSleepTimeUs(); 2720 standbyDelay = microseconds(activeSleepTime*2); 2721 } 2722 2723 // put audio hardware into standby after short delay 2724 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2725 mSuspended)) { 2726 // wait until we have something to do... 2727 if (!mStandby) { 2728 ALOGV("Audio hardware entering standby, mixer %p", this); 2729 mOutput->stream->common.standby(&mOutput->stream->common); 2730 mStandby = true; 2731 mBytesWritten = 0; 2732 } 2733 2734 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2735 // we're about to wait, flush the binder command buffer 2736 IPCThreadState::self()->flushCommands(); 2737 2738 if (exitPending()) break; 2739 2740 releaseWakeLock_l(); 2741 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2742 mWaitWorkCV.wait(mLock); 2743 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2744 acquireWakeLock_l(); 2745 2746 if (!mMasterMute) { 2747 char value[PROPERTY_VALUE_MAX]; 2748 property_get("ro.audio.silent", value, "0"); 2749 if (atoi(value)) { 2750 ALOGD("Silence is golden"); 2751 setMasterMute_l(true); 2752 } 2753 } 2754 2755 standbyTime = systemTime() + standbyDelay; 2756 sleepTime = idleSleepTime; 2757 continue; 2758 } 2759 } 2760 2761 effectChains = mEffectChains; 2762 2763 // find out which tracks need to be processed 2764 if (mActiveTracks.size() != 0) { 2765 sp<Track> t = mActiveTracks[0].promote(); 2766 if (t == 0) continue; 2767 2768 Track* const track = t.get(); 2769 audio_track_cblk_t* cblk = track->cblk(); 2770 2771 // The first time a track is added we wait 2772 // for all its buffers to be filled before processing it 2773 if (cblk->framesReady() && track->isReady() && 2774 !track->isPaused() && !track->isTerminated()) 2775 { 2776 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2777 2778 if (track->mFillingUpStatus == Track::FS_FILLED) { 2779 track->mFillingUpStatus = Track::FS_ACTIVE; 2780 mLeftVolFloat = mRightVolFloat = 0; 2781 mLeftVolShort = mRightVolShort = 0; 2782 if (track->mState == TrackBase::RESUMING) { 2783 track->mState = TrackBase::ACTIVE; 2784 rampVolume = true; 2785 } 2786 } else if (cblk->server != 0) { 2787 // If the track is stopped before the first frame was mixed, 2788 // do not apply ramp 2789 rampVolume = true; 2790 } 2791 // compute volume for this track 2792 float left, right; 2793 if (track->isMuted() || mMasterMute || track->isPausing() || 2794 mStreamTypes[track->streamType()].mute) { 2795 left = right = 0; 2796 if (track->isPausing()) { 2797 track->setPaused(); 2798 } 2799 } else { 2800 float typeVolume = mStreamTypes[track->streamType()].volume; 2801 float v = mMasterVolume * typeVolume; 2802 uint32_t vlr = cblk->getVolumeLR(); 2803 float v_clamped = v * (vlr & 0xFFFF); 2804 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2805 left = v_clamped/MAX_GAIN; 2806 v_clamped = v * (vlr >> 16); 2807 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2808 right = v_clamped/MAX_GAIN; 2809 } 2810 2811 if (left != mLeftVolFloat || right != mRightVolFloat) { 2812 mLeftVolFloat = left; 2813 mRightVolFloat = right; 2814 2815 // If audio HAL implements volume control, 2816 // force software volume to nominal value 2817 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2818 left = 1.0f; 2819 right = 1.0f; 2820 } 2821 2822 // Convert volumes from float to 8.24 2823 uint32_t vl = (uint32_t)(left * (1 << 24)); 2824 uint32_t vr = (uint32_t)(right * (1 << 24)); 2825 2826 // Delegate volume control to effect in track effect chain if needed 2827 // only one effect chain can be present on DirectOutputThread, so if 2828 // there is one, the track is connected to it 2829 if (!effectChains.isEmpty()) { 2830 // Do not ramp volume if volume is controlled by effect 2831 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2832 rampVolume = false; 2833 } 2834 } 2835 2836 // Convert volumes from 8.24 to 4.12 format 2837 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2838 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2839 leftVol = (uint16_t)v_clamped; 2840 v_clamped = (vr + (1 << 11)) >> 12; 2841 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2842 rightVol = (uint16_t)v_clamped; 2843 } else { 2844 leftVol = mLeftVolShort; 2845 rightVol = mRightVolShort; 2846 rampVolume = false; 2847 } 2848 2849 // reset retry count 2850 track->mRetryCount = kMaxTrackRetriesDirect; 2851 activeTrack = t; 2852 mixerStatus = MIXER_TRACKS_READY; 2853 } else { 2854 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2855 if (track->isStopped()) { 2856 track->reset(); 2857 } 2858 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2859 // We have consumed all the buffers of this track. 2860 // Remove it from the list of active tracks. 2861 trackToRemove = track; 2862 } else { 2863 // No buffers for this track. Give it a few chances to 2864 // fill a buffer, then remove it from active list. 2865 if (--(track->mRetryCount) <= 0) { 2866 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2867 trackToRemove = track; 2868 } else { 2869 mixerStatus = MIXER_TRACKS_ENABLED; 2870 } 2871 } 2872 } 2873 } 2874 2875 // remove all the tracks that need to be... 2876 if (CC_UNLIKELY(trackToRemove != 0)) { 2877 mActiveTracks.remove(trackToRemove); 2878 if (!effectChains.isEmpty()) { 2879 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2880 trackToRemove->sessionId()); 2881 effectChains[0]->decActiveTrackCnt(); 2882 } 2883 if (trackToRemove->isTerminated()) { 2884 removeTrack_l(trackToRemove); 2885 } 2886 } 2887 2888 lockEffectChains_l(effectChains); 2889 } 2890 2891 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2892 AudioBufferProvider::Buffer buffer; 2893 size_t frameCount = mFrameCount; 2894 int8_t *curBuf = (int8_t *)mMixBuffer; 2895 // output audio to hardware 2896 while (frameCount) { 2897 buffer.frameCount = frameCount; 2898 activeTrack->getNextBuffer(&buffer, 2899 AudioBufferProvider::kInvalidPTS); 2900 if (CC_UNLIKELY(buffer.raw == NULL)) { 2901 memset(curBuf, 0, frameCount * mFrameSize); 2902 break; 2903 } 2904 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2905 frameCount -= buffer.frameCount; 2906 curBuf += buffer.frameCount * mFrameSize; 2907 activeTrack->releaseBuffer(&buffer); 2908 } 2909 sleepTime = 0; 2910 standbyTime = systemTime() + standbyDelay; 2911 } else { 2912 if (sleepTime == 0) { 2913 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2914 sleepTime = activeSleepTime; 2915 } else { 2916 sleepTime = idleSleepTime; 2917 } 2918 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2919 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2920 sleepTime = 0; 2921 } 2922 } 2923 2924 if (mSuspended) { 2925 sleepTime = suspendSleepTimeUs(); 2926 } 2927 // sleepTime == 0 means we must write to audio hardware 2928 if (sleepTime == 0) { 2929 if (mixerStatus == MIXER_TRACKS_READY) { 2930 applyVolume(leftVol, rightVol, rampVolume); 2931 } 2932 for (size_t i = 0; i < effectChains.size(); i ++) { 2933 effectChains[i]->process_l(); 2934 } 2935 unlockEffectChains(effectChains); 2936 2937 mLastWriteTime = systemTime(); 2938 mInWrite = true; 2939 mBytesWritten += mixBufferSize; 2940 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2941 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2942 mNumWrites++; 2943 mInWrite = false; 2944 mStandby = false; 2945 } else { 2946 unlockEffectChains(effectChains); 2947 usleep(sleepTime); 2948 } 2949 2950 // finally let go of removed track, without the lock held 2951 // since we can't guarantee the destructors won't acquire that 2952 // same lock. 2953 trackToRemove.clear(); 2954 activeTrack.clear(); 2955 2956 // Effect chains will be actually deleted here if they were removed from 2957 // mEffectChains list during mixing or effects processing 2958 effectChains.clear(); 2959 } 2960 2961 if (!mStandby) { 2962 mOutput->stream->common.standby(&mOutput->stream->common); 2963 } 2964 2965 releaseWakeLock(); 2966 2967 ALOGV("DirectOutputThread %p exiting", this); 2968 return false; 2969} 2970 2971// getTrackName_l() must be called with ThreadBase::mLock held 2972int AudioFlinger::DirectOutputThread::getTrackName_l() 2973{ 2974 return 0; 2975} 2976 2977// deleteTrackName_l() must be called with ThreadBase::mLock held 2978void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2979{ 2980} 2981 2982// checkForNewParameters_l() must be called with ThreadBase::mLock held 2983bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2984{ 2985 bool reconfig = false; 2986 2987 while (!mNewParameters.isEmpty()) { 2988 status_t status = NO_ERROR; 2989 String8 keyValuePair = mNewParameters[0]; 2990 AudioParameter param = AudioParameter(keyValuePair); 2991 int value; 2992 2993 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2994 // do not accept frame count changes if tracks are open as the track buffer 2995 // size depends on frame count and correct behavior would not be garantied 2996 // if frame count is changed after track creation 2997 if (!mTracks.isEmpty()) { 2998 status = INVALID_OPERATION; 2999 } else { 3000 reconfig = true; 3001 } 3002 } 3003 if (status == NO_ERROR) { 3004 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3005 keyValuePair.string()); 3006 if (!mStandby && status == INVALID_OPERATION) { 3007 mOutput->stream->common.standby(&mOutput->stream->common); 3008 mStandby = true; 3009 mBytesWritten = 0; 3010 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3011 keyValuePair.string()); 3012 } 3013 if (status == NO_ERROR && reconfig) { 3014 readOutputParameters(); 3015 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3016 } 3017 } 3018 3019 mNewParameters.removeAt(0); 3020 3021 mParamStatus = status; 3022 mParamCond.signal(); 3023 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3024 // already timed out waiting for the status and will never signal the condition. 3025 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3026 } 3027 return reconfig; 3028} 3029 3030uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3031{ 3032 uint32_t time; 3033 if (audio_is_linear_pcm(mFormat)) { 3034 time = PlaybackThread::activeSleepTimeUs(); 3035 } else { 3036 time = 10000; 3037 } 3038 return time; 3039} 3040 3041uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3042{ 3043 uint32_t time; 3044 if (audio_is_linear_pcm(mFormat)) { 3045 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3046 } else { 3047 time = 10000; 3048 } 3049 return time; 3050} 3051 3052uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3053{ 3054 uint32_t time; 3055 if (audio_is_linear_pcm(mFormat)) { 3056 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3057 } else { 3058 time = 10000; 3059 } 3060 return time; 3061} 3062 3063 3064// ---------------------------------------------------------------------------- 3065 3066AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3067 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3068 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3069 mWaitTimeMs(UINT_MAX) 3070{ 3071 addOutputTrack(mainThread); 3072} 3073 3074AudioFlinger::DuplicatingThread::~DuplicatingThread() 3075{ 3076 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3077 mOutputTracks[i]->destroy(); 3078 } 3079} 3080 3081bool AudioFlinger::DuplicatingThread::threadLoop() 3082{ 3083 Vector< sp<Track> > tracksToRemove; 3084 mixer_state mixerStatus = MIXER_IDLE; 3085 nsecs_t standbyTime = systemTime(); 3086 size_t mixBufferSize = mFrameCount*mFrameSize; 3087 SortedVector< sp<OutputTrack> > outputTracks; 3088 uint32_t writeFrames = 0; 3089 uint32_t activeSleepTime = activeSleepTimeUs(); 3090 uint32_t idleSleepTime = idleSleepTimeUs(); 3091 uint32_t sleepTime = idleSleepTime; 3092 Vector< sp<EffectChain> > effectChains; 3093 3094 acquireWakeLock(); 3095 3096 while (!exitPending()) 3097 { 3098 processConfigEvents(); 3099 3100 mixerStatus = MIXER_IDLE; 3101 { // scope for the mLock 3102 3103 Mutex::Autolock _l(mLock); 3104 3105 if (checkForNewParameters_l()) { 3106 mixBufferSize = mFrameCount*mFrameSize; 3107 updateWaitTime(); 3108 activeSleepTime = activeSleepTimeUs(); 3109 idleSleepTime = idleSleepTimeUs(); 3110 } 3111 3112 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3113 3114 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3115 outputTracks.add(mOutputTracks[i]); 3116 } 3117 3118 // put audio hardware into standby after short delay 3119 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3120 mSuspended)) { 3121 if (!mStandby) { 3122 for (size_t i = 0; i < outputTracks.size(); i++) { 3123 outputTracks[i]->stop(); 3124 } 3125 mStandby = true; 3126 mBytesWritten = 0; 3127 } 3128 3129 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3130 // we're about to wait, flush the binder command buffer 3131 IPCThreadState::self()->flushCommands(); 3132 outputTracks.clear(); 3133 3134 if (exitPending()) break; 3135 3136 releaseWakeLock_l(); 3137 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3138 mWaitWorkCV.wait(mLock); 3139 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3140 acquireWakeLock_l(); 3141 3142 mPrevMixerStatus = MIXER_IDLE; 3143 if (!mMasterMute) { 3144 char value[PROPERTY_VALUE_MAX]; 3145 property_get("ro.audio.silent", value, "0"); 3146 if (atoi(value)) { 3147 ALOGD("Silence is golden"); 3148 setMasterMute_l(true); 3149 } 3150 } 3151 3152 standbyTime = systemTime() + mStandbyTimeInNsecs; 3153 sleepTime = idleSleepTime; 3154 continue; 3155 } 3156 } 3157 3158 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3159 3160 // prevent any changes in effect chain list and in each effect chain 3161 // during mixing and effect process as the audio buffers could be deleted 3162 // or modified if an effect is created or deleted 3163 lockEffectChains_l(effectChains); 3164 } 3165 3166 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3167 // mix buffers... 3168 if (outputsReady(outputTracks)) { 3169 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3170 } else { 3171 memset(mMixBuffer, 0, mixBufferSize); 3172 } 3173 sleepTime = 0; 3174 writeFrames = mFrameCount; 3175 } else { 3176 if (sleepTime == 0) { 3177 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3178 sleepTime = activeSleepTime; 3179 } else { 3180 sleepTime = idleSleepTime; 3181 } 3182 } else if (mBytesWritten != 0) { 3183 // flush remaining overflow buffers in output tracks 3184 for (size_t i = 0; i < outputTracks.size(); i++) { 3185 if (outputTracks[i]->isActive()) { 3186 sleepTime = 0; 3187 writeFrames = 0; 3188 memset(mMixBuffer, 0, mixBufferSize); 3189 break; 3190 } 3191 } 3192 } 3193 } 3194 3195 if (mSuspended) { 3196 sleepTime = suspendSleepTimeUs(); 3197 } 3198 // sleepTime == 0 means we must write to audio hardware 3199 if (sleepTime == 0) { 3200 for (size_t i = 0; i < effectChains.size(); i ++) { 3201 effectChains[i]->process_l(); 3202 } 3203 // enable changes in effect chain 3204 unlockEffectChains(effectChains); 3205 3206 standbyTime = systemTime() + mStandbyTimeInNsecs; 3207 for (size_t i = 0; i < outputTracks.size(); i++) { 3208 outputTracks[i]->write(mMixBuffer, writeFrames); 3209 } 3210 mStandby = false; 3211 mBytesWritten += mixBufferSize; 3212 } else { 3213 // enable changes in effect chain 3214 unlockEffectChains(effectChains); 3215 usleep(sleepTime); 3216 } 3217 3218 // finally let go of all our tracks, without the lock held 3219 // since we can't guarantee the destructors won't acquire that 3220 // same lock. 3221 tracksToRemove.clear(); 3222 outputTracks.clear(); 3223 3224 // Effect chains will be actually deleted here if they were removed from 3225 // mEffectChains list during mixing or effects processing 3226 effectChains.clear(); 3227 } 3228 3229 releaseWakeLock(); 3230 3231 return false; 3232} 3233 3234void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3235{ 3236 // FIXME explain this formula 3237 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3238 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3239 this, 3240 mSampleRate, 3241 mFormat, 3242 mChannelMask, 3243 frameCount); 3244 if (outputTrack->cblk() != NULL) { 3245 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3246 mOutputTracks.add(outputTrack); 3247 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3248 updateWaitTime(); 3249 } 3250} 3251 3252void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3253{ 3254 Mutex::Autolock _l(mLock); 3255 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3256 if (mOutputTracks[i]->thread() == thread) { 3257 mOutputTracks[i]->destroy(); 3258 mOutputTracks.removeAt(i); 3259 updateWaitTime(); 3260 return; 3261 } 3262 } 3263 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3264} 3265 3266void AudioFlinger::DuplicatingThread::updateWaitTime() 3267{ 3268 mWaitTimeMs = UINT_MAX; 3269 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3270 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3271 if (strong != 0) { 3272 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3273 if (waitTimeMs < mWaitTimeMs) { 3274 mWaitTimeMs = waitTimeMs; 3275 } 3276 } 3277 } 3278} 3279 3280 3281bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3282{ 3283 for (size_t i = 0; i < outputTracks.size(); i++) { 3284 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3285 if (thread == 0) { 3286 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3287 return false; 3288 } 3289 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3290 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3291 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3292 return false; 3293 } 3294 } 3295 return true; 3296} 3297 3298uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3299{ 3300 return (mWaitTimeMs * 1000) / 2; 3301} 3302 3303// ---------------------------------------------------------------------------- 3304 3305// TrackBase constructor must be called with AudioFlinger::mLock held 3306AudioFlinger::ThreadBase::TrackBase::TrackBase( 3307 const wp<ThreadBase>& thread, 3308 const sp<Client>& client, 3309 uint32_t sampleRate, 3310 audio_format_t format, 3311 uint32_t channelMask, 3312 int frameCount, 3313 uint32_t flags, 3314 const sp<IMemory>& sharedBuffer, 3315 int sessionId) 3316 : RefBase(), 3317 mThread(thread), 3318 mClient(client), 3319 mCblk(NULL), 3320 // mBuffer 3321 // mBufferEnd 3322 mFrameCount(0), 3323 mState(IDLE), 3324 mFormat(format), 3325 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3326 mSessionId(sessionId) 3327 // mChannelCount 3328 // mChannelMask 3329{ 3330 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3331 3332 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3333 size_t size = sizeof(audio_track_cblk_t); 3334 uint8_t channelCount = popcount(channelMask); 3335 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3336 if (sharedBuffer == 0) { 3337 size += bufferSize; 3338 } 3339 3340 if (client != NULL) { 3341 mCblkMemory = client->heap()->allocate(size); 3342 if (mCblkMemory != 0) { 3343 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3344 if (mCblk != NULL) { // construct the shared structure in-place. 3345 new(mCblk) audio_track_cblk_t(); 3346 // clear all buffers 3347 mCblk->frameCount = frameCount; 3348 mCblk->sampleRate = sampleRate; 3349 mChannelCount = channelCount; 3350 mChannelMask = channelMask; 3351 if (sharedBuffer == 0) { 3352 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3353 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3354 // Force underrun condition to avoid false underrun callback until first data is 3355 // written to buffer (other flags are cleared) 3356 mCblk->flags = CBLK_UNDERRUN_ON; 3357 } else { 3358 mBuffer = sharedBuffer->pointer(); 3359 } 3360 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3361 } 3362 } else { 3363 ALOGE("not enough memory for AudioTrack size=%u", size); 3364 client->heap()->dump("AudioTrack"); 3365 return; 3366 } 3367 } else { 3368 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3369 // construct the shared structure in-place. 3370 new(mCblk) audio_track_cblk_t(); 3371 // clear all buffers 3372 mCblk->frameCount = frameCount; 3373 mCblk->sampleRate = sampleRate; 3374 mChannelCount = channelCount; 3375 mChannelMask = channelMask; 3376 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3377 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3378 // Force underrun condition to avoid false underrun callback until first data is 3379 // written to buffer (other flags are cleared) 3380 mCblk->flags = CBLK_UNDERRUN_ON; 3381 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3382 } 3383} 3384 3385AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3386{ 3387 if (mCblk != NULL) { 3388 if (mClient == 0) { 3389 delete mCblk; 3390 } else { 3391 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3392 } 3393 } 3394 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3395 if (mClient != 0) { 3396 // Client destructor must run with AudioFlinger mutex locked 3397 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3398 // If the client's reference count drops to zero, the associated destructor 3399 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3400 // relying on the automatic clear() at end of scope. 3401 mClient.clear(); 3402 } 3403} 3404 3405void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3406{ 3407 buffer->raw = NULL; 3408 mFrameCount = buffer->frameCount; 3409 step(); 3410 buffer->frameCount = 0; 3411} 3412 3413bool AudioFlinger::ThreadBase::TrackBase::step() { 3414 bool result; 3415 audio_track_cblk_t* cblk = this->cblk(); 3416 3417 result = cblk->stepServer(mFrameCount); 3418 if (!result) { 3419 ALOGV("stepServer failed acquiring cblk mutex"); 3420 mFlags |= STEPSERVER_FAILED; 3421 } 3422 return result; 3423} 3424 3425void AudioFlinger::ThreadBase::TrackBase::reset() { 3426 audio_track_cblk_t* cblk = this->cblk(); 3427 3428 cblk->user = 0; 3429 cblk->server = 0; 3430 cblk->userBase = 0; 3431 cblk->serverBase = 0; 3432 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3433 ALOGV("TrackBase::reset"); 3434} 3435 3436int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3437 return (int)mCblk->sampleRate; 3438} 3439 3440void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3441 audio_track_cblk_t* cblk = this->cblk(); 3442 size_t frameSize = cblk->frameSize; 3443 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3444 int8_t *bufferEnd = bufferStart + frames * frameSize; 3445 3446 // Check validity of returned pointer in case the track control block would have been corrupted. 3447 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3448 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3449 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3450 server %d, serverBase %d, user %d, userBase %d", 3451 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3452 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3453 return NULL; 3454 } 3455 3456 return bufferStart; 3457} 3458 3459// ---------------------------------------------------------------------------- 3460 3461// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3462AudioFlinger::PlaybackThread::Track::Track( 3463 const wp<ThreadBase>& thread, 3464 const sp<Client>& client, 3465 audio_stream_type_t streamType, 3466 uint32_t sampleRate, 3467 audio_format_t format, 3468 uint32_t channelMask, 3469 int frameCount, 3470 const sp<IMemory>& sharedBuffer, 3471 int sessionId) 3472 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3473 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3474 mAuxEffectId(0), mHasVolumeController(false) 3475{ 3476 if (mCblk != NULL) { 3477 sp<ThreadBase> baseThread = thread.promote(); 3478 if (baseThread != 0) { 3479 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3480 mName = playbackThread->getTrackName_l(); 3481 mMainBuffer = playbackThread->mixBuffer(); 3482 } 3483 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3484 if (mName < 0) { 3485 ALOGE("no more track names available"); 3486 } 3487 mStreamType = streamType; 3488 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3489 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3490 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3491 } 3492} 3493 3494AudioFlinger::PlaybackThread::Track::~Track() 3495{ 3496 ALOGV("PlaybackThread::Track destructor"); 3497 sp<ThreadBase> thread = mThread.promote(); 3498 if (thread != 0) { 3499 Mutex::Autolock _l(thread->mLock); 3500 mState = TERMINATED; 3501 } 3502} 3503 3504void AudioFlinger::PlaybackThread::Track::destroy() 3505{ 3506 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3507 // by removing it from mTracks vector, so there is a risk that this Tracks's 3508 // destructor is called. As the destructor needs to lock mLock, 3509 // we must acquire a strong reference on this Track before locking mLock 3510 // here so that the destructor is called only when exiting this function. 3511 // On the other hand, as long as Track::destroy() is only called by 3512 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3513 // this Track with its member mTrack. 3514 sp<Track> keep(this); 3515 { // scope for mLock 3516 sp<ThreadBase> thread = mThread.promote(); 3517 if (thread != 0) { 3518 if (!isOutputTrack()) { 3519 if (mState == ACTIVE || mState == RESUMING) { 3520 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3521 3522 // to track the speaker usage 3523 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3524 } 3525 AudioSystem::releaseOutput(thread->id()); 3526 } 3527 Mutex::Autolock _l(thread->mLock); 3528 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3529 playbackThread->destroyTrack_l(this); 3530 } 3531 } 3532} 3533 3534void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3535{ 3536 uint32_t vlr = mCblk->getVolumeLR(); 3537 snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 3538 mName - AudioMixer::TRACK0, 3539 (mClient == 0) ? getpid_cached : mClient->pid(), 3540 mStreamType, 3541 mFormat, 3542 mChannelMask, 3543 mSessionId, 3544 mFrameCount, 3545 mState, 3546 mMute, 3547 mFillingUpStatus, 3548 mCblk->sampleRate, 3549 vlr & 0xFFFF, 3550 vlr >> 16, 3551 mCblk->server, 3552 mCblk->user, 3553 (int)mMainBuffer, 3554 (int)mAuxBuffer); 3555} 3556 3557status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3558 AudioBufferProvider::Buffer* buffer, int64_t pts) 3559{ 3560 audio_track_cblk_t* cblk = this->cblk(); 3561 uint32_t framesReady; 3562 uint32_t framesReq = buffer->frameCount; 3563 3564 // Check if last stepServer failed, try to step now 3565 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3566 if (!step()) goto getNextBuffer_exit; 3567 ALOGV("stepServer recovered"); 3568 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3569 } 3570 3571 framesReady = cblk->framesReady(); 3572 3573 if (CC_LIKELY(framesReady)) { 3574 uint32_t s = cblk->server; 3575 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3576 3577 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3578 if (framesReq > framesReady) { 3579 framesReq = framesReady; 3580 } 3581 if (s + framesReq > bufferEnd) { 3582 framesReq = bufferEnd - s; 3583 } 3584 3585 buffer->raw = getBuffer(s, framesReq); 3586 if (buffer->raw == NULL) goto getNextBuffer_exit; 3587 3588 buffer->frameCount = framesReq; 3589 return NO_ERROR; 3590 } 3591 3592getNextBuffer_exit: 3593 buffer->raw = NULL; 3594 buffer->frameCount = 0; 3595 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3596 return NOT_ENOUGH_DATA; 3597} 3598 3599uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3600 return mCblk->framesReady(); 3601} 3602 3603bool AudioFlinger::PlaybackThread::Track::isReady() const { 3604 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3605 3606 if (framesReady() >= mCblk->frameCount || 3607 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3608 mFillingUpStatus = FS_FILLED; 3609 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3610 return true; 3611 } 3612 return false; 3613} 3614 3615status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3616{ 3617 status_t status = NO_ERROR; 3618 ALOGV("start(%d), calling pid %d session %d tid %d", 3619 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3620 sp<ThreadBase> thread = mThread.promote(); 3621 if (thread != 0) { 3622 Mutex::Autolock _l(thread->mLock); 3623 track_state state = mState; 3624 // here the track could be either new, or restarted 3625 // in both cases "unstop" the track 3626 if (mState == PAUSED) { 3627 mState = TrackBase::RESUMING; 3628 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3629 } else { 3630 mState = TrackBase::ACTIVE; 3631 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3632 } 3633 3634 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3635 thread->mLock.unlock(); 3636 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3637 thread->mLock.lock(); 3638 3639 // to track the speaker usage 3640 if (status == NO_ERROR) { 3641 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3642 } 3643 } 3644 if (status == NO_ERROR) { 3645 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3646 playbackThread->addTrack_l(this); 3647 } else { 3648 mState = state; 3649 } 3650 } else { 3651 status = BAD_VALUE; 3652 } 3653 return status; 3654} 3655 3656void AudioFlinger::PlaybackThread::Track::stop() 3657{ 3658 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3659 sp<ThreadBase> thread = mThread.promote(); 3660 if (thread != 0) { 3661 Mutex::Autolock _l(thread->mLock); 3662 track_state state = mState; 3663 if (mState > STOPPED) { 3664 mState = STOPPED; 3665 // If the track is not active (PAUSED and buffers full), flush buffers 3666 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3667 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3668 reset(); 3669 } 3670 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3671 } 3672 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3673 thread->mLock.unlock(); 3674 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3675 thread->mLock.lock(); 3676 3677 // to track the speaker usage 3678 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3679 } 3680 } 3681} 3682 3683void AudioFlinger::PlaybackThread::Track::pause() 3684{ 3685 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3686 sp<ThreadBase> thread = mThread.promote(); 3687 if (thread != 0) { 3688 Mutex::Autolock _l(thread->mLock); 3689 if (mState == ACTIVE || mState == RESUMING) { 3690 mState = PAUSING; 3691 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3692 if (!isOutputTrack()) { 3693 thread->mLock.unlock(); 3694 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3695 thread->mLock.lock(); 3696 3697 // to track the speaker usage 3698 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3699 } 3700 } 3701 } 3702} 3703 3704void AudioFlinger::PlaybackThread::Track::flush() 3705{ 3706 ALOGV("flush(%d)", mName); 3707 sp<ThreadBase> thread = mThread.promote(); 3708 if (thread != 0) { 3709 Mutex::Autolock _l(thread->mLock); 3710 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3711 return; 3712 } 3713 // No point remaining in PAUSED state after a flush => go to 3714 // STOPPED state 3715 mState = STOPPED; 3716 3717 // do not reset the track if it is still in the process of being stopped or paused. 3718 // this will be done by prepareTracks_l() when the track is stopped. 3719 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3720 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3721 reset(); 3722 } 3723 } 3724} 3725 3726void AudioFlinger::PlaybackThread::Track::reset() 3727{ 3728 // Do not reset twice to avoid discarding data written just after a flush and before 3729 // the audioflinger thread detects the track is stopped. 3730 if (!mResetDone) { 3731 TrackBase::reset(); 3732 // Force underrun condition to avoid false underrun callback until first data is 3733 // written to buffer 3734 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3735 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3736 mFillingUpStatus = FS_FILLING; 3737 mResetDone = true; 3738 } 3739} 3740 3741void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3742{ 3743 mMute = muted; 3744} 3745 3746status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3747{ 3748 status_t status = DEAD_OBJECT; 3749 sp<ThreadBase> thread = mThread.promote(); 3750 if (thread != 0) { 3751 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3752 status = playbackThread->attachAuxEffect(this, EffectId); 3753 } 3754 return status; 3755} 3756 3757void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3758{ 3759 mAuxEffectId = EffectId; 3760 mAuxBuffer = buffer; 3761} 3762 3763// timed audio tracks 3764 3765sp<AudioFlinger::PlaybackThread::TimedTrack> 3766AudioFlinger::PlaybackThread::TimedTrack::create( 3767 const wp<ThreadBase>& thread, 3768 const sp<Client>& client, 3769 audio_stream_type_t streamType, 3770 uint32_t sampleRate, 3771 audio_format_t format, 3772 uint32_t channelMask, 3773 int frameCount, 3774 const sp<IMemory>& sharedBuffer, 3775 int sessionId) { 3776 if (!client->reserveTimedTrack()) 3777 return NULL; 3778 3779 sp<TimedTrack> track = new TimedTrack( 3780 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3781 sharedBuffer, sessionId); 3782 3783 if (track == NULL) { 3784 client->releaseTimedTrack(); 3785 return NULL; 3786 } 3787 3788 return track; 3789} 3790 3791AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3792 const wp<ThreadBase>& thread, 3793 const sp<Client>& client, 3794 audio_stream_type_t streamType, 3795 uint32_t sampleRate, 3796 audio_format_t format, 3797 uint32_t channelMask, 3798 int frameCount, 3799 const sp<IMemory>& sharedBuffer, 3800 int sessionId) 3801 : Track(thread, client, streamType, sampleRate, format, channelMask, 3802 frameCount, sharedBuffer, sessionId), 3803 mTimedSilenceBuffer(NULL), 3804 mTimedSilenceBufferSize(0), 3805 mTimedAudioOutputOnTime(false), 3806 mMediaTimeTransformValid(false) 3807{ 3808 LocalClock lc; 3809 mLocalTimeFreq = lc.getLocalFreq(); 3810 3811 mLocalTimeToSampleTransform.a_zero = 0; 3812 mLocalTimeToSampleTransform.b_zero = 0; 3813 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3814 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3815 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3816 &mLocalTimeToSampleTransform.a_to_b_denom); 3817} 3818 3819AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3820 mClient->releaseTimedTrack(); 3821 delete [] mTimedSilenceBuffer; 3822} 3823 3824status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3825 size_t size, sp<IMemory>* buffer) { 3826 3827 Mutex::Autolock _l(mTimedBufferQueueLock); 3828 3829 trimTimedBufferQueue_l(); 3830 3831 // lazily initialize the shared memory heap for timed buffers 3832 if (mTimedMemoryDealer == NULL) { 3833 const int kTimedBufferHeapSize = 512 << 10; 3834 3835 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3836 "AudioFlingerTimed"); 3837 if (mTimedMemoryDealer == NULL) 3838 return NO_MEMORY; 3839 } 3840 3841 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3842 if (newBuffer == NULL) { 3843 newBuffer = mTimedMemoryDealer->allocate(size); 3844 if (newBuffer == NULL) 3845 return NO_MEMORY; 3846 } 3847 3848 *buffer = newBuffer; 3849 return NO_ERROR; 3850} 3851 3852// caller must hold mTimedBufferQueueLock 3853void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3854 int64_t mediaTimeNow; 3855 { 3856 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3857 if (!mMediaTimeTransformValid) 3858 return; 3859 3860 int64_t targetTimeNow; 3861 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3862 ? mCCHelper.getCommonTime(&targetTimeNow) 3863 : mCCHelper.getLocalTime(&targetTimeNow); 3864 3865 if (OK != res) 3866 return; 3867 3868 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3869 &mediaTimeNow)) { 3870 return; 3871 } 3872 } 3873 3874 size_t trimIndex; 3875 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3876 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3877 break; 3878 } 3879 3880 if (trimIndex) { 3881 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3882 } 3883} 3884 3885status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3886 const sp<IMemory>& buffer, int64_t pts) { 3887 3888 { 3889 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3890 if (!mMediaTimeTransformValid) 3891 return INVALID_OPERATION; 3892 } 3893 3894 Mutex::Autolock _l(mTimedBufferQueueLock); 3895 3896 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3897 3898 return NO_ERROR; 3899} 3900 3901status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3902 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3903 3904 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3905 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3906 target); 3907 3908 if (!(target == TimedAudioTrack::LOCAL_TIME || 3909 target == TimedAudioTrack::COMMON_TIME)) { 3910 return BAD_VALUE; 3911 } 3912 3913 Mutex::Autolock lock(mMediaTimeTransformLock); 3914 mMediaTimeTransform = xform; 3915 mMediaTimeTransformTarget = target; 3916 mMediaTimeTransformValid = true; 3917 3918 return NO_ERROR; 3919} 3920 3921#define min(a, b) ((a) < (b) ? (a) : (b)) 3922 3923// implementation of getNextBuffer for tracks whose buffers have timestamps 3924status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3925 AudioBufferProvider::Buffer* buffer, int64_t pts) 3926{ 3927 if (pts == AudioBufferProvider::kInvalidPTS) { 3928 buffer->raw = 0; 3929 buffer->frameCount = 0; 3930 return INVALID_OPERATION; 3931 } 3932 3933 // get ahold of the output stream that these samples will be written to 3934 sp<ThreadBase> thread = mThread.promote(); 3935 if (thread == NULL) { 3936 buffer->raw = 0; 3937 buffer->frameCount = 0; 3938 return INVALID_OPERATION; 3939 } 3940 PlaybackThread* playbackThread = static_cast<PlaybackThread*>(thread.get()); 3941 3942 Mutex::Autolock _l(mTimedBufferQueueLock); 3943 3944 while (true) { 3945 3946 // if we have no timed buffers, then fail 3947 if (mTimedBufferQueue.isEmpty()) { 3948 buffer->raw = 0; 3949 buffer->frameCount = 0; 3950 return NOT_ENOUGH_DATA; 3951 } 3952 3953 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3954 3955 // calculate the PTS of the head of the timed buffer queue expressed in 3956 // local time 3957 int64_t headLocalPTS; 3958 { 3959 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3960 3961 assert(mMediaTimeTransformValid); 3962 3963 if (mMediaTimeTransform.a_to_b_denom == 0) { 3964 // the transform represents a pause, so yield silence 3965 timedYieldSilence(buffer->frameCount, buffer); 3966 return NO_ERROR; 3967 } 3968 3969 int64_t transformedPTS; 3970 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3971 &transformedPTS)) { 3972 // the transform failed. this shouldn't happen, but if it does 3973 // then just drop this buffer 3974 ALOGW("timedGetNextBuffer transform failed"); 3975 buffer->raw = 0; 3976 buffer->frameCount = 0; 3977 mTimedBufferQueue.removeAt(0); 3978 return NO_ERROR; 3979 } 3980 3981 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3982 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3983 &headLocalPTS)) { 3984 buffer->raw = 0; 3985 buffer->frameCount = 0; 3986 return INVALID_OPERATION; 3987 } 3988 } else { 3989 headLocalPTS = transformedPTS; 3990 } 3991 } 3992 3993 // adjust the head buffer's PTS to reflect the portion of the head buffer 3994 // that has already been consumed 3995 int64_t effectivePTS = headLocalPTS + 3996 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3997 3998 // Calculate the delta in samples between the head of the input buffer 3999 // queue and the start of the next output buffer that will be written. 4000 // If the transformation fails because of over or underflow, it means 4001 // that the sample's position in the output stream is so far out of 4002 // whack that it should just be dropped. 4003 int64_t sampleDelta; 4004 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4005 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4006 mTimedBufferQueue.removeAt(0); 4007 continue; 4008 } 4009 if (!mLocalTimeToSampleTransform.doForwardTransform( 4010 (effectivePTS - pts) << 32, &sampleDelta)) { 4011 ALOGV("*** too late during sample rate transform: dropped buffer"); 4012 mTimedBufferQueue.removeAt(0); 4013 continue; 4014 } 4015 4016 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4017 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4018 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4019 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4020 4021 // if the delta between the ideal placement for the next input sample and 4022 // the current output position is within this threshold, then we will 4023 // concatenate the next input samples to the previous output 4024 const int64_t kSampleContinuityThreshold = 4025 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4026 4027 // if this is the first buffer of audio that we're emitting from this track 4028 // then it should be almost exactly on time. 4029 const int64_t kSampleStartupThreshold = 1LL << 32; 4030 4031 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4032 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4033 // the next input is close enough to being on time, so concatenate it 4034 // with the last output 4035 timedYieldSamples(buffer); 4036 4037 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4038 return NO_ERROR; 4039 } else if (sampleDelta > 0) { 4040 // the gap between the current output position and the proper start of 4041 // the next input sample is too big, so fill it with silence 4042 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4043 4044 timedYieldSilence(framesUntilNextInput, buffer); 4045 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4046 return NO_ERROR; 4047 } else { 4048 // the next input sample is late 4049 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4050 size_t onTimeSamplePosition = 4051 head.position() + lateFrames * mCblk->frameSize; 4052 4053 if (onTimeSamplePosition > head.buffer()->size()) { 4054 // all the remaining samples in the head are too late, so 4055 // drop it and move on 4056 ALOGV("*** too late: dropped buffer"); 4057 mTimedBufferQueue.removeAt(0); 4058 continue; 4059 } else { 4060 // skip over the late samples 4061 head.setPosition(onTimeSamplePosition); 4062 4063 // yield the available samples 4064 timedYieldSamples(buffer); 4065 4066 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4067 return NO_ERROR; 4068 } 4069 } 4070 } 4071} 4072 4073// Yield samples from the timed buffer queue head up to the given output 4074// buffer's capacity. 4075// 4076// Caller must hold mTimedBufferQueueLock 4077void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4078 AudioBufferProvider::Buffer* buffer) { 4079 4080 const TimedBuffer& head = mTimedBufferQueue[0]; 4081 4082 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4083 head.position()); 4084 4085 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4086 mCblk->frameSize); 4087 size_t framesRequested = buffer->frameCount; 4088 buffer->frameCount = min(framesLeftInHead, framesRequested); 4089 4090 mTimedAudioOutputOnTime = true; 4091} 4092 4093// Yield samples of silence up to the given output buffer's capacity 4094// 4095// Caller must hold mTimedBufferQueueLock 4096void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4097 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4098 4099 // lazily allocate a buffer filled with silence 4100 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4101 delete [] mTimedSilenceBuffer; 4102 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4103 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4104 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4105 } 4106 4107 buffer->raw = mTimedSilenceBuffer; 4108 size_t framesRequested = buffer->frameCount; 4109 buffer->frameCount = min(numFrames, framesRequested); 4110 4111 mTimedAudioOutputOnTime = false; 4112} 4113 4114void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4115 AudioBufferProvider::Buffer* buffer) { 4116 4117 Mutex::Autolock _l(mTimedBufferQueueLock); 4118 4119 // If the buffer which was just released is part of the buffer at the head 4120 // of the queue, be sure to update the amt of the buffer which has been 4121 // consumed. If the buffer being returned is not part of the head of the 4122 // queue, its either because the buffer is part of the silence buffer, or 4123 // because the head of the timed queue was trimmed after the mixer called 4124 // getNextBuffer but before the mixer called releaseBuffer. 4125 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4126 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4127 4128 void* start = head.buffer()->pointer(); 4129 void* end = head.buffer()->pointer() + head.buffer()->size(); 4130 4131 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4132 head.setPosition(head.position() + 4133 (buffer->frameCount * mCblk->frameSize)); 4134 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4135 mTimedBufferQueue.removeAt(0); 4136 } 4137 } 4138 } 4139 4140 buffer->raw = 0; 4141 buffer->frameCount = 0; 4142} 4143 4144uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4145 Mutex::Autolock _l(mTimedBufferQueueLock); 4146 4147 uint32_t frames = 0; 4148 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4149 const TimedBuffer& tb = mTimedBufferQueue[i]; 4150 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4151 } 4152 4153 return frames; 4154} 4155 4156AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4157 : mPTS(0), mPosition(0) {} 4158 4159AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4160 const sp<IMemory>& buffer, int64_t pts) 4161 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4162 4163// ---------------------------------------------------------------------------- 4164 4165// RecordTrack constructor must be called with AudioFlinger::mLock held 4166AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4167 const wp<ThreadBase>& thread, 4168 const sp<Client>& client, 4169 uint32_t sampleRate, 4170 audio_format_t format, 4171 uint32_t channelMask, 4172 int frameCount, 4173 uint32_t flags, 4174 int sessionId) 4175 : TrackBase(thread, client, sampleRate, format, 4176 channelMask, frameCount, flags, 0, sessionId), 4177 mOverflow(false) 4178{ 4179 if (mCblk != NULL) { 4180 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4181 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4182 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4183 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4184 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4185 } else { 4186 mCblk->frameSize = sizeof(int8_t); 4187 } 4188 } 4189} 4190 4191AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4192{ 4193 sp<ThreadBase> thread = mThread.promote(); 4194 if (thread != 0) { 4195 AudioSystem::releaseInput(thread->id()); 4196 } 4197} 4198 4199status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4200{ 4201 audio_track_cblk_t* cblk = this->cblk(); 4202 uint32_t framesAvail; 4203 uint32_t framesReq = buffer->frameCount; 4204 4205 // Check if last stepServer failed, try to step now 4206 if (mFlags & TrackBase::STEPSERVER_FAILED) { 4207 if (!step()) goto getNextBuffer_exit; 4208 ALOGV("stepServer recovered"); 4209 mFlags &= ~TrackBase::STEPSERVER_FAILED; 4210 } 4211 4212 framesAvail = cblk->framesAvailable_l(); 4213 4214 if (CC_LIKELY(framesAvail)) { 4215 uint32_t s = cblk->server; 4216 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4217 4218 if (framesReq > framesAvail) { 4219 framesReq = framesAvail; 4220 } 4221 if (s + framesReq > bufferEnd) { 4222 framesReq = bufferEnd - s; 4223 } 4224 4225 buffer->raw = getBuffer(s, framesReq); 4226 if (buffer->raw == NULL) goto getNextBuffer_exit; 4227 4228 buffer->frameCount = framesReq; 4229 return NO_ERROR; 4230 } 4231 4232getNextBuffer_exit: 4233 buffer->raw = NULL; 4234 buffer->frameCount = 0; 4235 return NOT_ENOUGH_DATA; 4236} 4237 4238status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4239{ 4240 sp<ThreadBase> thread = mThread.promote(); 4241 if (thread != 0) { 4242 RecordThread *recordThread = (RecordThread *)thread.get(); 4243 return recordThread->start(this, tid); 4244 } else { 4245 return BAD_VALUE; 4246 } 4247} 4248 4249void AudioFlinger::RecordThread::RecordTrack::stop() 4250{ 4251 sp<ThreadBase> thread = mThread.promote(); 4252 if (thread != 0) { 4253 RecordThread *recordThread = (RecordThread *)thread.get(); 4254 recordThread->stop(this); 4255 TrackBase::reset(); 4256 // Force overerrun condition to avoid false overrun callback until first data is 4257 // read from buffer 4258 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4259 } 4260} 4261 4262void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4263{ 4264 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4265 (mClient == 0) ? getpid_cached : mClient->pid(), 4266 mFormat, 4267 mChannelMask, 4268 mSessionId, 4269 mFrameCount, 4270 mState, 4271 mCblk->sampleRate, 4272 mCblk->server, 4273 mCblk->user); 4274} 4275 4276 4277// ---------------------------------------------------------------------------- 4278 4279AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4280 const wp<ThreadBase>& thread, 4281 DuplicatingThread *sourceThread, 4282 uint32_t sampleRate, 4283 audio_format_t format, 4284 uint32_t channelMask, 4285 int frameCount) 4286 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4287 mActive(false), mSourceThread(sourceThread) 4288{ 4289 4290 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 4291 if (mCblk != NULL) { 4292 mCblk->flags |= CBLK_DIRECTION_OUT; 4293 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4294 mOutBuffer.frameCount = 0; 4295 playbackThread->mTracks.add(this); 4296 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4297 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4298 mCblk, mBuffer, mCblk->buffers, 4299 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4300 } else { 4301 ALOGW("Error creating output track on thread %p", playbackThread); 4302 } 4303} 4304 4305AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4306{ 4307 clearBufferQueue(); 4308} 4309 4310status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4311{ 4312 status_t status = Track::start(tid); 4313 if (status != NO_ERROR) { 4314 return status; 4315 } 4316 4317 mActive = true; 4318 mRetryCount = 127; 4319 return status; 4320} 4321 4322void AudioFlinger::PlaybackThread::OutputTrack::stop() 4323{ 4324 Track::stop(); 4325 clearBufferQueue(); 4326 mOutBuffer.frameCount = 0; 4327 mActive = false; 4328} 4329 4330bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4331{ 4332 Buffer *pInBuffer; 4333 Buffer inBuffer; 4334 uint32_t channelCount = mChannelCount; 4335 bool outputBufferFull = false; 4336 inBuffer.frameCount = frames; 4337 inBuffer.i16 = data; 4338 4339 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4340 4341 if (!mActive && frames != 0) { 4342 start(0); 4343 sp<ThreadBase> thread = mThread.promote(); 4344 if (thread != 0) { 4345 MixerThread *mixerThread = (MixerThread *)thread.get(); 4346 if (mCblk->frameCount > frames){ 4347 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4348 uint32_t startFrames = (mCblk->frameCount - frames); 4349 pInBuffer = new Buffer; 4350 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4351 pInBuffer->frameCount = startFrames; 4352 pInBuffer->i16 = pInBuffer->mBuffer; 4353 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4354 mBufferQueue.add(pInBuffer); 4355 } else { 4356 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4357 } 4358 } 4359 } 4360 } 4361 4362 while (waitTimeLeftMs) { 4363 // First write pending buffers, then new data 4364 if (mBufferQueue.size()) { 4365 pInBuffer = mBufferQueue.itemAt(0); 4366 } else { 4367 pInBuffer = &inBuffer; 4368 } 4369 4370 if (pInBuffer->frameCount == 0) { 4371 break; 4372 } 4373 4374 if (mOutBuffer.frameCount == 0) { 4375 mOutBuffer.frameCount = pInBuffer->frameCount; 4376 nsecs_t startTime = systemTime(); 4377 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4378 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4379 outputBufferFull = true; 4380 break; 4381 } 4382 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4383 if (waitTimeLeftMs >= waitTimeMs) { 4384 waitTimeLeftMs -= waitTimeMs; 4385 } else { 4386 waitTimeLeftMs = 0; 4387 } 4388 } 4389 4390 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4391 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4392 mCblk->stepUser(outFrames); 4393 pInBuffer->frameCount -= outFrames; 4394 pInBuffer->i16 += outFrames * channelCount; 4395 mOutBuffer.frameCount -= outFrames; 4396 mOutBuffer.i16 += outFrames * channelCount; 4397 4398 if (pInBuffer->frameCount == 0) { 4399 if (mBufferQueue.size()) { 4400 mBufferQueue.removeAt(0); 4401 delete [] pInBuffer->mBuffer; 4402 delete pInBuffer; 4403 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4404 } else { 4405 break; 4406 } 4407 } 4408 } 4409 4410 // If we could not write all frames, allocate a buffer and queue it for next time. 4411 if (inBuffer.frameCount) { 4412 sp<ThreadBase> thread = mThread.promote(); 4413 if (thread != 0 && !thread->standby()) { 4414 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4415 pInBuffer = new Buffer; 4416 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4417 pInBuffer->frameCount = inBuffer.frameCount; 4418 pInBuffer->i16 = pInBuffer->mBuffer; 4419 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4420 mBufferQueue.add(pInBuffer); 4421 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4422 } else { 4423 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4424 } 4425 } 4426 } 4427 4428 // Calling write() with a 0 length buffer, means that no more data will be written: 4429 // If no more buffers are pending, fill output track buffer to make sure it is started 4430 // by output mixer. 4431 if (frames == 0 && mBufferQueue.size() == 0) { 4432 if (mCblk->user < mCblk->frameCount) { 4433 frames = mCblk->frameCount - mCblk->user; 4434 pInBuffer = new Buffer; 4435 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4436 pInBuffer->frameCount = frames; 4437 pInBuffer->i16 = pInBuffer->mBuffer; 4438 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4439 mBufferQueue.add(pInBuffer); 4440 } else if (mActive) { 4441 stop(); 4442 } 4443 } 4444 4445 return outputBufferFull; 4446} 4447 4448status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4449{ 4450 int active; 4451 status_t result; 4452 audio_track_cblk_t* cblk = mCblk; 4453 uint32_t framesReq = buffer->frameCount; 4454 4455// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4456 buffer->frameCount = 0; 4457 4458 uint32_t framesAvail = cblk->framesAvailable(); 4459 4460 4461 if (framesAvail == 0) { 4462 Mutex::Autolock _l(cblk->lock); 4463 goto start_loop_here; 4464 while (framesAvail == 0) { 4465 active = mActive; 4466 if (CC_UNLIKELY(!active)) { 4467 ALOGV("Not active and NO_MORE_BUFFERS"); 4468 return NO_MORE_BUFFERS; 4469 } 4470 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4471 if (result != NO_ERROR) { 4472 return NO_MORE_BUFFERS; 4473 } 4474 // read the server count again 4475 start_loop_here: 4476 framesAvail = cblk->framesAvailable_l(); 4477 } 4478 } 4479 4480// if (framesAvail < framesReq) { 4481// return NO_MORE_BUFFERS; 4482// } 4483 4484 if (framesReq > framesAvail) { 4485 framesReq = framesAvail; 4486 } 4487 4488 uint32_t u = cblk->user; 4489 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4490 4491 if (u + framesReq > bufferEnd) { 4492 framesReq = bufferEnd - u; 4493 } 4494 4495 buffer->frameCount = framesReq; 4496 buffer->raw = (void *)cblk->buffer(u); 4497 return NO_ERROR; 4498} 4499 4500 4501void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4502{ 4503 size_t size = mBufferQueue.size(); 4504 4505 for (size_t i = 0; i < size; i++) { 4506 Buffer *pBuffer = mBufferQueue.itemAt(i); 4507 delete [] pBuffer->mBuffer; 4508 delete pBuffer; 4509 } 4510 mBufferQueue.clear(); 4511} 4512 4513// ---------------------------------------------------------------------------- 4514 4515AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4516 : RefBase(), 4517 mAudioFlinger(audioFlinger), 4518 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4519 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4520 mPid(pid), 4521 mTimedTrackCount(0) 4522{ 4523 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4524} 4525 4526// Client destructor must be called with AudioFlinger::mLock held 4527AudioFlinger::Client::~Client() 4528{ 4529 mAudioFlinger->removeClient_l(mPid); 4530} 4531 4532sp<MemoryDealer> AudioFlinger::Client::heap() const 4533{ 4534 return mMemoryDealer; 4535} 4536 4537// Reserve one of the limited slots for a timed audio track associated 4538// with this client 4539bool AudioFlinger::Client::reserveTimedTrack() 4540{ 4541 const int kMaxTimedTracksPerClient = 4; 4542 4543 Mutex::Autolock _l(mTimedTrackLock); 4544 4545 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4546 ALOGW("can not create timed track - pid %d has exceeded the limit", 4547 mPid); 4548 return false; 4549 } 4550 4551 mTimedTrackCount++; 4552 return true; 4553} 4554 4555// Release a slot for a timed audio track 4556void AudioFlinger::Client::releaseTimedTrack() 4557{ 4558 Mutex::Autolock _l(mTimedTrackLock); 4559 mTimedTrackCount--; 4560} 4561 4562// ---------------------------------------------------------------------------- 4563 4564AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4565 const sp<IAudioFlingerClient>& client, 4566 pid_t pid) 4567 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4568{ 4569} 4570 4571AudioFlinger::NotificationClient::~NotificationClient() 4572{ 4573} 4574 4575void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4576{ 4577 sp<NotificationClient> keep(this); 4578 mAudioFlinger->removeNotificationClient(mPid); 4579} 4580 4581// ---------------------------------------------------------------------------- 4582 4583AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4584 : BnAudioTrack(), 4585 mTrack(track) 4586{ 4587} 4588 4589AudioFlinger::TrackHandle::~TrackHandle() { 4590 // just stop the track on deletion, associated resources 4591 // will be freed from the main thread once all pending buffers have 4592 // been played. Unless it's not in the active track list, in which 4593 // case we free everything now... 4594 mTrack->destroy(); 4595} 4596 4597sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4598 return mTrack->getCblk(); 4599} 4600 4601status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4602 return mTrack->start(tid); 4603} 4604 4605void AudioFlinger::TrackHandle::stop() { 4606 mTrack->stop(); 4607} 4608 4609void AudioFlinger::TrackHandle::flush() { 4610 mTrack->flush(); 4611} 4612 4613void AudioFlinger::TrackHandle::mute(bool e) { 4614 mTrack->mute(e); 4615} 4616 4617void AudioFlinger::TrackHandle::pause() { 4618 mTrack->pause(); 4619} 4620 4621status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4622{ 4623 return mTrack->attachAuxEffect(EffectId); 4624} 4625 4626status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4627 sp<IMemory>* buffer) { 4628 if (!mTrack->isTimedTrack()) 4629 return INVALID_OPERATION; 4630 4631 PlaybackThread::TimedTrack* tt = 4632 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4633 return tt->allocateTimedBuffer(size, buffer); 4634} 4635 4636status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4637 int64_t pts) { 4638 if (!mTrack->isTimedTrack()) 4639 return INVALID_OPERATION; 4640 4641 PlaybackThread::TimedTrack* tt = 4642 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4643 return tt->queueTimedBuffer(buffer, pts); 4644} 4645 4646status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4647 const LinearTransform& xform, int target) { 4648 4649 if (!mTrack->isTimedTrack()) 4650 return INVALID_OPERATION; 4651 4652 PlaybackThread::TimedTrack* tt = 4653 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4654 return tt->setMediaTimeTransform( 4655 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4656} 4657 4658status_t AudioFlinger::TrackHandle::onTransact( 4659 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4660{ 4661 return BnAudioTrack::onTransact(code, data, reply, flags); 4662} 4663 4664// ---------------------------------------------------------------------------- 4665 4666sp<IAudioRecord> AudioFlinger::openRecord( 4667 pid_t pid, 4668 audio_io_handle_t input, 4669 uint32_t sampleRate, 4670 audio_format_t format, 4671 uint32_t channelMask, 4672 int frameCount, 4673 uint32_t flags, 4674 int *sessionId, 4675 status_t *status) 4676{ 4677 sp<RecordThread::RecordTrack> recordTrack; 4678 sp<RecordHandle> recordHandle; 4679 sp<Client> client; 4680 status_t lStatus; 4681 RecordThread *thread; 4682 size_t inFrameCount; 4683 int lSessionId; 4684 4685 // check calling permissions 4686 if (!recordingAllowed()) { 4687 lStatus = PERMISSION_DENIED; 4688 goto Exit; 4689 } 4690 4691 // add client to list 4692 { // scope for mLock 4693 Mutex::Autolock _l(mLock); 4694 thread = checkRecordThread_l(input); 4695 if (thread == NULL) { 4696 lStatus = BAD_VALUE; 4697 goto Exit; 4698 } 4699 4700 client = registerPid_l(pid); 4701 4702 // If no audio session id is provided, create one here 4703 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4704 lSessionId = *sessionId; 4705 } else { 4706 lSessionId = nextUniqueId(); 4707 if (sessionId != NULL) { 4708 *sessionId = lSessionId; 4709 } 4710 } 4711 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4712 recordTrack = thread->createRecordTrack_l(client, 4713 sampleRate, 4714 format, 4715 channelMask, 4716 frameCount, 4717 flags, 4718 lSessionId, 4719 &lStatus); 4720 } 4721 if (lStatus != NO_ERROR) { 4722 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4723 // destructor is called by the TrackBase destructor with mLock held 4724 client.clear(); 4725 recordTrack.clear(); 4726 goto Exit; 4727 } 4728 4729 // return to handle to client 4730 recordHandle = new RecordHandle(recordTrack); 4731 lStatus = NO_ERROR; 4732 4733Exit: 4734 if (status) { 4735 *status = lStatus; 4736 } 4737 return recordHandle; 4738} 4739 4740// ---------------------------------------------------------------------------- 4741 4742AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4743 : BnAudioRecord(), 4744 mRecordTrack(recordTrack) 4745{ 4746} 4747 4748AudioFlinger::RecordHandle::~RecordHandle() { 4749 stop(); 4750} 4751 4752sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4753 return mRecordTrack->getCblk(); 4754} 4755 4756status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4757 ALOGV("RecordHandle::start()"); 4758 return mRecordTrack->start(tid); 4759} 4760 4761void AudioFlinger::RecordHandle::stop() { 4762 ALOGV("RecordHandle::stop()"); 4763 mRecordTrack->stop(); 4764} 4765 4766status_t AudioFlinger::RecordHandle::onTransact( 4767 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4768{ 4769 return BnAudioRecord::onTransact(code, data, reply, flags); 4770} 4771 4772// ---------------------------------------------------------------------------- 4773 4774AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4775 AudioStreamIn *input, 4776 uint32_t sampleRate, 4777 uint32_t channels, 4778 audio_io_handle_t id, 4779 uint32_t device) : 4780 ThreadBase(audioFlinger, id, device, RECORD), 4781 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4782 // mRsmpInIndex and mInputBytes set by readInputParameters() 4783 mReqChannelCount(popcount(channels)), 4784 mReqSampleRate(sampleRate) 4785 // mBytesRead is only meaningful while active, and so is cleared in start() 4786 // (but might be better to also clear here for dump?) 4787{ 4788 snprintf(mName, kNameLength, "AudioIn_%d", id); 4789 4790 readInputParameters(); 4791} 4792 4793 4794AudioFlinger::RecordThread::~RecordThread() 4795{ 4796 delete[] mRsmpInBuffer; 4797 delete mResampler; 4798 delete[] mRsmpOutBuffer; 4799} 4800 4801void AudioFlinger::RecordThread::onFirstRef() 4802{ 4803 run(mName, PRIORITY_URGENT_AUDIO); 4804} 4805 4806status_t AudioFlinger::RecordThread::readyToRun() 4807{ 4808 status_t status = initCheck(); 4809 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4810 return status; 4811} 4812 4813bool AudioFlinger::RecordThread::threadLoop() 4814{ 4815 AudioBufferProvider::Buffer buffer; 4816 sp<RecordTrack> activeTrack; 4817 Vector< sp<EffectChain> > effectChains; 4818 4819 nsecs_t lastWarning = 0; 4820 4821 acquireWakeLock(); 4822 4823 // start recording 4824 while (!exitPending()) { 4825 4826 processConfigEvents(); 4827 4828 { // scope for mLock 4829 Mutex::Autolock _l(mLock); 4830 checkForNewParameters_l(); 4831 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4832 if (!mStandby) { 4833 mInput->stream->common.standby(&mInput->stream->common); 4834 mStandby = true; 4835 } 4836 4837 if (exitPending()) break; 4838 4839 releaseWakeLock_l(); 4840 ALOGV("RecordThread: loop stopping"); 4841 // go to sleep 4842 mWaitWorkCV.wait(mLock); 4843 ALOGV("RecordThread: loop starting"); 4844 acquireWakeLock_l(); 4845 continue; 4846 } 4847 if (mActiveTrack != 0) { 4848 if (mActiveTrack->mState == TrackBase::PAUSING) { 4849 if (!mStandby) { 4850 mInput->stream->common.standby(&mInput->stream->common); 4851 mStandby = true; 4852 } 4853 mActiveTrack.clear(); 4854 mStartStopCond.broadcast(); 4855 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4856 if (mReqChannelCount != mActiveTrack->channelCount()) { 4857 mActiveTrack.clear(); 4858 mStartStopCond.broadcast(); 4859 } else if (mBytesRead != 0) { 4860 // record start succeeds only if first read from audio input 4861 // succeeds 4862 if (mBytesRead > 0) { 4863 mActiveTrack->mState = TrackBase::ACTIVE; 4864 } else { 4865 mActiveTrack.clear(); 4866 } 4867 mStartStopCond.broadcast(); 4868 } 4869 mStandby = false; 4870 } 4871 } 4872 lockEffectChains_l(effectChains); 4873 } 4874 4875 if (mActiveTrack != 0) { 4876 if (mActiveTrack->mState != TrackBase::ACTIVE && 4877 mActiveTrack->mState != TrackBase::RESUMING) { 4878 unlockEffectChains(effectChains); 4879 usleep(kRecordThreadSleepUs); 4880 continue; 4881 } 4882 for (size_t i = 0; i < effectChains.size(); i ++) { 4883 effectChains[i]->process_l(); 4884 } 4885 4886 buffer.frameCount = mFrameCount; 4887 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4888 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4889 size_t framesOut = buffer.frameCount; 4890 if (mResampler == NULL) { 4891 // no resampling 4892 while (framesOut) { 4893 size_t framesIn = mFrameCount - mRsmpInIndex; 4894 if (framesIn) { 4895 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4896 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4897 if (framesIn > framesOut) 4898 framesIn = framesOut; 4899 mRsmpInIndex += framesIn; 4900 framesOut -= framesIn; 4901 if ((int)mChannelCount == mReqChannelCount || 4902 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4903 memcpy(dst, src, framesIn * mFrameSize); 4904 } else { 4905 int16_t *src16 = (int16_t *)src; 4906 int16_t *dst16 = (int16_t *)dst; 4907 if (mChannelCount == 1) { 4908 while (framesIn--) { 4909 *dst16++ = *src16; 4910 *dst16++ = *src16++; 4911 } 4912 } else { 4913 while (framesIn--) { 4914 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4915 src16 += 2; 4916 } 4917 } 4918 } 4919 } 4920 if (framesOut && mFrameCount == mRsmpInIndex) { 4921 if (framesOut == mFrameCount && 4922 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4923 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4924 framesOut = 0; 4925 } else { 4926 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4927 mRsmpInIndex = 0; 4928 } 4929 if (mBytesRead < 0) { 4930 ALOGE("Error reading audio input"); 4931 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4932 // Force input into standby so that it tries to 4933 // recover at next read attempt 4934 mInput->stream->common.standby(&mInput->stream->common); 4935 usleep(kRecordThreadSleepUs); 4936 } 4937 mRsmpInIndex = mFrameCount; 4938 framesOut = 0; 4939 buffer.frameCount = 0; 4940 } 4941 } 4942 } 4943 } else { 4944 // resampling 4945 4946 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4947 // alter output frame count as if we were expecting stereo samples 4948 if (mChannelCount == 1 && mReqChannelCount == 1) { 4949 framesOut >>= 1; 4950 } 4951 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4952 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4953 // are 32 bit aligned which should be always true. 4954 if (mChannelCount == 2 && mReqChannelCount == 1) { 4955 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4956 // the resampler always outputs stereo samples: do post stereo to mono conversion 4957 int16_t *src = (int16_t *)mRsmpOutBuffer; 4958 int16_t *dst = buffer.i16; 4959 while (framesOut--) { 4960 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4961 src += 2; 4962 } 4963 } else { 4964 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4965 } 4966 4967 } 4968 mActiveTrack->releaseBuffer(&buffer); 4969 mActiveTrack->overflow(); 4970 } 4971 // client isn't retrieving buffers fast enough 4972 else { 4973 if (!mActiveTrack->setOverflow()) { 4974 nsecs_t now = systemTime(); 4975 if ((now - lastWarning) > kWarningThrottleNs) { 4976 ALOGW("RecordThread: buffer overflow"); 4977 lastWarning = now; 4978 } 4979 } 4980 // Release the processor for a while before asking for a new buffer. 4981 // This will give the application more chance to read from the buffer and 4982 // clear the overflow. 4983 usleep(kRecordThreadSleepUs); 4984 } 4985 } 4986 // enable changes in effect chain 4987 unlockEffectChains(effectChains); 4988 effectChains.clear(); 4989 } 4990 4991 if (!mStandby) { 4992 mInput->stream->common.standby(&mInput->stream->common); 4993 } 4994 mActiveTrack.clear(); 4995 4996 mStartStopCond.broadcast(); 4997 4998 releaseWakeLock(); 4999 5000 ALOGV("RecordThread %p exiting", this); 5001 return false; 5002} 5003 5004 5005sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5006 const sp<AudioFlinger::Client>& client, 5007 uint32_t sampleRate, 5008 audio_format_t format, 5009 int channelMask, 5010 int frameCount, 5011 uint32_t flags, 5012 int sessionId, 5013 status_t *status) 5014{ 5015 sp<RecordTrack> track; 5016 status_t lStatus; 5017 5018 lStatus = initCheck(); 5019 if (lStatus != NO_ERROR) { 5020 ALOGE("Audio driver not initialized."); 5021 goto Exit; 5022 } 5023 5024 { // scope for mLock 5025 Mutex::Autolock _l(mLock); 5026 5027 track = new RecordTrack(this, client, sampleRate, 5028 format, channelMask, frameCount, flags, sessionId); 5029 5030 if (track->getCblk() == 0) { 5031 lStatus = NO_MEMORY; 5032 goto Exit; 5033 } 5034 5035 mTrack = track.get(); 5036 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5037 bool suspend = audio_is_bluetooth_sco_device( 5038 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5039 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5040 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5041 } 5042 lStatus = NO_ERROR; 5043 5044Exit: 5045 if (status) { 5046 *status = lStatus; 5047 } 5048 return track; 5049} 5050 5051status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5052{ 5053 ALOGV("RecordThread::start tid=%d", tid); 5054 sp <ThreadBase> strongMe = this; 5055 status_t status = NO_ERROR; 5056 { 5057 AutoMutex lock(mLock); 5058 if (mActiveTrack != 0) { 5059 if (recordTrack != mActiveTrack.get()) { 5060 status = -EBUSY; 5061 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5062 mActiveTrack->mState = TrackBase::ACTIVE; 5063 } 5064 return status; 5065 } 5066 5067 recordTrack->mState = TrackBase::IDLE; 5068 mActiveTrack = recordTrack; 5069 mLock.unlock(); 5070 status_t status = AudioSystem::startInput(mId); 5071 mLock.lock(); 5072 if (status != NO_ERROR) { 5073 mActiveTrack.clear(); 5074 return status; 5075 } 5076 mRsmpInIndex = mFrameCount; 5077 mBytesRead = 0; 5078 if (mResampler != NULL) { 5079 mResampler->reset(); 5080 } 5081 mActiveTrack->mState = TrackBase::RESUMING; 5082 // signal thread to start 5083 ALOGV("Signal record thread"); 5084 mWaitWorkCV.signal(); 5085 // do not wait for mStartStopCond if exiting 5086 if (exitPending()) { 5087 mActiveTrack.clear(); 5088 status = INVALID_OPERATION; 5089 goto startError; 5090 } 5091 mStartStopCond.wait(mLock); 5092 if (mActiveTrack == 0) { 5093 ALOGV("Record failed to start"); 5094 status = BAD_VALUE; 5095 goto startError; 5096 } 5097 ALOGV("Record started OK"); 5098 return status; 5099 } 5100startError: 5101 AudioSystem::stopInput(mId); 5102 return status; 5103} 5104 5105void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5106 ALOGV("RecordThread::stop"); 5107 sp <ThreadBase> strongMe = this; 5108 { 5109 AutoMutex lock(mLock); 5110 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5111 mActiveTrack->mState = TrackBase::PAUSING; 5112 // do not wait for mStartStopCond if exiting 5113 if (exitPending()) { 5114 return; 5115 } 5116 mStartStopCond.wait(mLock); 5117 // if we have been restarted, recordTrack == mActiveTrack.get() here 5118 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5119 mLock.unlock(); 5120 AudioSystem::stopInput(mId); 5121 mLock.lock(); 5122 ALOGV("Record stopped OK"); 5123 } 5124 } 5125 } 5126} 5127 5128status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5129{ 5130 const size_t SIZE = 256; 5131 char buffer[SIZE]; 5132 String8 result; 5133 5134 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5135 result.append(buffer); 5136 5137 if (mActiveTrack != 0) { 5138 result.append("Active Track:\n"); 5139 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5140 mActiveTrack->dump(buffer, SIZE); 5141 result.append(buffer); 5142 5143 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5144 result.append(buffer); 5145 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5146 result.append(buffer); 5147 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5148 result.append(buffer); 5149 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5150 result.append(buffer); 5151 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5152 result.append(buffer); 5153 5154 5155 } else { 5156 result.append("No record client\n"); 5157 } 5158 write(fd, result.string(), result.size()); 5159 5160 dumpBase(fd, args); 5161 dumpEffectChains(fd, args); 5162 5163 return NO_ERROR; 5164} 5165 5166status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5167{ 5168 size_t framesReq = buffer->frameCount; 5169 size_t framesReady = mFrameCount - mRsmpInIndex; 5170 int channelCount; 5171 5172 if (framesReady == 0) { 5173 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5174 if (mBytesRead < 0) { 5175 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5176 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5177 // Force input into standby so that it tries to 5178 // recover at next read attempt 5179 mInput->stream->common.standby(&mInput->stream->common); 5180 usleep(kRecordThreadSleepUs); 5181 } 5182 buffer->raw = NULL; 5183 buffer->frameCount = 0; 5184 return NOT_ENOUGH_DATA; 5185 } 5186 mRsmpInIndex = 0; 5187 framesReady = mFrameCount; 5188 } 5189 5190 if (framesReq > framesReady) { 5191 framesReq = framesReady; 5192 } 5193 5194 if (mChannelCount == 1 && mReqChannelCount == 2) { 5195 channelCount = 1; 5196 } else { 5197 channelCount = 2; 5198 } 5199 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5200 buffer->frameCount = framesReq; 5201 return NO_ERROR; 5202} 5203 5204void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5205{ 5206 mRsmpInIndex += buffer->frameCount; 5207 buffer->frameCount = 0; 5208} 5209 5210bool AudioFlinger::RecordThread::checkForNewParameters_l() 5211{ 5212 bool reconfig = false; 5213 5214 while (!mNewParameters.isEmpty()) { 5215 status_t status = NO_ERROR; 5216 String8 keyValuePair = mNewParameters[0]; 5217 AudioParameter param = AudioParameter(keyValuePair); 5218 int value; 5219 audio_format_t reqFormat = mFormat; 5220 int reqSamplingRate = mReqSampleRate; 5221 int reqChannelCount = mReqChannelCount; 5222 5223 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5224 reqSamplingRate = value; 5225 reconfig = true; 5226 } 5227 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5228 reqFormat = (audio_format_t) value; 5229 reconfig = true; 5230 } 5231 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5232 reqChannelCount = popcount(value); 5233 reconfig = true; 5234 } 5235 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5236 // do not accept frame count changes if tracks are open as the track buffer 5237 // size depends on frame count and correct behavior would not be guaranteed 5238 // if frame count is changed after track creation 5239 if (mActiveTrack != 0) { 5240 status = INVALID_OPERATION; 5241 } else { 5242 reconfig = true; 5243 } 5244 } 5245 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5246 // forward device change to effects that have requested to be 5247 // aware of attached audio device. 5248 for (size_t i = 0; i < mEffectChains.size(); i++) { 5249 mEffectChains[i]->setDevice_l(value); 5250 } 5251 // store input device and output device but do not forward output device to audio HAL. 5252 // Note that status is ignored by the caller for output device 5253 // (see AudioFlinger::setParameters() 5254 if (value & AUDIO_DEVICE_OUT_ALL) { 5255 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5256 status = BAD_VALUE; 5257 } else { 5258 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5259 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5260 if (mTrack != NULL) { 5261 bool suspend = audio_is_bluetooth_sco_device( 5262 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5263 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5264 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5265 } 5266 } 5267 mDevice |= (uint32_t)value; 5268 } 5269 if (status == NO_ERROR) { 5270 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5271 if (status == INVALID_OPERATION) { 5272 mInput->stream->common.standby(&mInput->stream->common); 5273 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5274 } 5275 if (reconfig) { 5276 if (status == BAD_VALUE && 5277 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5278 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5279 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5280 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5281 (reqChannelCount < 3)) { 5282 status = NO_ERROR; 5283 } 5284 if (status == NO_ERROR) { 5285 readInputParameters(); 5286 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5287 } 5288 } 5289 } 5290 5291 mNewParameters.removeAt(0); 5292 5293 mParamStatus = status; 5294 mParamCond.signal(); 5295 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5296 // already timed out waiting for the status and will never signal the condition. 5297 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5298 } 5299 return reconfig; 5300} 5301 5302String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5303{ 5304 char *s; 5305 String8 out_s8 = String8(); 5306 5307 Mutex::Autolock _l(mLock); 5308 if (initCheck() != NO_ERROR) { 5309 return out_s8; 5310 } 5311 5312 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5313 out_s8 = String8(s); 5314 free(s); 5315 return out_s8; 5316} 5317 5318void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5319 AudioSystem::OutputDescriptor desc; 5320 void *param2 = NULL; 5321 5322 switch (event) { 5323 case AudioSystem::INPUT_OPENED: 5324 case AudioSystem::INPUT_CONFIG_CHANGED: 5325 desc.channels = mChannelMask; 5326 desc.samplingRate = mSampleRate; 5327 desc.format = mFormat; 5328 desc.frameCount = mFrameCount; 5329 desc.latency = 0; 5330 param2 = &desc; 5331 break; 5332 5333 case AudioSystem::INPUT_CLOSED: 5334 default: 5335 break; 5336 } 5337 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5338} 5339 5340void AudioFlinger::RecordThread::readInputParameters() 5341{ 5342 delete mRsmpInBuffer; 5343 // mRsmpInBuffer is always assigned a new[] below 5344 delete mRsmpOutBuffer; 5345 mRsmpOutBuffer = NULL; 5346 delete mResampler; 5347 mResampler = NULL; 5348 5349 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5350 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5351 mChannelCount = (uint16_t)popcount(mChannelMask); 5352 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5353 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5354 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5355 mFrameCount = mInputBytes / mFrameSize; 5356 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5357 5358 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5359 { 5360 int channelCount; 5361 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5362 // stereo to mono post process as the resampler always outputs stereo. 5363 if (mChannelCount == 1 && mReqChannelCount == 2) { 5364 channelCount = 1; 5365 } else { 5366 channelCount = 2; 5367 } 5368 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5369 mResampler->setSampleRate(mSampleRate); 5370 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5371 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5372 5373 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5374 if (mChannelCount == 1 && mReqChannelCount == 1) { 5375 mFrameCount >>= 1; 5376 } 5377 5378 } 5379 mRsmpInIndex = mFrameCount; 5380} 5381 5382unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5383{ 5384 Mutex::Autolock _l(mLock); 5385 if (initCheck() != NO_ERROR) { 5386 return 0; 5387 } 5388 5389 return mInput->stream->get_input_frames_lost(mInput->stream); 5390} 5391 5392uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5393{ 5394 Mutex::Autolock _l(mLock); 5395 uint32_t result = 0; 5396 if (getEffectChain_l(sessionId) != 0) { 5397 result = EFFECT_SESSION; 5398 } 5399 5400 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5401 result |= TRACK_SESSION; 5402 } 5403 5404 return result; 5405} 5406 5407AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5408{ 5409 Mutex::Autolock _l(mLock); 5410 return mTrack; 5411} 5412 5413AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5414{ 5415 Mutex::Autolock _l(mLock); 5416 return mInput; 5417} 5418 5419AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5420{ 5421 Mutex::Autolock _l(mLock); 5422 AudioStreamIn *input = mInput; 5423 mInput = NULL; 5424 return input; 5425} 5426 5427// this method must always be called either with ThreadBase mLock held or inside the thread loop 5428audio_stream_t* AudioFlinger::RecordThread::stream() 5429{ 5430 if (mInput == NULL) { 5431 return NULL; 5432 } 5433 return &mInput->stream->common; 5434} 5435 5436 5437// ---------------------------------------------------------------------------- 5438 5439audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5440 uint32_t *pSamplingRate, 5441 audio_format_t *pFormat, 5442 uint32_t *pChannels, 5443 uint32_t *pLatencyMs, 5444 uint32_t flags) 5445{ 5446 status_t status; 5447 PlaybackThread *thread = NULL; 5448 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5449 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5450 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5451 uint32_t channels = pChannels ? *pChannels : 0; 5452 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5453 audio_stream_out_t *outStream; 5454 audio_hw_device_t *outHwDev; 5455 5456 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5457 pDevices ? *pDevices : 0, 5458 samplingRate, 5459 format, 5460 channels, 5461 flags); 5462 5463 if (pDevices == NULL || *pDevices == 0) { 5464 return 0; 5465 } 5466 5467 Mutex::Autolock _l(mLock); 5468 5469 outHwDev = findSuitableHwDev_l(*pDevices); 5470 if (outHwDev == NULL) 5471 return 0; 5472 5473 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5474 &channels, &samplingRate, &outStream); 5475 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5476 outStream, 5477 samplingRate, 5478 format, 5479 channels, 5480 status); 5481 5482 mHardwareStatus = AUDIO_HW_IDLE; 5483 if (outStream != NULL) { 5484 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5485 audio_io_handle_t id = nextUniqueId(); 5486 5487 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5488 (format != AUDIO_FORMAT_PCM_16_BIT) || 5489 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5490 thread = new DirectOutputThread(this, output, id, *pDevices); 5491 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5492 } else { 5493 thread = new MixerThread(this, output, id, *pDevices); 5494 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5495 } 5496 mPlaybackThreads.add(id, thread); 5497 5498 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5499 if (pFormat != NULL) *pFormat = format; 5500 if (pChannels != NULL) *pChannels = channels; 5501 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5502 5503 // notify client processes of the new output creation 5504 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5505 return id; 5506 } 5507 5508 return 0; 5509} 5510 5511audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5512 audio_io_handle_t output2) 5513{ 5514 Mutex::Autolock _l(mLock); 5515 MixerThread *thread1 = checkMixerThread_l(output1); 5516 MixerThread *thread2 = checkMixerThread_l(output2); 5517 5518 if (thread1 == NULL || thread2 == NULL) { 5519 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5520 return 0; 5521 } 5522 5523 audio_io_handle_t id = nextUniqueId(); 5524 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5525 thread->addOutputTrack(thread2); 5526 mPlaybackThreads.add(id, thread); 5527 // notify client processes of the new output creation 5528 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5529 return id; 5530} 5531 5532status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5533{ 5534 // keep strong reference on the playback thread so that 5535 // it is not destroyed while exit() is executed 5536 sp <PlaybackThread> thread; 5537 { 5538 Mutex::Autolock _l(mLock); 5539 thread = checkPlaybackThread_l(output); 5540 if (thread == NULL) { 5541 return BAD_VALUE; 5542 } 5543 5544 ALOGV("closeOutput() %d", output); 5545 5546 if (thread->type() == ThreadBase::MIXER) { 5547 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5548 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5549 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5550 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5551 } 5552 } 5553 } 5554 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5555 mPlaybackThreads.removeItem(output); 5556 } 5557 thread->exit(); 5558 // The thread entity (active unit of execution) is no longer running here, 5559 // but the ThreadBase container still exists. 5560 5561 if (thread->type() != ThreadBase::DUPLICATING) { 5562 AudioStreamOut *out = thread->clearOutput(); 5563 assert(out != NULL); 5564 // from now on thread->mOutput is NULL 5565 out->hwDev->close_output_stream(out->hwDev, out->stream); 5566 delete out; 5567 } 5568 return NO_ERROR; 5569} 5570 5571status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5572{ 5573 Mutex::Autolock _l(mLock); 5574 PlaybackThread *thread = checkPlaybackThread_l(output); 5575 5576 if (thread == NULL) { 5577 return BAD_VALUE; 5578 } 5579 5580 ALOGV("suspendOutput() %d", output); 5581 thread->suspend(); 5582 5583 return NO_ERROR; 5584} 5585 5586status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5587{ 5588 Mutex::Autolock _l(mLock); 5589 PlaybackThread *thread = checkPlaybackThread_l(output); 5590 5591 if (thread == NULL) { 5592 return BAD_VALUE; 5593 } 5594 5595 ALOGV("restoreOutput() %d", output); 5596 5597 thread->restore(); 5598 5599 return NO_ERROR; 5600} 5601 5602audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5603 uint32_t *pSamplingRate, 5604 audio_format_t *pFormat, 5605 uint32_t *pChannels, 5606 audio_in_acoustics_t acoustics) 5607{ 5608 status_t status; 5609 RecordThread *thread = NULL; 5610 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5611 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5612 uint32_t channels = pChannels ? *pChannels : 0; 5613 uint32_t reqSamplingRate = samplingRate; 5614 audio_format_t reqFormat = format; 5615 uint32_t reqChannels = channels; 5616 audio_stream_in_t *inStream; 5617 audio_hw_device_t *inHwDev; 5618 5619 if (pDevices == NULL || *pDevices == 0) { 5620 return 0; 5621 } 5622 5623 Mutex::Autolock _l(mLock); 5624 5625 inHwDev = findSuitableHwDev_l(*pDevices); 5626 if (inHwDev == NULL) 5627 return 0; 5628 5629 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5630 &channels, &samplingRate, 5631 acoustics, 5632 &inStream); 5633 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5634 inStream, 5635 samplingRate, 5636 format, 5637 channels, 5638 acoustics, 5639 status); 5640 5641 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5642 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5643 // or stereo to mono conversions on 16 bit PCM inputs. 5644 if (inStream == NULL && status == BAD_VALUE && 5645 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5646 (samplingRate <= 2 * reqSamplingRate) && 5647 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5648 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5649 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5650 &channels, &samplingRate, 5651 acoustics, 5652 &inStream); 5653 } 5654 5655 if (inStream != NULL) { 5656 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5657 5658 audio_io_handle_t id = nextUniqueId(); 5659 // Start record thread 5660 // RecorThread require both input and output device indication to forward to audio 5661 // pre processing modules 5662 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5663 thread = new RecordThread(this, 5664 input, 5665 reqSamplingRate, 5666 reqChannels, 5667 id, 5668 device); 5669 mRecordThreads.add(id, thread); 5670 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5671 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5672 if (pFormat != NULL) *pFormat = format; 5673 if (pChannels != NULL) *pChannels = reqChannels; 5674 5675 input->stream->common.standby(&input->stream->common); 5676 5677 // notify client processes of the new input creation 5678 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5679 return id; 5680 } 5681 5682 return 0; 5683} 5684 5685status_t AudioFlinger::closeInput(audio_io_handle_t input) 5686{ 5687 // keep strong reference on the record thread so that 5688 // it is not destroyed while exit() is executed 5689 sp <RecordThread> thread; 5690 { 5691 Mutex::Autolock _l(mLock); 5692 thread = checkRecordThread_l(input); 5693 if (thread == NULL) { 5694 return BAD_VALUE; 5695 } 5696 5697 ALOGV("closeInput() %d", input); 5698 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5699 mRecordThreads.removeItem(input); 5700 } 5701 thread->exit(); 5702 // The thread entity (active unit of execution) is no longer running here, 5703 // but the ThreadBase container still exists. 5704 5705 AudioStreamIn *in = thread->clearInput(); 5706 assert(in != NULL); 5707 // from now on thread->mInput is NULL 5708 in->hwDev->close_input_stream(in->hwDev, in->stream); 5709 delete in; 5710 5711 return NO_ERROR; 5712} 5713 5714status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5715{ 5716 Mutex::Autolock _l(mLock); 5717 MixerThread *dstThread = checkMixerThread_l(output); 5718 if (dstThread == NULL) { 5719 ALOGW("setStreamOutput() bad output id %d", output); 5720 return BAD_VALUE; 5721 } 5722 5723 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5724 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5725 5726 dstThread->setStreamValid(stream, true); 5727 5728 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5729 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5730 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5731 MixerThread *srcThread = (MixerThread *)thread; 5732 srcThread->setStreamValid(stream, false); 5733 srcThread->invalidateTracks(stream); 5734 } 5735 } 5736 5737 return NO_ERROR; 5738} 5739 5740 5741int AudioFlinger::newAudioSessionId() 5742{ 5743 return nextUniqueId(); 5744} 5745 5746void AudioFlinger::acquireAudioSessionId(int audioSession) 5747{ 5748 Mutex::Autolock _l(mLock); 5749 pid_t caller = IPCThreadState::self()->getCallingPid(); 5750 ALOGV("acquiring %d from %d", audioSession, caller); 5751 size_t num = mAudioSessionRefs.size(); 5752 for (size_t i = 0; i< num; i++) { 5753 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5754 if (ref->sessionid == audioSession && ref->pid == caller) { 5755 ref->cnt++; 5756 ALOGV(" incremented refcount to %d", ref->cnt); 5757 return; 5758 } 5759 } 5760 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5761 ALOGV(" added new entry for %d", audioSession); 5762} 5763 5764void AudioFlinger::releaseAudioSessionId(int audioSession) 5765{ 5766 Mutex::Autolock _l(mLock); 5767 pid_t caller = IPCThreadState::self()->getCallingPid(); 5768 ALOGV("releasing %d from %d", audioSession, caller); 5769 size_t num = mAudioSessionRefs.size(); 5770 for (size_t i = 0; i< num; i++) { 5771 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5772 if (ref->sessionid == audioSession && ref->pid == caller) { 5773 ref->cnt--; 5774 ALOGV(" decremented refcount to %d", ref->cnt); 5775 if (ref->cnt == 0) { 5776 mAudioSessionRefs.removeAt(i); 5777 delete ref; 5778 purgeStaleEffects_l(); 5779 } 5780 return; 5781 } 5782 } 5783 ALOGW("session id %d not found for pid %d", audioSession, caller); 5784} 5785 5786void AudioFlinger::purgeStaleEffects_l() { 5787 5788 ALOGV("purging stale effects"); 5789 5790 Vector< sp<EffectChain> > chains; 5791 5792 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5793 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5794 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5795 sp<EffectChain> ec = t->mEffectChains[j]; 5796 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5797 chains.push(ec); 5798 } 5799 } 5800 } 5801 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5802 sp<RecordThread> t = mRecordThreads.valueAt(i); 5803 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5804 sp<EffectChain> ec = t->mEffectChains[j]; 5805 chains.push(ec); 5806 } 5807 } 5808 5809 for (size_t i = 0; i < chains.size(); i++) { 5810 sp<EffectChain> ec = chains[i]; 5811 int sessionid = ec->sessionId(); 5812 sp<ThreadBase> t = ec->mThread.promote(); 5813 if (t == 0) { 5814 continue; 5815 } 5816 size_t numsessionrefs = mAudioSessionRefs.size(); 5817 bool found = false; 5818 for (size_t k = 0; k < numsessionrefs; k++) { 5819 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5820 if (ref->sessionid == sessionid) { 5821 ALOGV(" session %d still exists for %d with %d refs", 5822 sessionid, ref->pid, ref->cnt); 5823 found = true; 5824 break; 5825 } 5826 } 5827 if (!found) { 5828 // remove all effects from the chain 5829 while (ec->mEffects.size()) { 5830 sp<EffectModule> effect = ec->mEffects[0]; 5831 effect->unPin(); 5832 Mutex::Autolock _l (t->mLock); 5833 t->removeEffect_l(effect); 5834 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5835 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5836 if (handle != 0) { 5837 handle->mEffect.clear(); 5838 if (handle->mHasControl && handle->mEnabled) { 5839 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5840 } 5841 } 5842 } 5843 AudioSystem::unregisterEffect(effect->id()); 5844 } 5845 } 5846 } 5847 return; 5848} 5849 5850// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5851AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5852{ 5853 return mPlaybackThreads.valueFor(output).get(); 5854} 5855 5856// checkMixerThread_l() must be called with AudioFlinger::mLock held 5857AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5858{ 5859 PlaybackThread *thread = checkPlaybackThread_l(output); 5860 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5861} 5862 5863// checkRecordThread_l() must be called with AudioFlinger::mLock held 5864AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5865{ 5866 return mRecordThreads.valueFor(input).get(); 5867} 5868 5869uint32_t AudioFlinger::nextUniqueId() 5870{ 5871 return android_atomic_inc(&mNextUniqueId); 5872} 5873 5874AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5875{ 5876 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5877 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5878 AudioStreamOut *output = thread->getOutput(); 5879 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5880 return thread; 5881 } 5882 } 5883 return NULL; 5884} 5885 5886uint32_t AudioFlinger::primaryOutputDevice_l() 5887{ 5888 PlaybackThread *thread = primaryPlaybackThread_l(); 5889 5890 if (thread == NULL) { 5891 return 0; 5892 } 5893 5894 return thread->device(); 5895} 5896 5897 5898// ---------------------------------------------------------------------------- 5899// Effect management 5900// ---------------------------------------------------------------------------- 5901 5902 5903status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5904{ 5905 Mutex::Autolock _l(mLock); 5906 return EffectQueryNumberEffects(numEffects); 5907} 5908 5909status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5910{ 5911 Mutex::Autolock _l(mLock); 5912 return EffectQueryEffect(index, descriptor); 5913} 5914 5915status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5916 effect_descriptor_t *descriptor) const 5917{ 5918 Mutex::Autolock _l(mLock); 5919 return EffectGetDescriptor(pUuid, descriptor); 5920} 5921 5922 5923sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5924 effect_descriptor_t *pDesc, 5925 const sp<IEffectClient>& effectClient, 5926 int32_t priority, 5927 audio_io_handle_t io, 5928 int sessionId, 5929 status_t *status, 5930 int *id, 5931 int *enabled) 5932{ 5933 status_t lStatus = NO_ERROR; 5934 sp<EffectHandle> handle; 5935 effect_descriptor_t desc; 5936 5937 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5938 pid, effectClient.get(), priority, sessionId, io); 5939 5940 if (pDesc == NULL) { 5941 lStatus = BAD_VALUE; 5942 goto Exit; 5943 } 5944 5945 // check audio settings permission for global effects 5946 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5947 lStatus = PERMISSION_DENIED; 5948 goto Exit; 5949 } 5950 5951 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5952 // that can only be created by audio policy manager (running in same process) 5953 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5954 lStatus = PERMISSION_DENIED; 5955 goto Exit; 5956 } 5957 5958 if (io == 0) { 5959 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5960 // output must be specified by AudioPolicyManager when using session 5961 // AUDIO_SESSION_OUTPUT_STAGE 5962 lStatus = BAD_VALUE; 5963 goto Exit; 5964 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5965 // if the output returned by getOutputForEffect() is removed before we lock the 5966 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5967 // and we will exit safely 5968 io = AudioSystem::getOutputForEffect(&desc); 5969 } 5970 } 5971 5972 { 5973 Mutex::Autolock _l(mLock); 5974 5975 5976 if (!EffectIsNullUuid(&pDesc->uuid)) { 5977 // if uuid is specified, request effect descriptor 5978 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5979 if (lStatus < 0) { 5980 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5981 goto Exit; 5982 } 5983 } else { 5984 // if uuid is not specified, look for an available implementation 5985 // of the required type in effect factory 5986 if (EffectIsNullUuid(&pDesc->type)) { 5987 ALOGW("createEffect() no effect type"); 5988 lStatus = BAD_VALUE; 5989 goto Exit; 5990 } 5991 uint32_t numEffects = 0; 5992 effect_descriptor_t d; 5993 d.flags = 0; // prevent compiler warning 5994 bool found = false; 5995 5996 lStatus = EffectQueryNumberEffects(&numEffects); 5997 if (lStatus < 0) { 5998 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5999 goto Exit; 6000 } 6001 for (uint32_t i = 0; i < numEffects; i++) { 6002 lStatus = EffectQueryEffect(i, &desc); 6003 if (lStatus < 0) { 6004 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6005 continue; 6006 } 6007 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6008 // If matching type found save effect descriptor. If the session is 6009 // 0 and the effect is not auxiliary, continue enumeration in case 6010 // an auxiliary version of this effect type is available 6011 found = true; 6012 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6013 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6014 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6015 break; 6016 } 6017 } 6018 } 6019 if (!found) { 6020 lStatus = BAD_VALUE; 6021 ALOGW("createEffect() effect not found"); 6022 goto Exit; 6023 } 6024 // For same effect type, chose auxiliary version over insert version if 6025 // connect to output mix (Compliance to OpenSL ES) 6026 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6027 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6028 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6029 } 6030 } 6031 6032 // Do not allow auxiliary effects on a session different from 0 (output mix) 6033 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6034 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6035 lStatus = INVALID_OPERATION; 6036 goto Exit; 6037 } 6038 6039 // check recording permission for visualizer 6040 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6041 !recordingAllowed()) { 6042 lStatus = PERMISSION_DENIED; 6043 goto Exit; 6044 } 6045 6046 // return effect descriptor 6047 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6048 6049 // If output is not specified try to find a matching audio session ID in one of the 6050 // output threads. 6051 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6052 // because of code checking output when entering the function. 6053 // Note: io is never 0 when creating an effect on an input 6054 if (io == 0) { 6055 // look for the thread where the specified audio session is present 6056 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6057 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6058 io = mPlaybackThreads.keyAt(i); 6059 break; 6060 } 6061 } 6062 if (io == 0) { 6063 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6064 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6065 io = mRecordThreads.keyAt(i); 6066 break; 6067 } 6068 } 6069 } 6070 // If no output thread contains the requested session ID, default to 6071 // first output. The effect chain will be moved to the correct output 6072 // thread when a track with the same session ID is created 6073 if (io == 0 && mPlaybackThreads.size()) { 6074 io = mPlaybackThreads.keyAt(0); 6075 } 6076 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6077 } 6078 ThreadBase *thread = checkRecordThread_l(io); 6079 if (thread == NULL) { 6080 thread = checkPlaybackThread_l(io); 6081 if (thread == NULL) { 6082 ALOGE("createEffect() unknown output thread"); 6083 lStatus = BAD_VALUE; 6084 goto Exit; 6085 } 6086 } 6087 6088 sp<Client> client = registerPid_l(pid); 6089 6090 // create effect on selected output thread 6091 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6092 &desc, enabled, &lStatus); 6093 if (handle != 0 && id != NULL) { 6094 *id = handle->id(); 6095 } 6096 } 6097 6098Exit: 6099 if(status) { 6100 *status = lStatus; 6101 } 6102 return handle; 6103} 6104 6105status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6106 audio_io_handle_t dstOutput) 6107{ 6108 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6109 sessionId, srcOutput, dstOutput); 6110 Mutex::Autolock _l(mLock); 6111 if (srcOutput == dstOutput) { 6112 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6113 return NO_ERROR; 6114 } 6115 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6116 if (srcThread == NULL) { 6117 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6118 return BAD_VALUE; 6119 } 6120 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6121 if (dstThread == NULL) { 6122 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6123 return BAD_VALUE; 6124 } 6125 6126 Mutex::Autolock _dl(dstThread->mLock); 6127 Mutex::Autolock _sl(srcThread->mLock); 6128 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6129 6130 return NO_ERROR; 6131} 6132 6133// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6134status_t AudioFlinger::moveEffectChain_l(int sessionId, 6135 AudioFlinger::PlaybackThread *srcThread, 6136 AudioFlinger::PlaybackThread *dstThread, 6137 bool reRegister) 6138{ 6139 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6140 sessionId, srcThread, dstThread); 6141 6142 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6143 if (chain == 0) { 6144 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6145 sessionId, srcThread); 6146 return INVALID_OPERATION; 6147 } 6148 6149 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6150 // so that a new chain is created with correct parameters when first effect is added. This is 6151 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6152 // removed. 6153 srcThread->removeEffectChain_l(chain); 6154 6155 // transfer all effects one by one so that new effect chain is created on new thread with 6156 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6157 audio_io_handle_t dstOutput = dstThread->id(); 6158 sp<EffectChain> dstChain; 6159 uint32_t strategy = 0; // prevent compiler warning 6160 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6161 while (effect != 0) { 6162 srcThread->removeEffect_l(effect); 6163 dstThread->addEffect_l(effect); 6164 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6165 if (effect->state() == EffectModule::ACTIVE || 6166 effect->state() == EffectModule::STOPPING) { 6167 effect->start(); 6168 } 6169 // if the move request is not received from audio policy manager, the effect must be 6170 // re-registered with the new strategy and output 6171 if (dstChain == 0) { 6172 dstChain = effect->chain().promote(); 6173 if (dstChain == 0) { 6174 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6175 srcThread->addEffect_l(effect); 6176 return NO_INIT; 6177 } 6178 strategy = dstChain->strategy(); 6179 } 6180 if (reRegister) { 6181 AudioSystem::unregisterEffect(effect->id()); 6182 AudioSystem::registerEffect(&effect->desc(), 6183 dstOutput, 6184 strategy, 6185 sessionId, 6186 effect->id()); 6187 } 6188 effect = chain->getEffectFromId_l(0); 6189 } 6190 6191 return NO_ERROR; 6192} 6193 6194 6195// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6196sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6197 const sp<AudioFlinger::Client>& client, 6198 const sp<IEffectClient>& effectClient, 6199 int32_t priority, 6200 int sessionId, 6201 effect_descriptor_t *desc, 6202 int *enabled, 6203 status_t *status 6204 ) 6205{ 6206 sp<EffectModule> effect; 6207 sp<EffectHandle> handle; 6208 status_t lStatus; 6209 sp<EffectChain> chain; 6210 bool chainCreated = false; 6211 bool effectCreated = false; 6212 bool effectRegistered = false; 6213 6214 lStatus = initCheck(); 6215 if (lStatus != NO_ERROR) { 6216 ALOGW("createEffect_l() Audio driver not initialized."); 6217 goto Exit; 6218 } 6219 6220 // Do not allow effects with session ID 0 on direct output or duplicating threads 6221 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6222 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6223 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6224 desc->name, sessionId); 6225 lStatus = BAD_VALUE; 6226 goto Exit; 6227 } 6228 // Only Pre processor effects are allowed on input threads and only on input threads 6229 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6230 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6231 desc->name, desc->flags, mType); 6232 lStatus = BAD_VALUE; 6233 goto Exit; 6234 } 6235 6236 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6237 6238 { // scope for mLock 6239 Mutex::Autolock _l(mLock); 6240 6241 // check for existing effect chain with the requested audio session 6242 chain = getEffectChain_l(sessionId); 6243 if (chain == 0) { 6244 // create a new chain for this session 6245 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6246 chain = new EffectChain(this, sessionId); 6247 addEffectChain_l(chain); 6248 chain->setStrategy(getStrategyForSession_l(sessionId)); 6249 chainCreated = true; 6250 } else { 6251 effect = chain->getEffectFromDesc_l(desc); 6252 } 6253 6254 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6255 6256 if (effect == 0) { 6257 int id = mAudioFlinger->nextUniqueId(); 6258 // Check CPU and memory usage 6259 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6260 if (lStatus != NO_ERROR) { 6261 goto Exit; 6262 } 6263 effectRegistered = true; 6264 // create a new effect module if none present in the chain 6265 effect = new EffectModule(this, chain, desc, id, sessionId); 6266 lStatus = effect->status(); 6267 if (lStatus != NO_ERROR) { 6268 goto Exit; 6269 } 6270 lStatus = chain->addEffect_l(effect); 6271 if (lStatus != NO_ERROR) { 6272 goto Exit; 6273 } 6274 effectCreated = true; 6275 6276 effect->setDevice(mDevice); 6277 effect->setMode(mAudioFlinger->getMode()); 6278 } 6279 // create effect handle and connect it to effect module 6280 handle = new EffectHandle(effect, client, effectClient, priority); 6281 lStatus = effect->addHandle(handle); 6282 if (enabled != NULL) { 6283 *enabled = (int)effect->isEnabled(); 6284 } 6285 } 6286 6287Exit: 6288 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6289 Mutex::Autolock _l(mLock); 6290 if (effectCreated) { 6291 chain->removeEffect_l(effect); 6292 } 6293 if (effectRegistered) { 6294 AudioSystem::unregisterEffect(effect->id()); 6295 } 6296 if (chainCreated) { 6297 removeEffectChain_l(chain); 6298 } 6299 handle.clear(); 6300 } 6301 6302 if(status) { 6303 *status = lStatus; 6304 } 6305 return handle; 6306} 6307 6308sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6309{ 6310 sp<EffectChain> chain = getEffectChain_l(sessionId); 6311 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6312} 6313 6314// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6315// PlaybackThread::mLock held 6316status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6317{ 6318 // check for existing effect chain with the requested audio session 6319 int sessionId = effect->sessionId(); 6320 sp<EffectChain> chain = getEffectChain_l(sessionId); 6321 bool chainCreated = false; 6322 6323 if (chain == 0) { 6324 // create a new chain for this session 6325 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6326 chain = new EffectChain(this, sessionId); 6327 addEffectChain_l(chain); 6328 chain->setStrategy(getStrategyForSession_l(sessionId)); 6329 chainCreated = true; 6330 } 6331 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6332 6333 if (chain->getEffectFromId_l(effect->id()) != 0) { 6334 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6335 this, effect->desc().name, chain.get()); 6336 return BAD_VALUE; 6337 } 6338 6339 status_t status = chain->addEffect_l(effect); 6340 if (status != NO_ERROR) { 6341 if (chainCreated) { 6342 removeEffectChain_l(chain); 6343 } 6344 return status; 6345 } 6346 6347 effect->setDevice(mDevice); 6348 effect->setMode(mAudioFlinger->getMode()); 6349 return NO_ERROR; 6350} 6351 6352void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6353 6354 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6355 effect_descriptor_t desc = effect->desc(); 6356 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6357 detachAuxEffect_l(effect->id()); 6358 } 6359 6360 sp<EffectChain> chain = effect->chain().promote(); 6361 if (chain != 0) { 6362 // remove effect chain if removing last effect 6363 if (chain->removeEffect_l(effect) == 0) { 6364 removeEffectChain_l(chain); 6365 } 6366 } else { 6367 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6368 } 6369} 6370 6371void AudioFlinger::ThreadBase::lockEffectChains_l( 6372 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6373{ 6374 effectChains = mEffectChains; 6375 for (size_t i = 0; i < mEffectChains.size(); i++) { 6376 mEffectChains[i]->lock(); 6377 } 6378} 6379 6380void AudioFlinger::ThreadBase::unlockEffectChains( 6381 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6382{ 6383 for (size_t i = 0; i < effectChains.size(); i++) { 6384 effectChains[i]->unlock(); 6385 } 6386} 6387 6388sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6389{ 6390 Mutex::Autolock _l(mLock); 6391 return getEffectChain_l(sessionId); 6392} 6393 6394sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6395{ 6396 size_t size = mEffectChains.size(); 6397 for (size_t i = 0; i < size; i++) { 6398 if (mEffectChains[i]->sessionId() == sessionId) { 6399 return mEffectChains[i]; 6400 } 6401 } 6402 return 0; 6403} 6404 6405void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6406{ 6407 Mutex::Autolock _l(mLock); 6408 size_t size = mEffectChains.size(); 6409 for (size_t i = 0; i < size; i++) { 6410 mEffectChains[i]->setMode_l(mode); 6411 } 6412} 6413 6414void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6415 const wp<EffectHandle>& handle, 6416 bool unpinIfLast) { 6417 6418 Mutex::Autolock _l(mLock); 6419 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6420 // delete the effect module if removing last handle on it 6421 if (effect->removeHandle(handle) == 0) { 6422 if (!effect->isPinned() || unpinIfLast) { 6423 removeEffect_l(effect); 6424 AudioSystem::unregisterEffect(effect->id()); 6425 } 6426 } 6427} 6428 6429status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6430{ 6431 int session = chain->sessionId(); 6432 int16_t *buffer = mMixBuffer; 6433 bool ownsBuffer = false; 6434 6435 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6436 if (session > 0) { 6437 // Only one effect chain can be present in direct output thread and it uses 6438 // the mix buffer as input 6439 if (mType != DIRECT) { 6440 size_t numSamples = mFrameCount * mChannelCount; 6441 buffer = new int16_t[numSamples]; 6442 memset(buffer, 0, numSamples * sizeof(int16_t)); 6443 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6444 ownsBuffer = true; 6445 } 6446 6447 // Attach all tracks with same session ID to this chain. 6448 for (size_t i = 0; i < mTracks.size(); ++i) { 6449 sp<Track> track = mTracks[i]; 6450 if (session == track->sessionId()) { 6451 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6452 track->setMainBuffer(buffer); 6453 chain->incTrackCnt(); 6454 } 6455 } 6456 6457 // indicate all active tracks in the chain 6458 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6459 sp<Track> track = mActiveTracks[i].promote(); 6460 if (track == 0) continue; 6461 if (session == track->sessionId()) { 6462 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6463 chain->incActiveTrackCnt(); 6464 } 6465 } 6466 } 6467 6468 chain->setInBuffer(buffer, ownsBuffer); 6469 chain->setOutBuffer(mMixBuffer); 6470 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6471 // chains list in order to be processed last as it contains output stage effects 6472 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6473 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6474 // after track specific effects and before output stage 6475 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6476 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6477 // Effect chain for other sessions are inserted at beginning of effect 6478 // chains list to be processed before output mix effects. Relative order between other 6479 // sessions is not important 6480 size_t size = mEffectChains.size(); 6481 size_t i = 0; 6482 for (i = 0; i < size; i++) { 6483 if (mEffectChains[i]->sessionId() < session) break; 6484 } 6485 mEffectChains.insertAt(chain, i); 6486 checkSuspendOnAddEffectChain_l(chain); 6487 6488 return NO_ERROR; 6489} 6490 6491size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6492{ 6493 int session = chain->sessionId(); 6494 6495 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6496 6497 for (size_t i = 0; i < mEffectChains.size(); i++) { 6498 if (chain == mEffectChains[i]) { 6499 mEffectChains.removeAt(i); 6500 // detach all active tracks from the chain 6501 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6502 sp<Track> track = mActiveTracks[i].promote(); 6503 if (track == 0) continue; 6504 if (session == track->sessionId()) { 6505 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6506 chain.get(), session); 6507 chain->decActiveTrackCnt(); 6508 } 6509 } 6510 6511 // detach all tracks with same session ID from this chain 6512 for (size_t i = 0; i < mTracks.size(); ++i) { 6513 sp<Track> track = mTracks[i]; 6514 if (session == track->sessionId()) { 6515 track->setMainBuffer(mMixBuffer); 6516 chain->decTrackCnt(); 6517 } 6518 } 6519 break; 6520 } 6521 } 6522 return mEffectChains.size(); 6523} 6524 6525status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6526 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6527{ 6528 Mutex::Autolock _l(mLock); 6529 return attachAuxEffect_l(track, EffectId); 6530} 6531 6532status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6533 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6534{ 6535 status_t status = NO_ERROR; 6536 6537 if (EffectId == 0) { 6538 track->setAuxBuffer(0, NULL); 6539 } else { 6540 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6541 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6542 if (effect != 0) { 6543 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6544 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6545 } else { 6546 status = INVALID_OPERATION; 6547 } 6548 } else { 6549 status = BAD_VALUE; 6550 } 6551 } 6552 return status; 6553} 6554 6555void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6556{ 6557 for (size_t i = 0; i < mTracks.size(); ++i) { 6558 sp<Track> track = mTracks[i]; 6559 if (track->auxEffectId() == effectId) { 6560 attachAuxEffect_l(track, 0); 6561 } 6562 } 6563} 6564 6565status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6566{ 6567 // only one chain per input thread 6568 if (mEffectChains.size() != 0) { 6569 return INVALID_OPERATION; 6570 } 6571 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6572 6573 chain->setInBuffer(NULL); 6574 chain->setOutBuffer(NULL); 6575 6576 checkSuspendOnAddEffectChain_l(chain); 6577 6578 mEffectChains.add(chain); 6579 6580 return NO_ERROR; 6581} 6582 6583size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6584{ 6585 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6586 ALOGW_IF(mEffectChains.size() != 1, 6587 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6588 chain.get(), mEffectChains.size(), this); 6589 if (mEffectChains.size() == 1) { 6590 mEffectChains.removeAt(0); 6591 } 6592 return 0; 6593} 6594 6595// ---------------------------------------------------------------------------- 6596// EffectModule implementation 6597// ---------------------------------------------------------------------------- 6598 6599#undef LOG_TAG 6600#define LOG_TAG "AudioFlinger::EffectModule" 6601 6602AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6603 const wp<AudioFlinger::EffectChain>& chain, 6604 effect_descriptor_t *desc, 6605 int id, 6606 int sessionId) 6607 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6608 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6609{ 6610 ALOGV("Constructor %p", this); 6611 int lStatus; 6612 sp<ThreadBase> thread = mThread.promote(); 6613 if (thread == 0) { 6614 return; 6615 } 6616 6617 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6618 6619 // create effect engine from effect factory 6620 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6621 6622 if (mStatus != NO_ERROR) { 6623 return; 6624 } 6625 lStatus = init(); 6626 if (lStatus < 0) { 6627 mStatus = lStatus; 6628 goto Error; 6629 } 6630 6631 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6632 mPinned = true; 6633 } 6634 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6635 return; 6636Error: 6637 EffectRelease(mEffectInterface); 6638 mEffectInterface = NULL; 6639 ALOGV("Constructor Error %d", mStatus); 6640} 6641 6642AudioFlinger::EffectModule::~EffectModule() 6643{ 6644 ALOGV("Destructor %p", this); 6645 if (mEffectInterface != NULL) { 6646 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6647 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6648 sp<ThreadBase> thread = mThread.promote(); 6649 if (thread != 0) { 6650 audio_stream_t *stream = thread->stream(); 6651 if (stream != NULL) { 6652 stream->remove_audio_effect(stream, mEffectInterface); 6653 } 6654 } 6655 } 6656 // release effect engine 6657 EffectRelease(mEffectInterface); 6658 } 6659} 6660 6661status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6662{ 6663 status_t status; 6664 6665 Mutex::Autolock _l(mLock); 6666 int priority = handle->priority(); 6667 size_t size = mHandles.size(); 6668 sp<EffectHandle> h; 6669 size_t i; 6670 for (i = 0; i < size; i++) { 6671 h = mHandles[i].promote(); 6672 if (h == 0) continue; 6673 if (h->priority() <= priority) break; 6674 } 6675 // if inserted in first place, move effect control from previous owner to this handle 6676 if (i == 0) { 6677 bool enabled = false; 6678 if (h != 0) { 6679 enabled = h->enabled(); 6680 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6681 } 6682 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6683 status = NO_ERROR; 6684 } else { 6685 status = ALREADY_EXISTS; 6686 } 6687 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6688 mHandles.insertAt(handle, i); 6689 return status; 6690} 6691 6692size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6693{ 6694 Mutex::Autolock _l(mLock); 6695 size_t size = mHandles.size(); 6696 size_t i; 6697 for (i = 0; i < size; i++) { 6698 if (mHandles[i] == handle) break; 6699 } 6700 if (i == size) { 6701 return size; 6702 } 6703 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6704 6705 bool enabled = false; 6706 EffectHandle *hdl = handle.unsafe_get(); 6707 if (hdl != NULL) { 6708 ALOGV("removeHandle() unsafe_get OK"); 6709 enabled = hdl->enabled(); 6710 } 6711 mHandles.removeAt(i); 6712 size = mHandles.size(); 6713 // if removed from first place, move effect control from this handle to next in line 6714 if (i == 0 && size != 0) { 6715 sp<EffectHandle> h = mHandles[0].promote(); 6716 if (h != 0) { 6717 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6718 } 6719 } 6720 6721 // Prevent calls to process() and other functions on effect interface from now on. 6722 // The effect engine will be released by the destructor when the last strong reference on 6723 // this object is released which can happen after next process is called. 6724 if (size == 0 && !mPinned) { 6725 mState = DESTROYED; 6726 } 6727 6728 return size; 6729} 6730 6731sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6732{ 6733 Mutex::Autolock _l(mLock); 6734 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6735} 6736 6737void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6738{ 6739 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6740 // keep a strong reference on this EffectModule to avoid calling the 6741 // destructor before we exit 6742 sp<EffectModule> keep(this); 6743 { 6744 sp<ThreadBase> thread = mThread.promote(); 6745 if (thread != 0) { 6746 thread->disconnectEffect(keep, handle, unpinIfLast); 6747 } 6748 } 6749} 6750 6751void AudioFlinger::EffectModule::updateState() { 6752 Mutex::Autolock _l(mLock); 6753 6754 switch (mState) { 6755 case RESTART: 6756 reset_l(); 6757 // FALL THROUGH 6758 6759 case STARTING: 6760 // clear auxiliary effect input buffer for next accumulation 6761 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6762 memset(mConfig.inputCfg.buffer.raw, 6763 0, 6764 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6765 } 6766 start_l(); 6767 mState = ACTIVE; 6768 break; 6769 case STOPPING: 6770 stop_l(); 6771 mDisableWaitCnt = mMaxDisableWaitCnt; 6772 mState = STOPPED; 6773 break; 6774 case STOPPED: 6775 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6776 // turn off sequence. 6777 if (--mDisableWaitCnt == 0) { 6778 reset_l(); 6779 mState = IDLE; 6780 } 6781 break; 6782 default: //IDLE , ACTIVE, DESTROYED 6783 break; 6784 } 6785} 6786 6787void AudioFlinger::EffectModule::process() 6788{ 6789 Mutex::Autolock _l(mLock); 6790 6791 if (mState == DESTROYED || mEffectInterface == NULL || 6792 mConfig.inputCfg.buffer.raw == NULL || 6793 mConfig.outputCfg.buffer.raw == NULL) { 6794 return; 6795 } 6796 6797 if (isProcessEnabled()) { 6798 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6799 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6800 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6801 mConfig.inputCfg.buffer.s32, 6802 mConfig.inputCfg.buffer.frameCount/2); 6803 } 6804 6805 // do the actual processing in the effect engine 6806 int ret = (*mEffectInterface)->process(mEffectInterface, 6807 &mConfig.inputCfg.buffer, 6808 &mConfig.outputCfg.buffer); 6809 6810 // force transition to IDLE state when engine is ready 6811 if (mState == STOPPED && ret == -ENODATA) { 6812 mDisableWaitCnt = 1; 6813 } 6814 6815 // clear auxiliary effect input buffer for next accumulation 6816 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6817 memset(mConfig.inputCfg.buffer.raw, 0, 6818 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6819 } 6820 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6821 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6822 // If an insert effect is idle and input buffer is different from output buffer, 6823 // accumulate input onto output 6824 sp<EffectChain> chain = mChain.promote(); 6825 if (chain != 0 && chain->activeTrackCnt() != 0) { 6826 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6827 int16_t *in = mConfig.inputCfg.buffer.s16; 6828 int16_t *out = mConfig.outputCfg.buffer.s16; 6829 for (size_t i = 0; i < frameCnt; i++) { 6830 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6831 } 6832 } 6833 } 6834} 6835 6836void AudioFlinger::EffectModule::reset_l() 6837{ 6838 if (mEffectInterface == NULL) { 6839 return; 6840 } 6841 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6842} 6843 6844status_t AudioFlinger::EffectModule::configure() 6845{ 6846 uint32_t channels; 6847 if (mEffectInterface == NULL) { 6848 return NO_INIT; 6849 } 6850 6851 sp<ThreadBase> thread = mThread.promote(); 6852 if (thread == 0) { 6853 return DEAD_OBJECT; 6854 } 6855 6856 // TODO: handle configuration of effects replacing track process 6857 if (thread->channelCount() == 1) { 6858 channels = AUDIO_CHANNEL_OUT_MONO; 6859 } else { 6860 channels = AUDIO_CHANNEL_OUT_STEREO; 6861 } 6862 6863 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6864 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6865 } else { 6866 mConfig.inputCfg.channels = channels; 6867 } 6868 mConfig.outputCfg.channels = channels; 6869 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6870 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6871 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6872 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6873 mConfig.inputCfg.bufferProvider.cookie = NULL; 6874 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6875 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6876 mConfig.outputCfg.bufferProvider.cookie = NULL; 6877 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6878 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6879 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6880 // Insert effect: 6881 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6882 // always overwrites output buffer: input buffer == output buffer 6883 // - in other sessions: 6884 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6885 // other effect: overwrites output buffer: input buffer == output buffer 6886 // Auxiliary effect: 6887 // accumulates in output buffer: input buffer != output buffer 6888 // Therefore: accumulate <=> input buffer != output buffer 6889 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6890 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6891 } else { 6892 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6893 } 6894 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6895 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6896 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6897 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6898 6899 ALOGV("configure() %p thread %p buffer %p framecount %d", 6900 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6901 6902 status_t cmdStatus; 6903 uint32_t size = sizeof(int); 6904 status_t status = (*mEffectInterface)->command(mEffectInterface, 6905 EFFECT_CMD_SET_CONFIG, 6906 sizeof(effect_config_t), 6907 &mConfig, 6908 &size, 6909 &cmdStatus); 6910 if (status == 0) { 6911 status = cmdStatus; 6912 } 6913 6914 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6915 (1000 * mConfig.outputCfg.buffer.frameCount); 6916 6917 return status; 6918} 6919 6920status_t AudioFlinger::EffectModule::init() 6921{ 6922 Mutex::Autolock _l(mLock); 6923 if (mEffectInterface == NULL) { 6924 return NO_INIT; 6925 } 6926 status_t cmdStatus; 6927 uint32_t size = sizeof(status_t); 6928 status_t status = (*mEffectInterface)->command(mEffectInterface, 6929 EFFECT_CMD_INIT, 6930 0, 6931 NULL, 6932 &size, 6933 &cmdStatus); 6934 if (status == 0) { 6935 status = cmdStatus; 6936 } 6937 return status; 6938} 6939 6940status_t AudioFlinger::EffectModule::start() 6941{ 6942 Mutex::Autolock _l(mLock); 6943 return start_l(); 6944} 6945 6946status_t AudioFlinger::EffectModule::start_l() 6947{ 6948 if (mEffectInterface == NULL) { 6949 return NO_INIT; 6950 } 6951 status_t cmdStatus; 6952 uint32_t size = sizeof(status_t); 6953 status_t status = (*mEffectInterface)->command(mEffectInterface, 6954 EFFECT_CMD_ENABLE, 6955 0, 6956 NULL, 6957 &size, 6958 &cmdStatus); 6959 if (status == 0) { 6960 status = cmdStatus; 6961 } 6962 if (status == 0 && 6963 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6964 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6965 sp<ThreadBase> thread = mThread.promote(); 6966 if (thread != 0) { 6967 audio_stream_t *stream = thread->stream(); 6968 if (stream != NULL) { 6969 stream->add_audio_effect(stream, mEffectInterface); 6970 } 6971 } 6972 } 6973 return status; 6974} 6975 6976status_t AudioFlinger::EffectModule::stop() 6977{ 6978 Mutex::Autolock _l(mLock); 6979 return stop_l(); 6980} 6981 6982status_t AudioFlinger::EffectModule::stop_l() 6983{ 6984 if (mEffectInterface == NULL) { 6985 return NO_INIT; 6986 } 6987 status_t cmdStatus; 6988 uint32_t size = sizeof(status_t); 6989 status_t status = (*mEffectInterface)->command(mEffectInterface, 6990 EFFECT_CMD_DISABLE, 6991 0, 6992 NULL, 6993 &size, 6994 &cmdStatus); 6995 if (status == 0) { 6996 status = cmdStatus; 6997 } 6998 if (status == 0 && 6999 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7000 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7001 sp<ThreadBase> thread = mThread.promote(); 7002 if (thread != 0) { 7003 audio_stream_t *stream = thread->stream(); 7004 if (stream != NULL) { 7005 stream->remove_audio_effect(stream, mEffectInterface); 7006 } 7007 } 7008 } 7009 return status; 7010} 7011 7012status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7013 uint32_t cmdSize, 7014 void *pCmdData, 7015 uint32_t *replySize, 7016 void *pReplyData) 7017{ 7018 Mutex::Autolock _l(mLock); 7019// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7020 7021 if (mState == DESTROYED || mEffectInterface == NULL) { 7022 return NO_INIT; 7023 } 7024 status_t status = (*mEffectInterface)->command(mEffectInterface, 7025 cmdCode, 7026 cmdSize, 7027 pCmdData, 7028 replySize, 7029 pReplyData); 7030 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7031 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7032 for (size_t i = 1; i < mHandles.size(); i++) { 7033 sp<EffectHandle> h = mHandles[i].promote(); 7034 if (h != 0) { 7035 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7036 } 7037 } 7038 } 7039 return status; 7040} 7041 7042status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7043{ 7044 7045 Mutex::Autolock _l(mLock); 7046 ALOGV("setEnabled %p enabled %d", this, enabled); 7047 7048 if (enabled != isEnabled()) { 7049 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7050 if (enabled && status != NO_ERROR) { 7051 return status; 7052 } 7053 7054 switch (mState) { 7055 // going from disabled to enabled 7056 case IDLE: 7057 mState = STARTING; 7058 break; 7059 case STOPPED: 7060 mState = RESTART; 7061 break; 7062 case STOPPING: 7063 mState = ACTIVE; 7064 break; 7065 7066 // going from enabled to disabled 7067 case RESTART: 7068 mState = STOPPED; 7069 break; 7070 case STARTING: 7071 mState = IDLE; 7072 break; 7073 case ACTIVE: 7074 mState = STOPPING; 7075 break; 7076 case DESTROYED: 7077 return NO_ERROR; // simply ignore as we are being destroyed 7078 } 7079 for (size_t i = 1; i < mHandles.size(); i++) { 7080 sp<EffectHandle> h = mHandles[i].promote(); 7081 if (h != 0) { 7082 h->setEnabled(enabled); 7083 } 7084 } 7085 } 7086 return NO_ERROR; 7087} 7088 7089bool AudioFlinger::EffectModule::isEnabled() const 7090{ 7091 switch (mState) { 7092 case RESTART: 7093 case STARTING: 7094 case ACTIVE: 7095 return true; 7096 case IDLE: 7097 case STOPPING: 7098 case STOPPED: 7099 case DESTROYED: 7100 default: 7101 return false; 7102 } 7103} 7104 7105bool AudioFlinger::EffectModule::isProcessEnabled() const 7106{ 7107 switch (mState) { 7108 case RESTART: 7109 case ACTIVE: 7110 case STOPPING: 7111 case STOPPED: 7112 return true; 7113 case IDLE: 7114 case STARTING: 7115 case DESTROYED: 7116 default: 7117 return false; 7118 } 7119} 7120 7121status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7122{ 7123 Mutex::Autolock _l(mLock); 7124 status_t status = NO_ERROR; 7125 7126 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7127 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7128 if (isProcessEnabled() && 7129 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7130 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7131 status_t cmdStatus; 7132 uint32_t volume[2]; 7133 uint32_t *pVolume = NULL; 7134 uint32_t size = sizeof(volume); 7135 volume[0] = *left; 7136 volume[1] = *right; 7137 if (controller) { 7138 pVolume = volume; 7139 } 7140 status = (*mEffectInterface)->command(mEffectInterface, 7141 EFFECT_CMD_SET_VOLUME, 7142 size, 7143 volume, 7144 &size, 7145 pVolume); 7146 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7147 *left = volume[0]; 7148 *right = volume[1]; 7149 } 7150 } 7151 return status; 7152} 7153 7154status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7155{ 7156 Mutex::Autolock _l(mLock); 7157 status_t status = NO_ERROR; 7158 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7159 // audio pre processing modules on RecordThread can receive both output and 7160 // input device indication in the same call 7161 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7162 if (dev) { 7163 status_t cmdStatus; 7164 uint32_t size = sizeof(status_t); 7165 7166 status = (*mEffectInterface)->command(mEffectInterface, 7167 EFFECT_CMD_SET_DEVICE, 7168 sizeof(uint32_t), 7169 &dev, 7170 &size, 7171 &cmdStatus); 7172 if (status == NO_ERROR) { 7173 status = cmdStatus; 7174 } 7175 } 7176 dev = device & AUDIO_DEVICE_IN_ALL; 7177 if (dev) { 7178 status_t cmdStatus; 7179 uint32_t size = sizeof(status_t); 7180 7181 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7182 EFFECT_CMD_SET_INPUT_DEVICE, 7183 sizeof(uint32_t), 7184 &dev, 7185 &size, 7186 &cmdStatus); 7187 if (status2 == NO_ERROR) { 7188 status2 = cmdStatus; 7189 } 7190 if (status == NO_ERROR) { 7191 status = status2; 7192 } 7193 } 7194 } 7195 return status; 7196} 7197 7198status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7199{ 7200 Mutex::Autolock _l(mLock); 7201 status_t status = NO_ERROR; 7202 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7203 status_t cmdStatus; 7204 uint32_t size = sizeof(status_t); 7205 status = (*mEffectInterface)->command(mEffectInterface, 7206 EFFECT_CMD_SET_AUDIO_MODE, 7207 sizeof(audio_mode_t), 7208 &mode, 7209 &size, 7210 &cmdStatus); 7211 if (status == NO_ERROR) { 7212 status = cmdStatus; 7213 } 7214 } 7215 return status; 7216} 7217 7218void AudioFlinger::EffectModule::setSuspended(bool suspended) 7219{ 7220 Mutex::Autolock _l(mLock); 7221 mSuspended = suspended; 7222} 7223 7224bool AudioFlinger::EffectModule::suspended() const 7225{ 7226 Mutex::Autolock _l(mLock); 7227 return mSuspended; 7228} 7229 7230status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7231{ 7232 const size_t SIZE = 256; 7233 char buffer[SIZE]; 7234 String8 result; 7235 7236 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7237 result.append(buffer); 7238 7239 bool locked = tryLock(mLock); 7240 // failed to lock - AudioFlinger is probably deadlocked 7241 if (!locked) { 7242 result.append("\t\tCould not lock Fx mutex:\n"); 7243 } 7244 7245 result.append("\t\tSession Status State Engine:\n"); 7246 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7247 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7248 result.append(buffer); 7249 7250 result.append("\t\tDescriptor:\n"); 7251 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7252 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7253 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7254 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7255 result.append(buffer); 7256 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7257 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7258 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7259 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7260 result.append(buffer); 7261 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7262 mDescriptor.apiVersion, 7263 mDescriptor.flags); 7264 result.append(buffer); 7265 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7266 mDescriptor.name); 7267 result.append(buffer); 7268 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7269 mDescriptor.implementor); 7270 result.append(buffer); 7271 7272 result.append("\t\t- Input configuration:\n"); 7273 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7274 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7275 (uint32_t)mConfig.inputCfg.buffer.raw, 7276 mConfig.inputCfg.buffer.frameCount, 7277 mConfig.inputCfg.samplingRate, 7278 mConfig.inputCfg.channels, 7279 mConfig.inputCfg.format); 7280 result.append(buffer); 7281 7282 result.append("\t\t- Output configuration:\n"); 7283 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7284 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7285 (uint32_t)mConfig.outputCfg.buffer.raw, 7286 mConfig.outputCfg.buffer.frameCount, 7287 mConfig.outputCfg.samplingRate, 7288 mConfig.outputCfg.channels, 7289 mConfig.outputCfg.format); 7290 result.append(buffer); 7291 7292 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7293 result.append(buffer); 7294 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7295 for (size_t i = 0; i < mHandles.size(); ++i) { 7296 sp<EffectHandle> handle = mHandles[i].promote(); 7297 if (handle != 0) { 7298 handle->dump(buffer, SIZE); 7299 result.append(buffer); 7300 } 7301 } 7302 7303 result.append("\n"); 7304 7305 write(fd, result.string(), result.length()); 7306 7307 if (locked) { 7308 mLock.unlock(); 7309 } 7310 7311 return NO_ERROR; 7312} 7313 7314// ---------------------------------------------------------------------------- 7315// EffectHandle implementation 7316// ---------------------------------------------------------------------------- 7317 7318#undef LOG_TAG 7319#define LOG_TAG "AudioFlinger::EffectHandle" 7320 7321AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7322 const sp<AudioFlinger::Client>& client, 7323 const sp<IEffectClient>& effectClient, 7324 int32_t priority) 7325 : BnEffect(), 7326 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7327 mPriority(priority), mHasControl(false), mEnabled(false) 7328{ 7329 ALOGV("constructor %p", this); 7330 7331 if (client == 0) { 7332 return; 7333 } 7334 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7335 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7336 if (mCblkMemory != 0) { 7337 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7338 7339 if (mCblk != NULL) { 7340 new(mCblk) effect_param_cblk_t(); 7341 mBuffer = (uint8_t *)mCblk + bufOffset; 7342 } 7343 } else { 7344 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7345 return; 7346 } 7347} 7348 7349AudioFlinger::EffectHandle::~EffectHandle() 7350{ 7351 ALOGV("Destructor %p", this); 7352 disconnect(false); 7353 ALOGV("Destructor DONE %p", this); 7354} 7355 7356status_t AudioFlinger::EffectHandle::enable() 7357{ 7358 ALOGV("enable %p", this); 7359 if (!mHasControl) return INVALID_OPERATION; 7360 if (mEffect == 0) return DEAD_OBJECT; 7361 7362 if (mEnabled) { 7363 return NO_ERROR; 7364 } 7365 7366 mEnabled = true; 7367 7368 sp<ThreadBase> thread = mEffect->thread().promote(); 7369 if (thread != 0) { 7370 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7371 } 7372 7373 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7374 if (mEffect->suspended()) { 7375 return NO_ERROR; 7376 } 7377 7378 status_t status = mEffect->setEnabled(true); 7379 if (status != NO_ERROR) { 7380 if (thread != 0) { 7381 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7382 } 7383 mEnabled = false; 7384 } 7385 return status; 7386} 7387 7388status_t AudioFlinger::EffectHandle::disable() 7389{ 7390 ALOGV("disable %p", this); 7391 if (!mHasControl) return INVALID_OPERATION; 7392 if (mEffect == 0) return DEAD_OBJECT; 7393 7394 if (!mEnabled) { 7395 return NO_ERROR; 7396 } 7397 mEnabled = false; 7398 7399 if (mEffect->suspended()) { 7400 return NO_ERROR; 7401 } 7402 7403 status_t status = mEffect->setEnabled(false); 7404 7405 sp<ThreadBase> thread = mEffect->thread().promote(); 7406 if (thread != 0) { 7407 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7408 } 7409 7410 return status; 7411} 7412 7413void AudioFlinger::EffectHandle::disconnect() 7414{ 7415 disconnect(true); 7416} 7417 7418void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7419{ 7420 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7421 if (mEffect == 0) { 7422 return; 7423 } 7424 mEffect->disconnect(this, unpinIfLast); 7425 7426 if (mHasControl && mEnabled) { 7427 sp<ThreadBase> thread = mEffect->thread().promote(); 7428 if (thread != 0) { 7429 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7430 } 7431 } 7432 7433 // release sp on module => module destructor can be called now 7434 mEffect.clear(); 7435 if (mClient != 0) { 7436 if (mCblk != NULL) { 7437 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7438 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7439 } 7440 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7441 // Client destructor must run with AudioFlinger mutex locked 7442 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7443 mClient.clear(); 7444 } 7445} 7446 7447status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7448 uint32_t cmdSize, 7449 void *pCmdData, 7450 uint32_t *replySize, 7451 void *pReplyData) 7452{ 7453// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7454// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7455 7456 // only get parameter command is permitted for applications not controlling the effect 7457 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7458 return INVALID_OPERATION; 7459 } 7460 if (mEffect == 0) return DEAD_OBJECT; 7461 if (mClient == 0) return INVALID_OPERATION; 7462 7463 // handle commands that are not forwarded transparently to effect engine 7464 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7465 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7466 // no risk to block the whole media server process or mixer threads is we are stuck here 7467 Mutex::Autolock _l(mCblk->lock); 7468 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7469 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7470 mCblk->serverIndex = 0; 7471 mCblk->clientIndex = 0; 7472 return BAD_VALUE; 7473 } 7474 status_t status = NO_ERROR; 7475 while (mCblk->serverIndex < mCblk->clientIndex) { 7476 int reply; 7477 uint32_t rsize = sizeof(int); 7478 int *p = (int *)(mBuffer + mCblk->serverIndex); 7479 int size = *p++; 7480 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7481 ALOGW("command(): invalid parameter block size"); 7482 break; 7483 } 7484 effect_param_t *param = (effect_param_t *)p; 7485 if (param->psize == 0 || param->vsize == 0) { 7486 ALOGW("command(): null parameter or value size"); 7487 mCblk->serverIndex += size; 7488 continue; 7489 } 7490 uint32_t psize = sizeof(effect_param_t) + 7491 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7492 param->vsize; 7493 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7494 psize, 7495 p, 7496 &rsize, 7497 &reply); 7498 // stop at first error encountered 7499 if (ret != NO_ERROR) { 7500 status = ret; 7501 *(int *)pReplyData = reply; 7502 break; 7503 } else if (reply != NO_ERROR) { 7504 *(int *)pReplyData = reply; 7505 break; 7506 } 7507 mCblk->serverIndex += size; 7508 } 7509 mCblk->serverIndex = 0; 7510 mCblk->clientIndex = 0; 7511 return status; 7512 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7513 *(int *)pReplyData = NO_ERROR; 7514 return enable(); 7515 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7516 *(int *)pReplyData = NO_ERROR; 7517 return disable(); 7518 } 7519 7520 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7521} 7522 7523void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7524{ 7525 ALOGV("setControl %p control %d", this, hasControl); 7526 7527 mHasControl = hasControl; 7528 mEnabled = enabled; 7529 7530 if (signal && mEffectClient != 0) { 7531 mEffectClient->controlStatusChanged(hasControl); 7532 } 7533} 7534 7535void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7536 uint32_t cmdSize, 7537 void *pCmdData, 7538 uint32_t replySize, 7539 void *pReplyData) 7540{ 7541 if (mEffectClient != 0) { 7542 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7543 } 7544} 7545 7546 7547 7548void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7549{ 7550 if (mEffectClient != 0) { 7551 mEffectClient->enableStatusChanged(enabled); 7552 } 7553} 7554 7555status_t AudioFlinger::EffectHandle::onTransact( 7556 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7557{ 7558 return BnEffect::onTransact(code, data, reply, flags); 7559} 7560 7561 7562void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7563{ 7564 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7565 7566 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7567 (mClient == 0) ? getpid_cached : mClient->pid(), 7568 mPriority, 7569 mHasControl, 7570 !locked, 7571 mCblk ? mCblk->clientIndex : 0, 7572 mCblk ? mCblk->serverIndex : 0 7573 ); 7574 7575 if (locked) { 7576 mCblk->lock.unlock(); 7577 } 7578} 7579 7580#undef LOG_TAG 7581#define LOG_TAG "AudioFlinger::EffectChain" 7582 7583AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7584 int sessionId) 7585 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7586 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7587 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7588{ 7589 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7590 sp<ThreadBase> thread = mThread.promote(); 7591 if (thread == 0) { 7592 return; 7593 } 7594 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7595 thread->frameCount(); 7596} 7597 7598AudioFlinger::EffectChain::~EffectChain() 7599{ 7600 if (mOwnInBuffer) { 7601 delete mInBuffer; 7602 } 7603 7604} 7605 7606// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7607sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7608{ 7609 size_t size = mEffects.size(); 7610 7611 for (size_t i = 0; i < size; i++) { 7612 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7613 return mEffects[i]; 7614 } 7615 } 7616 return 0; 7617} 7618 7619// getEffectFromId_l() must be called with ThreadBase::mLock held 7620sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7621{ 7622 size_t size = mEffects.size(); 7623 7624 for (size_t i = 0; i < size; i++) { 7625 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7626 if (id == 0 || mEffects[i]->id() == id) { 7627 return mEffects[i]; 7628 } 7629 } 7630 return 0; 7631} 7632 7633// getEffectFromType_l() must be called with ThreadBase::mLock held 7634sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7635 const effect_uuid_t *type) 7636{ 7637 size_t size = mEffects.size(); 7638 7639 for (size_t i = 0; i < size; i++) { 7640 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7641 return mEffects[i]; 7642 } 7643 } 7644 return 0; 7645} 7646 7647// Must be called with EffectChain::mLock locked 7648void AudioFlinger::EffectChain::process_l() 7649{ 7650 sp<ThreadBase> thread = mThread.promote(); 7651 if (thread == 0) { 7652 ALOGW("process_l(): cannot promote mixer thread"); 7653 return; 7654 } 7655 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7656 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7657 // always process effects unless no more tracks are on the session and the effect tail 7658 // has been rendered 7659 bool doProcess = true; 7660 if (!isGlobalSession) { 7661 bool tracksOnSession = (trackCnt() != 0); 7662 7663 if (!tracksOnSession && mTailBufferCount == 0) { 7664 doProcess = false; 7665 } 7666 7667 if (activeTrackCnt() == 0) { 7668 // if no track is active and the effect tail has not been rendered, 7669 // the input buffer must be cleared here as the mixer process will not do it 7670 if (tracksOnSession || mTailBufferCount > 0) { 7671 size_t numSamples = thread->frameCount() * thread->channelCount(); 7672 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7673 if (mTailBufferCount > 0) { 7674 mTailBufferCount--; 7675 } 7676 } 7677 } 7678 } 7679 7680 size_t size = mEffects.size(); 7681 if (doProcess) { 7682 for (size_t i = 0; i < size; i++) { 7683 mEffects[i]->process(); 7684 } 7685 } 7686 for (size_t i = 0; i < size; i++) { 7687 mEffects[i]->updateState(); 7688 } 7689} 7690 7691// addEffect_l() must be called with PlaybackThread::mLock held 7692status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7693{ 7694 effect_descriptor_t desc = effect->desc(); 7695 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7696 7697 Mutex::Autolock _l(mLock); 7698 effect->setChain(this); 7699 sp<ThreadBase> thread = mThread.promote(); 7700 if (thread == 0) { 7701 return NO_INIT; 7702 } 7703 effect->setThread(thread); 7704 7705 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7706 // Auxiliary effects are inserted at the beginning of mEffects vector as 7707 // they are processed first and accumulated in chain input buffer 7708 mEffects.insertAt(effect, 0); 7709 7710 // the input buffer for auxiliary effect contains mono samples in 7711 // 32 bit format. This is to avoid saturation in AudoMixer 7712 // accumulation stage. Saturation is done in EffectModule::process() before 7713 // calling the process in effect engine 7714 size_t numSamples = thread->frameCount(); 7715 int32_t *buffer = new int32_t[numSamples]; 7716 memset(buffer, 0, numSamples * sizeof(int32_t)); 7717 effect->setInBuffer((int16_t *)buffer); 7718 // auxiliary effects output samples to chain input buffer for further processing 7719 // by insert effects 7720 effect->setOutBuffer(mInBuffer); 7721 } else { 7722 // Insert effects are inserted at the end of mEffects vector as they are processed 7723 // after track and auxiliary effects. 7724 // Insert effect order as a function of indicated preference: 7725 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7726 // another effect is present 7727 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7728 // last effect claiming first position 7729 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7730 // first effect claiming last position 7731 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7732 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7733 // already present 7734 7735 size_t size = mEffects.size(); 7736 size_t idx_insert = size; 7737 ssize_t idx_insert_first = -1; 7738 ssize_t idx_insert_last = -1; 7739 7740 for (size_t i = 0; i < size; i++) { 7741 effect_descriptor_t d = mEffects[i]->desc(); 7742 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7743 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7744 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7745 // check invalid effect chaining combinations 7746 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7747 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7748 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7749 return INVALID_OPERATION; 7750 } 7751 // remember position of first insert effect and by default 7752 // select this as insert position for new effect 7753 if (idx_insert == size) { 7754 idx_insert = i; 7755 } 7756 // remember position of last insert effect claiming 7757 // first position 7758 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7759 idx_insert_first = i; 7760 } 7761 // remember position of first insert effect claiming 7762 // last position 7763 if (iPref == EFFECT_FLAG_INSERT_LAST && 7764 idx_insert_last == -1) { 7765 idx_insert_last = i; 7766 } 7767 } 7768 } 7769 7770 // modify idx_insert from first position if needed 7771 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7772 if (idx_insert_last != -1) { 7773 idx_insert = idx_insert_last; 7774 } else { 7775 idx_insert = size; 7776 } 7777 } else { 7778 if (idx_insert_first != -1) { 7779 idx_insert = idx_insert_first + 1; 7780 } 7781 } 7782 7783 // always read samples from chain input buffer 7784 effect->setInBuffer(mInBuffer); 7785 7786 // if last effect in the chain, output samples to chain 7787 // output buffer, otherwise to chain input buffer 7788 if (idx_insert == size) { 7789 if (idx_insert != 0) { 7790 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7791 mEffects[idx_insert-1]->configure(); 7792 } 7793 effect->setOutBuffer(mOutBuffer); 7794 } else { 7795 effect->setOutBuffer(mInBuffer); 7796 } 7797 mEffects.insertAt(effect, idx_insert); 7798 7799 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7800 } 7801 effect->configure(); 7802 return NO_ERROR; 7803} 7804 7805// removeEffect_l() must be called with PlaybackThread::mLock held 7806size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7807{ 7808 Mutex::Autolock _l(mLock); 7809 size_t size = mEffects.size(); 7810 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7811 7812 for (size_t i = 0; i < size; i++) { 7813 if (effect == mEffects[i]) { 7814 // calling stop here will remove pre-processing effect from the audio HAL. 7815 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7816 // the middle of a read from audio HAL 7817 if (mEffects[i]->state() == EffectModule::ACTIVE || 7818 mEffects[i]->state() == EffectModule::STOPPING) { 7819 mEffects[i]->stop(); 7820 } 7821 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7822 delete[] effect->inBuffer(); 7823 } else { 7824 if (i == size - 1 && i != 0) { 7825 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7826 mEffects[i - 1]->configure(); 7827 } 7828 } 7829 mEffects.removeAt(i); 7830 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7831 break; 7832 } 7833 } 7834 7835 return mEffects.size(); 7836} 7837 7838// setDevice_l() must be called with PlaybackThread::mLock held 7839void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7840{ 7841 size_t size = mEffects.size(); 7842 for (size_t i = 0; i < size; i++) { 7843 mEffects[i]->setDevice(device); 7844 } 7845} 7846 7847// setMode_l() must be called with PlaybackThread::mLock held 7848void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7849{ 7850 size_t size = mEffects.size(); 7851 for (size_t i = 0; i < size; i++) { 7852 mEffects[i]->setMode(mode); 7853 } 7854} 7855 7856// setVolume_l() must be called with PlaybackThread::mLock held 7857bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7858{ 7859 uint32_t newLeft = *left; 7860 uint32_t newRight = *right; 7861 bool hasControl = false; 7862 int ctrlIdx = -1; 7863 size_t size = mEffects.size(); 7864 7865 // first update volume controller 7866 for (size_t i = size; i > 0; i--) { 7867 if (mEffects[i - 1]->isProcessEnabled() && 7868 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7869 ctrlIdx = i - 1; 7870 hasControl = true; 7871 break; 7872 } 7873 } 7874 7875 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7876 if (hasControl) { 7877 *left = mNewLeftVolume; 7878 *right = mNewRightVolume; 7879 } 7880 return hasControl; 7881 } 7882 7883 mVolumeCtrlIdx = ctrlIdx; 7884 mLeftVolume = newLeft; 7885 mRightVolume = newRight; 7886 7887 // second get volume update from volume controller 7888 if (ctrlIdx >= 0) { 7889 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7890 mNewLeftVolume = newLeft; 7891 mNewRightVolume = newRight; 7892 } 7893 // then indicate volume to all other effects in chain. 7894 // Pass altered volume to effects before volume controller 7895 // and requested volume to effects after controller 7896 uint32_t lVol = newLeft; 7897 uint32_t rVol = newRight; 7898 7899 for (size_t i = 0; i < size; i++) { 7900 if ((int)i == ctrlIdx) continue; 7901 // this also works for ctrlIdx == -1 when there is no volume controller 7902 if ((int)i > ctrlIdx) { 7903 lVol = *left; 7904 rVol = *right; 7905 } 7906 mEffects[i]->setVolume(&lVol, &rVol, false); 7907 } 7908 *left = newLeft; 7909 *right = newRight; 7910 7911 return hasControl; 7912} 7913 7914status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7915{ 7916 const size_t SIZE = 256; 7917 char buffer[SIZE]; 7918 String8 result; 7919 7920 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7921 result.append(buffer); 7922 7923 bool locked = tryLock(mLock); 7924 // failed to lock - AudioFlinger is probably deadlocked 7925 if (!locked) { 7926 result.append("\tCould not lock mutex:\n"); 7927 } 7928 7929 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7930 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7931 mEffects.size(), 7932 (uint32_t)mInBuffer, 7933 (uint32_t)mOutBuffer, 7934 mActiveTrackCnt); 7935 result.append(buffer); 7936 write(fd, result.string(), result.size()); 7937 7938 for (size_t i = 0; i < mEffects.size(); ++i) { 7939 sp<EffectModule> effect = mEffects[i]; 7940 if (effect != 0) { 7941 effect->dump(fd, args); 7942 } 7943 } 7944 7945 if (locked) { 7946 mLock.unlock(); 7947 } 7948 7949 return NO_ERROR; 7950} 7951 7952// must be called with ThreadBase::mLock held 7953void AudioFlinger::EffectChain::setEffectSuspended_l( 7954 const effect_uuid_t *type, bool suspend) 7955{ 7956 sp<SuspendedEffectDesc> desc; 7957 // use effect type UUID timelow as key as there is no real risk of identical 7958 // timeLow fields among effect type UUIDs. 7959 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7960 if (suspend) { 7961 if (index >= 0) { 7962 desc = mSuspendedEffects.valueAt(index); 7963 } else { 7964 desc = new SuspendedEffectDesc(); 7965 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7966 mSuspendedEffects.add(type->timeLow, desc); 7967 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7968 } 7969 if (desc->mRefCount++ == 0) { 7970 sp<EffectModule> effect = getEffectIfEnabled(type); 7971 if (effect != 0) { 7972 desc->mEffect = effect; 7973 effect->setSuspended(true); 7974 effect->setEnabled(false); 7975 } 7976 } 7977 } else { 7978 if (index < 0) { 7979 return; 7980 } 7981 desc = mSuspendedEffects.valueAt(index); 7982 if (desc->mRefCount <= 0) { 7983 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7984 desc->mRefCount = 1; 7985 } 7986 if (--desc->mRefCount == 0) { 7987 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7988 if (desc->mEffect != 0) { 7989 sp<EffectModule> effect = desc->mEffect.promote(); 7990 if (effect != 0) { 7991 effect->setSuspended(false); 7992 sp<EffectHandle> handle = effect->controlHandle(); 7993 if (handle != 0) { 7994 effect->setEnabled(handle->enabled()); 7995 } 7996 } 7997 desc->mEffect.clear(); 7998 } 7999 mSuspendedEffects.removeItemsAt(index); 8000 } 8001 } 8002} 8003 8004// must be called with ThreadBase::mLock held 8005void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8006{ 8007 sp<SuspendedEffectDesc> desc; 8008 8009 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8010 if (suspend) { 8011 if (index >= 0) { 8012 desc = mSuspendedEffects.valueAt(index); 8013 } else { 8014 desc = new SuspendedEffectDesc(); 8015 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8016 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8017 } 8018 if (desc->mRefCount++ == 0) { 8019 Vector< sp<EffectModule> > effects; 8020 getSuspendEligibleEffects(effects); 8021 for (size_t i = 0; i < effects.size(); i++) { 8022 setEffectSuspended_l(&effects[i]->desc().type, true); 8023 } 8024 } 8025 } else { 8026 if (index < 0) { 8027 return; 8028 } 8029 desc = mSuspendedEffects.valueAt(index); 8030 if (desc->mRefCount <= 0) { 8031 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8032 desc->mRefCount = 1; 8033 } 8034 if (--desc->mRefCount == 0) { 8035 Vector<const effect_uuid_t *> types; 8036 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8037 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8038 continue; 8039 } 8040 types.add(&mSuspendedEffects.valueAt(i)->mType); 8041 } 8042 for (size_t i = 0; i < types.size(); i++) { 8043 setEffectSuspended_l(types[i], false); 8044 } 8045 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8046 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8047 } 8048 } 8049} 8050 8051 8052// The volume effect is used for automated tests only 8053#ifndef OPENSL_ES_H_ 8054static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8055 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8056const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8057#endif //OPENSL_ES_H_ 8058 8059bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8060{ 8061 // auxiliary effects and visualizer are never suspended on output mix 8062 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8063 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8064 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8065 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8066 return false; 8067 } 8068 return true; 8069} 8070 8071void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8072{ 8073 effects.clear(); 8074 for (size_t i = 0; i < mEffects.size(); i++) { 8075 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8076 effects.add(mEffects[i]); 8077 } 8078 } 8079} 8080 8081sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8082 const effect_uuid_t *type) 8083{ 8084 sp<EffectModule> effect = getEffectFromType_l(type); 8085 return effect != 0 && effect->isEnabled() ? effect : 0; 8086} 8087 8088void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8089 bool enabled) 8090{ 8091 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8092 if (enabled) { 8093 if (index < 0) { 8094 // if the effect is not suspend check if all effects are suspended 8095 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8096 if (index < 0) { 8097 return; 8098 } 8099 if (!isEffectEligibleForSuspend(effect->desc())) { 8100 return; 8101 } 8102 setEffectSuspended_l(&effect->desc().type, enabled); 8103 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8104 if (index < 0) { 8105 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8106 return; 8107 } 8108 } 8109 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8110 effect->desc().type.timeLow); 8111 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8112 // if effect is requested to suspended but was not yet enabled, supend it now. 8113 if (desc->mEffect == 0) { 8114 desc->mEffect = effect; 8115 effect->setEnabled(false); 8116 effect->setSuspended(true); 8117 } 8118 } else { 8119 if (index < 0) { 8120 return; 8121 } 8122 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8123 effect->desc().type.timeLow); 8124 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8125 desc->mEffect.clear(); 8126 effect->setSuspended(false); 8127 } 8128} 8129 8130#undef LOG_TAG 8131#define LOG_TAG "AudioFlinger" 8132 8133// ---------------------------------------------------------------------------- 8134 8135status_t AudioFlinger::onTransact( 8136 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8137{ 8138 return BnAudioFlinger::onTransact(code, data, reply, flags); 8139} 8140 8141}; // namespace android 8142