AudioFlinger.cpp revision 4ff14bae91075eb274eb1c2975982358946e7e63
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); 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 int8_t *curBuf; 2692 size_t mixBufferSize = mFrameCount*mFrameSize; 2693 uint32_t activeSleepTime = activeSleepTimeUs(); 2694 uint32_t idleSleepTime = idleSleepTimeUs(); 2695 uint32_t sleepTime = idleSleepTime; 2696 // use shorter standby delay as on normal output to release 2697 // hardware resources as soon as possible 2698 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2699 2700 acquireWakeLock(); 2701 2702 while (!exitPending()) 2703 { 2704 bool rampVolume; 2705 uint16_t leftVol; 2706 uint16_t rightVol; 2707 Vector< sp<EffectChain> > effectChains; 2708 2709 processConfigEvents(); 2710 2711 mixerStatus = MIXER_IDLE; 2712 2713 { // scope for the mLock 2714 2715 Mutex::Autolock _l(mLock); 2716 2717 if (checkForNewParameters_l()) { 2718 mixBufferSize = mFrameCount*mFrameSize; 2719 activeSleepTime = activeSleepTimeUs(); 2720 idleSleepTime = idleSleepTimeUs(); 2721 standbyDelay = microseconds(activeSleepTime*2); 2722 } 2723 2724 // put audio hardware into standby after short delay 2725 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2726 mSuspended)) { 2727 // wait until we have something to do... 2728 if (!mStandby) { 2729 ALOGV("Audio hardware entering standby, mixer %p", this); 2730 mOutput->stream->common.standby(&mOutput->stream->common); 2731 mStandby = true; 2732 mBytesWritten = 0; 2733 } 2734 2735 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2736 // we're about to wait, flush the binder command buffer 2737 IPCThreadState::self()->flushCommands(); 2738 2739 if (exitPending()) break; 2740 2741 releaseWakeLock_l(); 2742 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2743 mWaitWorkCV.wait(mLock); 2744 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2745 acquireWakeLock_l(); 2746 2747 if (!mMasterMute) { 2748 char value[PROPERTY_VALUE_MAX]; 2749 property_get("ro.audio.silent", value, "0"); 2750 if (atoi(value)) { 2751 ALOGD("Silence is golden"); 2752 setMasterMute_l(true); 2753 } 2754 } 2755 2756 standbyTime = systemTime() + standbyDelay; 2757 sleepTime = idleSleepTime; 2758 continue; 2759 } 2760 } 2761 2762 effectChains = mEffectChains; 2763 2764 // find out which tracks need to be processed 2765 if (mActiveTracks.size() != 0) { 2766 sp<Track> t = mActiveTracks[0].promote(); 2767 if (t == 0) continue; 2768 2769 Track* const track = t.get(); 2770 audio_track_cblk_t* cblk = track->cblk(); 2771 2772 // The first time a track is added we wait 2773 // for all its buffers to be filled before processing it 2774 if (cblk->framesReady() && track->isReady() && 2775 !track->isPaused() && !track->isTerminated()) 2776 { 2777 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2778 2779 if (track->mFillingUpStatus == Track::FS_FILLED) { 2780 track->mFillingUpStatus = Track::FS_ACTIVE; 2781 mLeftVolFloat = mRightVolFloat = 0; 2782 mLeftVolShort = mRightVolShort = 0; 2783 if (track->mState == TrackBase::RESUMING) { 2784 track->mState = TrackBase::ACTIVE; 2785 rampVolume = true; 2786 } 2787 } else if (cblk->server != 0) { 2788 // If the track is stopped before the first frame was mixed, 2789 // do not apply ramp 2790 rampVolume = true; 2791 } 2792 // compute volume for this track 2793 float left, right; 2794 if (track->isMuted() || mMasterMute || track->isPausing() || 2795 mStreamTypes[track->streamType()].mute) { 2796 left = right = 0; 2797 if (track->isPausing()) { 2798 track->setPaused(); 2799 } 2800 } else { 2801 float typeVolume = mStreamTypes[track->streamType()].volume; 2802 float v = mMasterVolume * typeVolume; 2803 uint32_t vlr = cblk->getVolumeLR(); 2804 float v_clamped = v * (vlr & 0xFFFF); 2805 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2806 left = v_clamped/MAX_GAIN; 2807 v_clamped = v * (vlr >> 16); 2808 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2809 right = v_clamped/MAX_GAIN; 2810 } 2811 2812 if (left != mLeftVolFloat || right != mRightVolFloat) { 2813 mLeftVolFloat = left; 2814 mRightVolFloat = right; 2815 2816 // If audio HAL implements volume control, 2817 // force software volume to nominal value 2818 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2819 left = 1.0f; 2820 right = 1.0f; 2821 } 2822 2823 // Convert volumes from float to 8.24 2824 uint32_t vl = (uint32_t)(left * (1 << 24)); 2825 uint32_t vr = (uint32_t)(right * (1 << 24)); 2826 2827 // Delegate volume control to effect in track effect chain if needed 2828 // only one effect chain can be present on DirectOutputThread, so if 2829 // there is one, the track is connected to it 2830 if (!effectChains.isEmpty()) { 2831 // Do not ramp volume if volume is controlled by effect 2832 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2833 rampVolume = false; 2834 } 2835 } 2836 2837 // Convert volumes from 8.24 to 4.12 format 2838 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2839 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2840 leftVol = (uint16_t)v_clamped; 2841 v_clamped = (vr + (1 << 11)) >> 12; 2842 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2843 rightVol = (uint16_t)v_clamped; 2844 } else { 2845 leftVol = mLeftVolShort; 2846 rightVol = mRightVolShort; 2847 rampVolume = false; 2848 } 2849 2850 // reset retry count 2851 track->mRetryCount = kMaxTrackRetriesDirect; 2852 activeTrack = t; 2853 mixerStatus = MIXER_TRACKS_READY; 2854 } else { 2855 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2856 if (track->isStopped()) { 2857 track->reset(); 2858 } 2859 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2860 // We have consumed all the buffers of this track. 2861 // Remove it from the list of active tracks. 2862 trackToRemove = track; 2863 } else { 2864 // No buffers for this track. Give it a few chances to 2865 // fill a buffer, then remove it from active list. 2866 if (--(track->mRetryCount) <= 0) { 2867 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2868 trackToRemove = track; 2869 } else { 2870 mixerStatus = MIXER_TRACKS_ENABLED; 2871 } 2872 } 2873 } 2874 } 2875 2876 // remove all the tracks that need to be... 2877 if (CC_UNLIKELY(trackToRemove != 0)) { 2878 mActiveTracks.remove(trackToRemove); 2879 if (!effectChains.isEmpty()) { 2880 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2881 trackToRemove->sessionId()); 2882 effectChains[0]->decActiveTrackCnt(); 2883 } 2884 if (trackToRemove->isTerminated()) { 2885 removeTrack_l(trackToRemove); 2886 } 2887 } 2888 2889 lockEffectChains_l(effectChains); 2890 } 2891 2892 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2893 AudioBufferProvider::Buffer buffer; 2894 size_t frameCount = mFrameCount; 2895 curBuf = (int8_t *)mMixBuffer; 2896 // output audio to hardware 2897 while (frameCount) { 2898 buffer.frameCount = frameCount; 2899 activeTrack->getNextBuffer(&buffer, 2900 AudioBufferProvider::kInvalidPTS); 2901 if (CC_UNLIKELY(buffer.raw == NULL)) { 2902 memset(curBuf, 0, frameCount * mFrameSize); 2903 break; 2904 } 2905 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2906 frameCount -= buffer.frameCount; 2907 curBuf += buffer.frameCount * mFrameSize; 2908 activeTrack->releaseBuffer(&buffer); 2909 } 2910 sleepTime = 0; 2911 standbyTime = systemTime() + standbyDelay; 2912 } else { 2913 if (sleepTime == 0) { 2914 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2915 sleepTime = activeSleepTime; 2916 } else { 2917 sleepTime = idleSleepTime; 2918 } 2919 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2920 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2921 sleepTime = 0; 2922 } 2923 } 2924 2925 if (mSuspended) { 2926 sleepTime = suspendSleepTimeUs(); 2927 } 2928 // sleepTime == 0 means we must write to audio hardware 2929 if (sleepTime == 0) { 2930 if (mixerStatus == MIXER_TRACKS_READY) { 2931 applyVolume(leftVol, rightVol, rampVolume); 2932 } 2933 for (size_t i = 0; i < effectChains.size(); i ++) { 2934 effectChains[i]->process_l(); 2935 } 2936 unlockEffectChains(effectChains); 2937 2938 mLastWriteTime = systemTime(); 2939 mInWrite = true; 2940 mBytesWritten += mixBufferSize; 2941 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2942 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2943 mNumWrites++; 2944 mInWrite = false; 2945 mStandby = false; 2946 } else { 2947 unlockEffectChains(effectChains); 2948 usleep(sleepTime); 2949 } 2950 2951 // finally let go of removed track, without the lock held 2952 // since we can't guarantee the destructors won't acquire that 2953 // same lock. 2954 trackToRemove.clear(); 2955 activeTrack.clear(); 2956 2957 // Effect chains will be actually deleted here if they were removed from 2958 // mEffectChains list during mixing or effects processing 2959 effectChains.clear(); 2960 } 2961 2962 if (!mStandby) { 2963 mOutput->stream->common.standby(&mOutput->stream->common); 2964 } 2965 2966 releaseWakeLock(); 2967 2968 ALOGV("DirectOutputThread %p exiting", this); 2969 return false; 2970} 2971 2972// getTrackName_l() must be called with ThreadBase::mLock held 2973int AudioFlinger::DirectOutputThread::getTrackName_l() 2974{ 2975 return 0; 2976} 2977 2978// deleteTrackName_l() must be called with ThreadBase::mLock held 2979void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2980{ 2981} 2982 2983// checkForNewParameters_l() must be called with ThreadBase::mLock held 2984bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2985{ 2986 bool reconfig = false; 2987 2988 while (!mNewParameters.isEmpty()) { 2989 status_t status = NO_ERROR; 2990 String8 keyValuePair = mNewParameters[0]; 2991 AudioParameter param = AudioParameter(keyValuePair); 2992 int value; 2993 2994 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2995 // do not accept frame count changes if tracks are open as the track buffer 2996 // size depends on frame count and correct behavior would not be garantied 2997 // if frame count is changed after track creation 2998 if (!mTracks.isEmpty()) { 2999 status = INVALID_OPERATION; 3000 } else { 3001 reconfig = true; 3002 } 3003 } 3004 if (status == NO_ERROR) { 3005 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3006 keyValuePair.string()); 3007 if (!mStandby && status == INVALID_OPERATION) { 3008 mOutput->stream->common.standby(&mOutput->stream->common); 3009 mStandby = true; 3010 mBytesWritten = 0; 3011 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3012 keyValuePair.string()); 3013 } 3014 if (status == NO_ERROR && reconfig) { 3015 readOutputParameters(); 3016 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3017 } 3018 } 3019 3020 mNewParameters.removeAt(0); 3021 3022 mParamStatus = status; 3023 mParamCond.signal(); 3024 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3025 // already timed out waiting for the status and will never signal the condition. 3026 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3027 } 3028 return reconfig; 3029} 3030 3031uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3032{ 3033 uint32_t time; 3034 if (audio_is_linear_pcm(mFormat)) { 3035 time = PlaybackThread::activeSleepTimeUs(); 3036 } else { 3037 time = 10000; 3038 } 3039 return time; 3040} 3041 3042uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3043{ 3044 uint32_t time; 3045 if (audio_is_linear_pcm(mFormat)) { 3046 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3047 } else { 3048 time = 10000; 3049 } 3050 return time; 3051} 3052 3053uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3054{ 3055 uint32_t time; 3056 if (audio_is_linear_pcm(mFormat)) { 3057 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3058 } else { 3059 time = 10000; 3060 } 3061 return time; 3062} 3063 3064 3065// ---------------------------------------------------------------------------- 3066 3067AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3068 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3069 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3070 mWaitTimeMs(UINT_MAX) 3071{ 3072 addOutputTrack(mainThread); 3073} 3074 3075AudioFlinger::DuplicatingThread::~DuplicatingThread() 3076{ 3077 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3078 mOutputTracks[i]->destroy(); 3079 } 3080} 3081 3082bool AudioFlinger::DuplicatingThread::threadLoop() 3083{ 3084 Vector< sp<Track> > tracksToRemove; 3085 mixer_state mixerStatus = MIXER_IDLE; 3086 nsecs_t standbyTime = systemTime(); 3087 size_t mixBufferSize = mFrameCount*mFrameSize; 3088 SortedVector< sp<OutputTrack> > outputTracks; 3089 uint32_t writeFrames = 0; 3090 uint32_t activeSleepTime = activeSleepTimeUs(); 3091 uint32_t idleSleepTime = idleSleepTimeUs(); 3092 uint32_t sleepTime = idleSleepTime; 3093 Vector< sp<EffectChain> > effectChains; 3094 3095 acquireWakeLock(); 3096 3097 while (!exitPending()) 3098 { 3099 processConfigEvents(); 3100 3101 mixerStatus = MIXER_IDLE; 3102 { // scope for the mLock 3103 3104 Mutex::Autolock _l(mLock); 3105 3106 if (checkForNewParameters_l()) { 3107 mixBufferSize = mFrameCount*mFrameSize; 3108 updateWaitTime(); 3109 activeSleepTime = activeSleepTimeUs(); 3110 idleSleepTime = idleSleepTimeUs(); 3111 } 3112 3113 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3114 3115 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3116 outputTracks.add(mOutputTracks[i]); 3117 } 3118 3119 // put audio hardware into standby after short delay 3120 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3121 mSuspended)) { 3122 if (!mStandby) { 3123 for (size_t i = 0; i < outputTracks.size(); i++) { 3124 outputTracks[i]->stop(); 3125 } 3126 mStandby = true; 3127 mBytesWritten = 0; 3128 } 3129 3130 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3131 // we're about to wait, flush the binder command buffer 3132 IPCThreadState::self()->flushCommands(); 3133 outputTracks.clear(); 3134 3135 if (exitPending()) break; 3136 3137 releaseWakeLock_l(); 3138 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3139 mWaitWorkCV.wait(mLock); 3140 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3141 acquireWakeLock_l(); 3142 3143 mPrevMixerStatus = MIXER_IDLE; 3144 if (!mMasterMute) { 3145 char value[PROPERTY_VALUE_MAX]; 3146 property_get("ro.audio.silent", value, "0"); 3147 if (atoi(value)) { 3148 ALOGD("Silence is golden"); 3149 setMasterMute_l(true); 3150 } 3151 } 3152 3153 standbyTime = systemTime() + mStandbyTimeInNsecs; 3154 sleepTime = idleSleepTime; 3155 continue; 3156 } 3157 } 3158 3159 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3160 3161 // prevent any changes in effect chain list and in each effect chain 3162 // during mixing and effect process as the audio buffers could be deleted 3163 // or modified if an effect is created or deleted 3164 lockEffectChains_l(effectChains); 3165 } 3166 3167 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3168 // mix buffers... 3169 if (outputsReady(outputTracks)) { 3170 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3171 } else { 3172 memset(mMixBuffer, 0, mixBufferSize); 3173 } 3174 sleepTime = 0; 3175 writeFrames = mFrameCount; 3176 } else { 3177 if (sleepTime == 0) { 3178 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3179 sleepTime = activeSleepTime; 3180 } else { 3181 sleepTime = idleSleepTime; 3182 } 3183 } else if (mBytesWritten != 0) { 3184 // flush remaining overflow buffers in output tracks 3185 for (size_t i = 0; i < outputTracks.size(); i++) { 3186 if (outputTracks[i]->isActive()) { 3187 sleepTime = 0; 3188 writeFrames = 0; 3189 memset(mMixBuffer, 0, mixBufferSize); 3190 break; 3191 } 3192 } 3193 } 3194 } 3195 3196 if (mSuspended) { 3197 sleepTime = suspendSleepTimeUs(); 3198 } 3199 // sleepTime == 0 means we must write to audio hardware 3200 if (sleepTime == 0) { 3201 for (size_t i = 0; i < effectChains.size(); i ++) { 3202 effectChains[i]->process_l(); 3203 } 3204 // enable changes in effect chain 3205 unlockEffectChains(effectChains); 3206 3207 standbyTime = systemTime() + mStandbyTimeInNsecs; 3208 for (size_t i = 0; i < outputTracks.size(); i++) { 3209 outputTracks[i]->write(mMixBuffer, writeFrames); 3210 } 3211 mStandby = false; 3212 mBytesWritten += mixBufferSize; 3213 } else { 3214 // enable changes in effect chain 3215 unlockEffectChains(effectChains); 3216 usleep(sleepTime); 3217 } 3218 3219 // finally let go of all our tracks, without the lock held 3220 // since we can't guarantee the destructors won't acquire that 3221 // same lock. 3222 tracksToRemove.clear(); 3223 outputTracks.clear(); 3224 3225 // Effect chains will be actually deleted here if they were removed from 3226 // mEffectChains list during mixing or effects processing 3227 effectChains.clear(); 3228 } 3229 3230 releaseWakeLock(); 3231 3232 return false; 3233} 3234 3235void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3236{ 3237 // FIXME explain this formula 3238 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3239 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3240 this, 3241 mSampleRate, 3242 mFormat, 3243 mChannelMask, 3244 frameCount); 3245 if (outputTrack->cblk() != NULL) { 3246 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3247 mOutputTracks.add(outputTrack); 3248 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3249 updateWaitTime(); 3250 } 3251} 3252 3253void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3254{ 3255 Mutex::Autolock _l(mLock); 3256 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3257 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3258 mOutputTracks[i]->destroy(); 3259 mOutputTracks.removeAt(i); 3260 updateWaitTime(); 3261 return; 3262 } 3263 } 3264 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3265} 3266 3267void AudioFlinger::DuplicatingThread::updateWaitTime() 3268{ 3269 mWaitTimeMs = UINT_MAX; 3270 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3271 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3272 if (strong != 0) { 3273 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3274 if (waitTimeMs < mWaitTimeMs) { 3275 mWaitTimeMs = waitTimeMs; 3276 } 3277 } 3278 } 3279} 3280 3281 3282bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3283{ 3284 for (size_t i = 0; i < outputTracks.size(); i++) { 3285 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3286 if (thread == 0) { 3287 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3288 return false; 3289 } 3290 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3291 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3292 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3293 return false; 3294 } 3295 } 3296 return true; 3297} 3298 3299uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3300{ 3301 return (mWaitTimeMs * 1000) / 2; 3302} 3303 3304// ---------------------------------------------------------------------------- 3305 3306// TrackBase constructor must be called with AudioFlinger::mLock held 3307AudioFlinger::ThreadBase::TrackBase::TrackBase( 3308 const wp<ThreadBase>& thread, 3309 const sp<Client>& client, 3310 uint32_t sampleRate, 3311 audio_format_t format, 3312 uint32_t channelMask, 3313 int frameCount, 3314 uint32_t flags, 3315 const sp<IMemory>& sharedBuffer, 3316 int sessionId) 3317 : RefBase(), 3318 mThread(thread), 3319 mClient(client), 3320 mCblk(NULL), 3321 // mBuffer 3322 // mBufferEnd 3323 mFrameCount(0), 3324 mState(IDLE), 3325 mFormat(format), 3326 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3327 mSessionId(sessionId) 3328 // mChannelCount 3329 // mChannelMask 3330{ 3331 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3332 3333 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3334 size_t size = sizeof(audio_track_cblk_t); 3335 uint8_t channelCount = popcount(channelMask); 3336 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3337 if (sharedBuffer == 0) { 3338 size += bufferSize; 3339 } 3340 3341 if (client != NULL) { 3342 mCblkMemory = client->heap()->allocate(size); 3343 if (mCblkMemory != 0) { 3344 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3345 if (mCblk != NULL) { // construct the shared structure in-place. 3346 new(mCblk) audio_track_cblk_t(); 3347 // clear all buffers 3348 mCblk->frameCount = frameCount; 3349 mCblk->sampleRate = sampleRate; 3350 mChannelCount = channelCount; 3351 mChannelMask = channelMask; 3352 if (sharedBuffer == 0) { 3353 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3354 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3355 // Force underrun condition to avoid false underrun callback until first data is 3356 // written to buffer (other flags are cleared) 3357 mCblk->flags = CBLK_UNDERRUN_ON; 3358 } else { 3359 mBuffer = sharedBuffer->pointer(); 3360 } 3361 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3362 } 3363 } else { 3364 ALOGE("not enough memory for AudioTrack size=%u", size); 3365 client->heap()->dump("AudioTrack"); 3366 return; 3367 } 3368 } else { 3369 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3370 // construct the shared structure in-place. 3371 new(mCblk) audio_track_cblk_t(); 3372 // clear all buffers 3373 mCblk->frameCount = frameCount; 3374 mCblk->sampleRate = sampleRate; 3375 mChannelCount = channelCount; 3376 mChannelMask = channelMask; 3377 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3378 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3379 // Force underrun condition to avoid false underrun callback until first data is 3380 // written to buffer (other flags are cleared) 3381 mCblk->flags = CBLK_UNDERRUN_ON; 3382 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3383 } 3384} 3385 3386AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3387{ 3388 if (mCblk != NULL) { 3389 if (mClient == 0) { 3390 delete mCblk; 3391 } else { 3392 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3393 } 3394 } 3395 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3396 if (mClient != 0) { 3397 // Client destructor must run with AudioFlinger mutex locked 3398 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3399 // If the client's reference count drops to zero, the associated destructor 3400 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3401 // relying on the automatic clear() at end of scope. 3402 mClient.clear(); 3403 } 3404} 3405 3406void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3407{ 3408 buffer->raw = NULL; 3409 mFrameCount = buffer->frameCount; 3410 step(); 3411 buffer->frameCount = 0; 3412} 3413 3414bool AudioFlinger::ThreadBase::TrackBase::step() { 3415 bool result; 3416 audio_track_cblk_t* cblk = this->cblk(); 3417 3418 result = cblk->stepServer(mFrameCount); 3419 if (!result) { 3420 ALOGV("stepServer failed acquiring cblk mutex"); 3421 mFlags |= STEPSERVER_FAILED; 3422 } 3423 return result; 3424} 3425 3426void AudioFlinger::ThreadBase::TrackBase::reset() { 3427 audio_track_cblk_t* cblk = this->cblk(); 3428 3429 cblk->user = 0; 3430 cblk->server = 0; 3431 cblk->userBase = 0; 3432 cblk->serverBase = 0; 3433 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3434 ALOGV("TrackBase::reset"); 3435} 3436 3437int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3438 return (int)mCblk->sampleRate; 3439} 3440 3441void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3442 audio_track_cblk_t* cblk = this->cblk(); 3443 size_t frameSize = cblk->frameSize; 3444 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3445 int8_t *bufferEnd = bufferStart + frames * frameSize; 3446 3447 // Check validity of returned pointer in case the track control block would have been corrupted. 3448 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3449 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3450 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3451 server %d, serverBase %d, user %d, userBase %d", 3452 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3453 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3454 return NULL; 3455 } 3456 3457 return bufferStart; 3458} 3459 3460// ---------------------------------------------------------------------------- 3461 3462// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3463AudioFlinger::PlaybackThread::Track::Track( 3464 const wp<ThreadBase>& thread, 3465 const sp<Client>& client, 3466 audio_stream_type_t streamType, 3467 uint32_t sampleRate, 3468 audio_format_t format, 3469 uint32_t channelMask, 3470 int frameCount, 3471 const sp<IMemory>& sharedBuffer, 3472 int sessionId) 3473 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3474 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3475 mAuxEffectId(0), mHasVolumeController(false) 3476{ 3477 if (mCblk != NULL) { 3478 sp<ThreadBase> baseThread = thread.promote(); 3479 if (baseThread != 0) { 3480 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3481 mName = playbackThread->getTrackName_l(); 3482 mMainBuffer = playbackThread->mixBuffer(); 3483 } 3484 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3485 if (mName < 0) { 3486 ALOGE("no more track names available"); 3487 } 3488 mStreamType = streamType; 3489 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3490 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3491 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3492 } 3493} 3494 3495AudioFlinger::PlaybackThread::Track::~Track() 3496{ 3497 ALOGV("PlaybackThread::Track destructor"); 3498 sp<ThreadBase> thread = mThread.promote(); 3499 if (thread != 0) { 3500 Mutex::Autolock _l(thread->mLock); 3501 mState = TERMINATED; 3502 } 3503} 3504 3505void AudioFlinger::PlaybackThread::Track::destroy() 3506{ 3507 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3508 // by removing it from mTracks vector, so there is a risk that this Tracks's 3509 // destructor is called. As the destructor needs to lock mLock, 3510 // we must acquire a strong reference on this Track before locking mLock 3511 // here so that the destructor is called only when exiting this function. 3512 // On the other hand, as long as Track::destroy() is only called by 3513 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3514 // this Track with its member mTrack. 3515 sp<Track> keep(this); 3516 { // scope for mLock 3517 sp<ThreadBase> thread = mThread.promote(); 3518 if (thread != 0) { 3519 if (!isOutputTrack()) { 3520 if (mState == ACTIVE || mState == RESUMING) { 3521 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3522 3523 // to track the speaker usage 3524 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3525 } 3526 AudioSystem::releaseOutput(thread->id()); 3527 } 3528 Mutex::Autolock _l(thread->mLock); 3529 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3530 playbackThread->destroyTrack_l(this); 3531 } 3532 } 3533} 3534 3535void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3536{ 3537 uint32_t vlr = mCblk->getVolumeLR(); 3538 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", 3539 mName - AudioMixer::TRACK0, 3540 (mClient == 0) ? getpid_cached : mClient->pid(), 3541 mStreamType, 3542 mFormat, 3543 mChannelMask, 3544 mSessionId, 3545 mFrameCount, 3546 mState, 3547 mMute, 3548 mFillingUpStatus, 3549 mCblk->sampleRate, 3550 vlr & 0xFFFF, 3551 vlr >> 16, 3552 mCblk->server, 3553 mCblk->user, 3554 (int)mMainBuffer, 3555 (int)mAuxBuffer); 3556} 3557 3558status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3559 AudioBufferProvider::Buffer* buffer, int64_t pts) 3560{ 3561 audio_track_cblk_t* cblk = this->cblk(); 3562 uint32_t framesReady; 3563 uint32_t framesReq = buffer->frameCount; 3564 3565 // Check if last stepServer failed, try to step now 3566 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3567 if (!step()) goto getNextBuffer_exit; 3568 ALOGV("stepServer recovered"); 3569 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3570 } 3571 3572 framesReady = cblk->framesReady(); 3573 3574 if (CC_LIKELY(framesReady)) { 3575 uint32_t s = cblk->server; 3576 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3577 3578 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3579 if (framesReq > framesReady) { 3580 framesReq = framesReady; 3581 } 3582 if (s + framesReq > bufferEnd) { 3583 framesReq = bufferEnd - s; 3584 } 3585 3586 buffer->raw = getBuffer(s, framesReq); 3587 if (buffer->raw == NULL) goto getNextBuffer_exit; 3588 3589 buffer->frameCount = framesReq; 3590 return NO_ERROR; 3591 } 3592 3593getNextBuffer_exit: 3594 buffer->raw = NULL; 3595 buffer->frameCount = 0; 3596 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3597 return NOT_ENOUGH_DATA; 3598} 3599 3600uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3601 return mCblk->framesReady(); 3602} 3603 3604bool AudioFlinger::PlaybackThread::Track::isReady() const { 3605 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3606 3607 if (framesReady() >= mCblk->frameCount || 3608 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3609 mFillingUpStatus = FS_FILLED; 3610 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3611 return true; 3612 } 3613 return false; 3614} 3615 3616status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3617{ 3618 status_t status = NO_ERROR; 3619 ALOGV("start(%d), calling pid %d session %d tid %d", 3620 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3621 sp<ThreadBase> thread = mThread.promote(); 3622 if (thread != 0) { 3623 Mutex::Autolock _l(thread->mLock); 3624 track_state state = mState; 3625 // here the track could be either new, or restarted 3626 // in both cases "unstop" the track 3627 if (mState == PAUSED) { 3628 mState = TrackBase::RESUMING; 3629 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3630 } else { 3631 mState = TrackBase::ACTIVE; 3632 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3633 } 3634 3635 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3636 thread->mLock.unlock(); 3637 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3638 thread->mLock.lock(); 3639 3640 // to track the speaker usage 3641 if (status == NO_ERROR) { 3642 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3643 } 3644 } 3645 if (status == NO_ERROR) { 3646 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3647 playbackThread->addTrack_l(this); 3648 } else { 3649 mState = state; 3650 } 3651 } else { 3652 status = BAD_VALUE; 3653 } 3654 return status; 3655} 3656 3657void AudioFlinger::PlaybackThread::Track::stop() 3658{ 3659 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3660 sp<ThreadBase> thread = mThread.promote(); 3661 if (thread != 0) { 3662 Mutex::Autolock _l(thread->mLock); 3663 track_state state = mState; 3664 if (mState > STOPPED) { 3665 mState = STOPPED; 3666 // If the track is not active (PAUSED and buffers full), flush buffers 3667 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3668 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3669 reset(); 3670 } 3671 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3672 } 3673 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3674 thread->mLock.unlock(); 3675 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3676 thread->mLock.lock(); 3677 3678 // to track the speaker usage 3679 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3680 } 3681 } 3682} 3683 3684void AudioFlinger::PlaybackThread::Track::pause() 3685{ 3686 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3687 sp<ThreadBase> thread = mThread.promote(); 3688 if (thread != 0) { 3689 Mutex::Autolock _l(thread->mLock); 3690 if (mState == ACTIVE || mState == RESUMING) { 3691 mState = PAUSING; 3692 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3693 if (!isOutputTrack()) { 3694 thread->mLock.unlock(); 3695 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3696 thread->mLock.lock(); 3697 3698 // to track the speaker usage 3699 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3700 } 3701 } 3702 } 3703} 3704 3705void AudioFlinger::PlaybackThread::Track::flush() 3706{ 3707 ALOGV("flush(%d)", mName); 3708 sp<ThreadBase> thread = mThread.promote(); 3709 if (thread != 0) { 3710 Mutex::Autolock _l(thread->mLock); 3711 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3712 return; 3713 } 3714 // No point remaining in PAUSED state after a flush => go to 3715 // STOPPED state 3716 mState = STOPPED; 3717 3718 // do not reset the track if it is still in the process of being stopped or paused. 3719 // this will be done by prepareTracks_l() when the track is stopped. 3720 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3721 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3722 reset(); 3723 } 3724 } 3725} 3726 3727void AudioFlinger::PlaybackThread::Track::reset() 3728{ 3729 // Do not reset twice to avoid discarding data written just after a flush and before 3730 // the audioflinger thread detects the track is stopped. 3731 if (!mResetDone) { 3732 TrackBase::reset(); 3733 // Force underrun condition to avoid false underrun callback until first data is 3734 // written to buffer 3735 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3736 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3737 mFillingUpStatus = FS_FILLING; 3738 mResetDone = true; 3739 } 3740} 3741 3742void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3743{ 3744 mMute = muted; 3745} 3746 3747status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3748{ 3749 status_t status = DEAD_OBJECT; 3750 sp<ThreadBase> thread = mThread.promote(); 3751 if (thread != 0) { 3752 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3753 status = playbackThread->attachAuxEffect(this, EffectId); 3754 } 3755 return status; 3756} 3757 3758void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3759{ 3760 mAuxEffectId = EffectId; 3761 mAuxBuffer = buffer; 3762} 3763 3764// timed audio tracks 3765 3766sp<AudioFlinger::PlaybackThread::TimedTrack> 3767AudioFlinger::PlaybackThread::TimedTrack::create( 3768 const wp<ThreadBase>& thread, 3769 const sp<Client>& client, 3770 audio_stream_type_t streamType, 3771 uint32_t sampleRate, 3772 audio_format_t format, 3773 uint32_t channelMask, 3774 int frameCount, 3775 const sp<IMemory>& sharedBuffer, 3776 int sessionId) { 3777 if (!client->reserveTimedTrack()) 3778 return NULL; 3779 3780 sp<TimedTrack> track = new TimedTrack( 3781 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3782 sharedBuffer, sessionId); 3783 3784 if (track == NULL) { 3785 client->releaseTimedTrack(); 3786 return NULL; 3787 } 3788 3789 return track; 3790} 3791 3792AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3793 const wp<ThreadBase>& thread, 3794 const sp<Client>& client, 3795 audio_stream_type_t streamType, 3796 uint32_t sampleRate, 3797 audio_format_t format, 3798 uint32_t channelMask, 3799 int frameCount, 3800 const sp<IMemory>& sharedBuffer, 3801 int sessionId) 3802 : Track(thread, client, streamType, sampleRate, format, channelMask, 3803 frameCount, sharedBuffer, sessionId), 3804 mTimedSilenceBuffer(NULL), 3805 mTimedSilenceBufferSize(0), 3806 mTimedAudioOutputOnTime(false), 3807 mMediaTimeTransformValid(false) 3808{ 3809 LocalClock lc; 3810 mLocalTimeFreq = lc.getLocalFreq(); 3811 3812 mLocalTimeToSampleTransform.a_zero = 0; 3813 mLocalTimeToSampleTransform.b_zero = 0; 3814 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3815 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3816 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3817 &mLocalTimeToSampleTransform.a_to_b_denom); 3818} 3819 3820AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3821 mClient->releaseTimedTrack(); 3822 delete [] mTimedSilenceBuffer; 3823} 3824 3825status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3826 size_t size, sp<IMemory>* buffer) { 3827 3828 Mutex::Autolock _l(mTimedBufferQueueLock); 3829 3830 trimTimedBufferQueue_l(); 3831 3832 // lazily initialize the shared memory heap for timed buffers 3833 if (mTimedMemoryDealer == NULL) { 3834 const int kTimedBufferHeapSize = 512 << 10; 3835 3836 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3837 "AudioFlingerTimed"); 3838 if (mTimedMemoryDealer == NULL) 3839 return NO_MEMORY; 3840 } 3841 3842 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3843 if (newBuffer == NULL) { 3844 newBuffer = mTimedMemoryDealer->allocate(size); 3845 if (newBuffer == NULL) 3846 return NO_MEMORY; 3847 } 3848 3849 *buffer = newBuffer; 3850 return NO_ERROR; 3851} 3852 3853// caller must hold mTimedBufferQueueLock 3854void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3855 int64_t mediaTimeNow; 3856 { 3857 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3858 if (!mMediaTimeTransformValid) 3859 return; 3860 3861 int64_t targetTimeNow; 3862 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3863 ? mCCHelper.getCommonTime(&targetTimeNow) 3864 : mCCHelper.getLocalTime(&targetTimeNow); 3865 3866 if (OK != res) 3867 return; 3868 3869 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3870 &mediaTimeNow)) { 3871 return; 3872 } 3873 } 3874 3875 size_t trimIndex; 3876 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3877 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3878 break; 3879 } 3880 3881 if (trimIndex) { 3882 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3883 } 3884} 3885 3886status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3887 const sp<IMemory>& buffer, int64_t pts) { 3888 3889 { 3890 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3891 if (!mMediaTimeTransformValid) 3892 return INVALID_OPERATION; 3893 } 3894 3895 Mutex::Autolock _l(mTimedBufferQueueLock); 3896 3897 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3898 3899 return NO_ERROR; 3900} 3901 3902status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3903 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3904 3905 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3906 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3907 target); 3908 3909 if (!(target == TimedAudioTrack::LOCAL_TIME || 3910 target == TimedAudioTrack::COMMON_TIME)) { 3911 return BAD_VALUE; 3912 } 3913 3914 Mutex::Autolock lock(mMediaTimeTransformLock); 3915 mMediaTimeTransform = xform; 3916 mMediaTimeTransformTarget = target; 3917 mMediaTimeTransformValid = true; 3918 3919 return NO_ERROR; 3920} 3921 3922#define min(a, b) ((a) < (b) ? (a) : (b)) 3923 3924// implementation of getNextBuffer for tracks whose buffers have timestamps 3925status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3926 AudioBufferProvider::Buffer* buffer, int64_t pts) 3927{ 3928 if (pts == AudioBufferProvider::kInvalidPTS) { 3929 buffer->raw = 0; 3930 buffer->frameCount = 0; 3931 return INVALID_OPERATION; 3932 } 3933 3934 // get ahold of the output stream that these samples will be written to 3935 sp<ThreadBase> thread = mThread.promote(); 3936 if (thread == NULL) { 3937 buffer->raw = 0; 3938 buffer->frameCount = 0; 3939 return INVALID_OPERATION; 3940 } 3941 PlaybackThread* playbackThread = static_cast<PlaybackThread*>(thread.get()); 3942 3943 Mutex::Autolock _l(mTimedBufferQueueLock); 3944 3945 while (true) { 3946 3947 // if we have no timed buffers, then fail 3948 if (mTimedBufferQueue.isEmpty()) { 3949 buffer->raw = 0; 3950 buffer->frameCount = 0; 3951 return NOT_ENOUGH_DATA; 3952 } 3953 3954 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3955 3956 // calculate the PTS of the head of the timed buffer queue expressed in 3957 // local time 3958 int64_t headLocalPTS; 3959 { 3960 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3961 3962 assert(mMediaTimeTransformValid); 3963 3964 if (mMediaTimeTransform.a_to_b_denom == 0) { 3965 // the transform represents a pause, so yield silence 3966 timedYieldSilence(buffer->frameCount, buffer); 3967 return NO_ERROR; 3968 } 3969 3970 int64_t transformedPTS; 3971 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3972 &transformedPTS)) { 3973 // the transform failed. this shouldn't happen, but if it does 3974 // then just drop this buffer 3975 ALOGW("timedGetNextBuffer transform failed"); 3976 buffer->raw = 0; 3977 buffer->frameCount = 0; 3978 mTimedBufferQueue.removeAt(0); 3979 return NO_ERROR; 3980 } 3981 3982 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3983 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3984 &headLocalPTS)) { 3985 buffer->raw = 0; 3986 buffer->frameCount = 0; 3987 return INVALID_OPERATION; 3988 } 3989 } else { 3990 headLocalPTS = transformedPTS; 3991 } 3992 } 3993 3994 // adjust the head buffer's PTS to reflect the portion of the head buffer 3995 // that has already been consumed 3996 int64_t effectivePTS = headLocalPTS + 3997 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3998 3999 // Calculate the delta in samples between the head of the input buffer 4000 // queue and the start of the next output buffer that will be written. 4001 // If the transformation fails because of over or underflow, it means 4002 // that the sample's position in the output stream is so far out of 4003 // whack that it should just be dropped. 4004 int64_t sampleDelta; 4005 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4006 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4007 mTimedBufferQueue.removeAt(0); 4008 continue; 4009 } 4010 if (!mLocalTimeToSampleTransform.doForwardTransform( 4011 (effectivePTS - pts) << 32, &sampleDelta)) { 4012 ALOGV("*** too late during sample rate transform: dropped buffer"); 4013 mTimedBufferQueue.removeAt(0); 4014 continue; 4015 } 4016 4017 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4018 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4019 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4020 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4021 4022 // if the delta between the ideal placement for the next input sample and 4023 // the current output position is within this threshold, then we will 4024 // concatenate the next input samples to the previous output 4025 const int64_t kSampleContinuityThreshold = 4026 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4027 4028 // if this is the first buffer of audio that we're emitting from this track 4029 // then it should be almost exactly on time. 4030 const int64_t kSampleStartupThreshold = 1LL << 32; 4031 4032 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4033 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4034 // the next input is close enough to being on time, so concatenate it 4035 // with the last output 4036 timedYieldSamples(buffer); 4037 4038 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4039 return NO_ERROR; 4040 } else if (sampleDelta > 0) { 4041 // the gap between the current output position and the proper start of 4042 // the next input sample is too big, so fill it with silence 4043 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4044 4045 timedYieldSilence(framesUntilNextInput, buffer); 4046 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4047 return NO_ERROR; 4048 } else { 4049 // the next input sample is late 4050 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4051 size_t onTimeSamplePosition = 4052 head.position() + lateFrames * mCblk->frameSize; 4053 4054 if (onTimeSamplePosition > head.buffer()->size()) { 4055 // all the remaining samples in the head are too late, so 4056 // drop it and move on 4057 ALOGV("*** too late: dropped buffer"); 4058 mTimedBufferQueue.removeAt(0); 4059 continue; 4060 } else { 4061 // skip over the late samples 4062 head.setPosition(onTimeSamplePosition); 4063 4064 // yield the available samples 4065 timedYieldSamples(buffer); 4066 4067 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4068 return NO_ERROR; 4069 } 4070 } 4071 } 4072} 4073 4074// Yield samples from the timed buffer queue head up to the given output 4075// buffer's capacity. 4076// 4077// Caller must hold mTimedBufferQueueLock 4078void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4079 AudioBufferProvider::Buffer* buffer) { 4080 4081 const TimedBuffer& head = mTimedBufferQueue[0]; 4082 4083 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4084 head.position()); 4085 4086 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4087 mCblk->frameSize); 4088 size_t framesRequested = buffer->frameCount; 4089 buffer->frameCount = min(framesLeftInHead, framesRequested); 4090 4091 mTimedAudioOutputOnTime = true; 4092} 4093 4094// Yield samples of silence up to the given output buffer's capacity 4095// 4096// Caller must hold mTimedBufferQueueLock 4097void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4098 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4099 4100 // lazily allocate a buffer filled with silence 4101 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4102 delete [] mTimedSilenceBuffer; 4103 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4104 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4105 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4106 } 4107 4108 buffer->raw = mTimedSilenceBuffer; 4109 size_t framesRequested = buffer->frameCount; 4110 buffer->frameCount = min(numFrames, framesRequested); 4111 4112 mTimedAudioOutputOnTime = false; 4113} 4114 4115void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4116 AudioBufferProvider::Buffer* buffer) { 4117 4118 Mutex::Autolock _l(mTimedBufferQueueLock); 4119 4120 if (buffer->raw != mTimedSilenceBuffer) { 4121 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4122 head.setPosition(head.position() + buffer->frameCount * mCblk->frameSize); 4123 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4124 mTimedBufferQueue.removeAt(0); 4125 } 4126 } 4127 4128 buffer->raw = 0; 4129 buffer->frameCount = 0; 4130} 4131 4132uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4133 Mutex::Autolock _l(mTimedBufferQueueLock); 4134 4135 uint32_t frames = 0; 4136 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4137 const TimedBuffer& tb = mTimedBufferQueue[i]; 4138 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4139 } 4140 4141 return frames; 4142} 4143 4144AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4145 : mPTS(0), mPosition(0) {} 4146 4147AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4148 const sp<IMemory>& buffer, int64_t pts) 4149 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4150 4151// ---------------------------------------------------------------------------- 4152 4153// RecordTrack constructor must be called with AudioFlinger::mLock held 4154AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4155 const wp<ThreadBase>& thread, 4156 const sp<Client>& client, 4157 uint32_t sampleRate, 4158 audio_format_t format, 4159 uint32_t channelMask, 4160 int frameCount, 4161 uint32_t flags, 4162 int sessionId) 4163 : TrackBase(thread, client, sampleRate, format, 4164 channelMask, frameCount, flags, 0, sessionId), 4165 mOverflow(false) 4166{ 4167 if (mCblk != NULL) { 4168 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4169 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4170 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4171 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4172 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4173 } else { 4174 mCblk->frameSize = sizeof(int8_t); 4175 } 4176 } 4177} 4178 4179AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4180{ 4181 sp<ThreadBase> thread = mThread.promote(); 4182 if (thread != 0) { 4183 AudioSystem::releaseInput(thread->id()); 4184 } 4185} 4186 4187status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4188{ 4189 audio_track_cblk_t* cblk = this->cblk(); 4190 uint32_t framesAvail; 4191 uint32_t framesReq = buffer->frameCount; 4192 4193 // Check if last stepServer failed, try to step now 4194 if (mFlags & TrackBase::STEPSERVER_FAILED) { 4195 if (!step()) goto getNextBuffer_exit; 4196 ALOGV("stepServer recovered"); 4197 mFlags &= ~TrackBase::STEPSERVER_FAILED; 4198 } 4199 4200 framesAvail = cblk->framesAvailable_l(); 4201 4202 if (CC_LIKELY(framesAvail)) { 4203 uint32_t s = cblk->server; 4204 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4205 4206 if (framesReq > framesAvail) { 4207 framesReq = framesAvail; 4208 } 4209 if (s + framesReq > bufferEnd) { 4210 framesReq = bufferEnd - s; 4211 } 4212 4213 buffer->raw = getBuffer(s, framesReq); 4214 if (buffer->raw == NULL) goto getNextBuffer_exit; 4215 4216 buffer->frameCount = framesReq; 4217 return NO_ERROR; 4218 } 4219 4220getNextBuffer_exit: 4221 buffer->raw = NULL; 4222 buffer->frameCount = 0; 4223 return NOT_ENOUGH_DATA; 4224} 4225 4226status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4227{ 4228 sp<ThreadBase> thread = mThread.promote(); 4229 if (thread != 0) { 4230 RecordThread *recordThread = (RecordThread *)thread.get(); 4231 return recordThread->start(this, tid); 4232 } else { 4233 return BAD_VALUE; 4234 } 4235} 4236 4237void AudioFlinger::RecordThread::RecordTrack::stop() 4238{ 4239 sp<ThreadBase> thread = mThread.promote(); 4240 if (thread != 0) { 4241 RecordThread *recordThread = (RecordThread *)thread.get(); 4242 recordThread->stop(this); 4243 TrackBase::reset(); 4244 // Force overerrun condition to avoid false overrun callback until first data is 4245 // read from buffer 4246 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4247 } 4248} 4249 4250void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4251{ 4252 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4253 (mClient == 0) ? getpid_cached : mClient->pid(), 4254 mFormat, 4255 mChannelMask, 4256 mSessionId, 4257 mFrameCount, 4258 mState, 4259 mCblk->sampleRate, 4260 mCblk->server, 4261 mCblk->user); 4262} 4263 4264 4265// ---------------------------------------------------------------------------- 4266 4267AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4268 const wp<ThreadBase>& thread, 4269 DuplicatingThread *sourceThread, 4270 uint32_t sampleRate, 4271 audio_format_t format, 4272 uint32_t channelMask, 4273 int frameCount) 4274 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4275 mActive(false), mSourceThread(sourceThread) 4276{ 4277 4278 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 4279 if (mCblk != NULL) { 4280 mCblk->flags |= CBLK_DIRECTION_OUT; 4281 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4282 mOutBuffer.frameCount = 0; 4283 playbackThread->mTracks.add(this); 4284 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4285 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4286 mCblk, mBuffer, mCblk->buffers, 4287 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4288 } else { 4289 ALOGW("Error creating output track on thread %p", playbackThread); 4290 } 4291} 4292 4293AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4294{ 4295 clearBufferQueue(); 4296} 4297 4298status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4299{ 4300 status_t status = Track::start(tid); 4301 if (status != NO_ERROR) { 4302 return status; 4303 } 4304 4305 mActive = true; 4306 mRetryCount = 127; 4307 return status; 4308} 4309 4310void AudioFlinger::PlaybackThread::OutputTrack::stop() 4311{ 4312 Track::stop(); 4313 clearBufferQueue(); 4314 mOutBuffer.frameCount = 0; 4315 mActive = false; 4316} 4317 4318bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4319{ 4320 Buffer *pInBuffer; 4321 Buffer inBuffer; 4322 uint32_t channelCount = mChannelCount; 4323 bool outputBufferFull = false; 4324 inBuffer.frameCount = frames; 4325 inBuffer.i16 = data; 4326 4327 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4328 4329 if (!mActive && frames != 0) { 4330 start(0); 4331 sp<ThreadBase> thread = mThread.promote(); 4332 if (thread != 0) { 4333 MixerThread *mixerThread = (MixerThread *)thread.get(); 4334 if (mCblk->frameCount > frames){ 4335 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4336 uint32_t startFrames = (mCblk->frameCount - frames); 4337 pInBuffer = new Buffer; 4338 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4339 pInBuffer->frameCount = startFrames; 4340 pInBuffer->i16 = pInBuffer->mBuffer; 4341 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4342 mBufferQueue.add(pInBuffer); 4343 } else { 4344 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4345 } 4346 } 4347 } 4348 } 4349 4350 while (waitTimeLeftMs) { 4351 // First write pending buffers, then new data 4352 if (mBufferQueue.size()) { 4353 pInBuffer = mBufferQueue.itemAt(0); 4354 } else { 4355 pInBuffer = &inBuffer; 4356 } 4357 4358 if (pInBuffer->frameCount == 0) { 4359 break; 4360 } 4361 4362 if (mOutBuffer.frameCount == 0) { 4363 mOutBuffer.frameCount = pInBuffer->frameCount; 4364 nsecs_t startTime = systemTime(); 4365 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4366 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4367 outputBufferFull = true; 4368 break; 4369 } 4370 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4371 if (waitTimeLeftMs >= waitTimeMs) { 4372 waitTimeLeftMs -= waitTimeMs; 4373 } else { 4374 waitTimeLeftMs = 0; 4375 } 4376 } 4377 4378 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4379 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4380 mCblk->stepUser(outFrames); 4381 pInBuffer->frameCount -= outFrames; 4382 pInBuffer->i16 += outFrames * channelCount; 4383 mOutBuffer.frameCount -= outFrames; 4384 mOutBuffer.i16 += outFrames * channelCount; 4385 4386 if (pInBuffer->frameCount == 0) { 4387 if (mBufferQueue.size()) { 4388 mBufferQueue.removeAt(0); 4389 delete [] pInBuffer->mBuffer; 4390 delete pInBuffer; 4391 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4392 } else { 4393 break; 4394 } 4395 } 4396 } 4397 4398 // If we could not write all frames, allocate a buffer and queue it for next time. 4399 if (inBuffer.frameCount) { 4400 sp<ThreadBase> thread = mThread.promote(); 4401 if (thread != 0 && !thread->standby()) { 4402 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4403 pInBuffer = new Buffer; 4404 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4405 pInBuffer->frameCount = inBuffer.frameCount; 4406 pInBuffer->i16 = pInBuffer->mBuffer; 4407 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4408 mBufferQueue.add(pInBuffer); 4409 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4410 } else { 4411 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4412 } 4413 } 4414 } 4415 4416 // Calling write() with a 0 length buffer, means that no more data will be written: 4417 // If no more buffers are pending, fill output track buffer to make sure it is started 4418 // by output mixer. 4419 if (frames == 0 && mBufferQueue.size() == 0) { 4420 if (mCblk->user < mCblk->frameCount) { 4421 frames = mCblk->frameCount - mCblk->user; 4422 pInBuffer = new Buffer; 4423 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4424 pInBuffer->frameCount = frames; 4425 pInBuffer->i16 = pInBuffer->mBuffer; 4426 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4427 mBufferQueue.add(pInBuffer); 4428 } else if (mActive) { 4429 stop(); 4430 } 4431 } 4432 4433 return outputBufferFull; 4434} 4435 4436status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4437{ 4438 int active; 4439 status_t result; 4440 audio_track_cblk_t* cblk = mCblk; 4441 uint32_t framesReq = buffer->frameCount; 4442 4443// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4444 buffer->frameCount = 0; 4445 4446 uint32_t framesAvail = cblk->framesAvailable(); 4447 4448 4449 if (framesAvail == 0) { 4450 Mutex::Autolock _l(cblk->lock); 4451 goto start_loop_here; 4452 while (framesAvail == 0) { 4453 active = mActive; 4454 if (CC_UNLIKELY(!active)) { 4455 ALOGV("Not active and NO_MORE_BUFFERS"); 4456 return NO_MORE_BUFFERS; 4457 } 4458 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4459 if (result != NO_ERROR) { 4460 return NO_MORE_BUFFERS; 4461 } 4462 // read the server count again 4463 start_loop_here: 4464 framesAvail = cblk->framesAvailable_l(); 4465 } 4466 } 4467 4468// if (framesAvail < framesReq) { 4469// return NO_MORE_BUFFERS; 4470// } 4471 4472 if (framesReq > framesAvail) { 4473 framesReq = framesAvail; 4474 } 4475 4476 uint32_t u = cblk->user; 4477 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4478 4479 if (u + framesReq > bufferEnd) { 4480 framesReq = bufferEnd - u; 4481 } 4482 4483 buffer->frameCount = framesReq; 4484 buffer->raw = (void *)cblk->buffer(u); 4485 return NO_ERROR; 4486} 4487 4488 4489void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4490{ 4491 size_t size = mBufferQueue.size(); 4492 Buffer *pBuffer; 4493 4494 for (size_t i = 0; i < size; i++) { 4495 pBuffer = mBufferQueue.itemAt(i); 4496 delete [] pBuffer->mBuffer; 4497 delete pBuffer; 4498 } 4499 mBufferQueue.clear(); 4500} 4501 4502// ---------------------------------------------------------------------------- 4503 4504AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4505 : RefBase(), 4506 mAudioFlinger(audioFlinger), 4507 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4508 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4509 mPid(pid), 4510 mTimedTrackCount(0) 4511{ 4512 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4513} 4514 4515// Client destructor must be called with AudioFlinger::mLock held 4516AudioFlinger::Client::~Client() 4517{ 4518 mAudioFlinger->removeClient_l(mPid); 4519} 4520 4521sp<MemoryDealer> AudioFlinger::Client::heap() const 4522{ 4523 return mMemoryDealer; 4524} 4525 4526// Reserve one of the limited slots for a timed audio track associated 4527// with this client 4528bool AudioFlinger::Client::reserveTimedTrack() 4529{ 4530 const int kMaxTimedTracksPerClient = 4; 4531 4532 Mutex::Autolock _l(mTimedTrackLock); 4533 4534 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4535 ALOGW("can not create timed track - pid %d has exceeded the limit", 4536 mPid); 4537 return false; 4538 } 4539 4540 mTimedTrackCount++; 4541 return true; 4542} 4543 4544// Release a slot for a timed audio track 4545void AudioFlinger::Client::releaseTimedTrack() 4546{ 4547 Mutex::Autolock _l(mTimedTrackLock); 4548 mTimedTrackCount--; 4549} 4550 4551// ---------------------------------------------------------------------------- 4552 4553AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4554 const sp<IAudioFlingerClient>& client, 4555 pid_t pid) 4556 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4557{ 4558} 4559 4560AudioFlinger::NotificationClient::~NotificationClient() 4561{ 4562} 4563 4564void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4565{ 4566 sp<NotificationClient> keep(this); 4567 { 4568 mAudioFlinger->removeNotificationClient(mPid); 4569 } 4570} 4571 4572// ---------------------------------------------------------------------------- 4573 4574AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4575 : BnAudioTrack(), 4576 mTrack(track) 4577{ 4578} 4579 4580AudioFlinger::TrackHandle::~TrackHandle() { 4581 // just stop the track on deletion, associated resources 4582 // will be freed from the main thread once all pending buffers have 4583 // been played. Unless it's not in the active track list, in which 4584 // case we free everything now... 4585 mTrack->destroy(); 4586} 4587 4588sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4589 return mTrack->getCblk(); 4590} 4591 4592status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4593 return mTrack->start(tid); 4594} 4595 4596void AudioFlinger::TrackHandle::stop() { 4597 mTrack->stop(); 4598} 4599 4600void AudioFlinger::TrackHandle::flush() { 4601 mTrack->flush(); 4602} 4603 4604void AudioFlinger::TrackHandle::mute(bool e) { 4605 mTrack->mute(e); 4606} 4607 4608void AudioFlinger::TrackHandle::pause() { 4609 mTrack->pause(); 4610} 4611 4612status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4613{ 4614 return mTrack->attachAuxEffect(EffectId); 4615} 4616 4617status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4618 sp<IMemory>* buffer) { 4619 if (!mTrack->isTimedTrack()) 4620 return INVALID_OPERATION; 4621 4622 PlaybackThread::TimedTrack* tt = 4623 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4624 return tt->allocateTimedBuffer(size, buffer); 4625} 4626 4627status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4628 int64_t pts) { 4629 if (!mTrack->isTimedTrack()) 4630 return INVALID_OPERATION; 4631 4632 PlaybackThread::TimedTrack* tt = 4633 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4634 return tt->queueTimedBuffer(buffer, pts); 4635} 4636 4637status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4638 const LinearTransform& xform, int target) { 4639 4640 if (!mTrack->isTimedTrack()) 4641 return INVALID_OPERATION; 4642 4643 PlaybackThread::TimedTrack* tt = 4644 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4645 return tt->setMediaTimeTransform( 4646 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4647} 4648 4649status_t AudioFlinger::TrackHandle::onTransact( 4650 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4651{ 4652 return BnAudioTrack::onTransact(code, data, reply, flags); 4653} 4654 4655// ---------------------------------------------------------------------------- 4656 4657sp<IAudioRecord> AudioFlinger::openRecord( 4658 pid_t pid, 4659 audio_io_handle_t input, 4660 uint32_t sampleRate, 4661 audio_format_t format, 4662 uint32_t channelMask, 4663 int frameCount, 4664 uint32_t flags, 4665 int *sessionId, 4666 status_t *status) 4667{ 4668 sp<RecordThread::RecordTrack> recordTrack; 4669 sp<RecordHandle> recordHandle; 4670 sp<Client> client; 4671 status_t lStatus; 4672 RecordThread *thread; 4673 size_t inFrameCount; 4674 int lSessionId; 4675 4676 // check calling permissions 4677 if (!recordingAllowed()) { 4678 lStatus = PERMISSION_DENIED; 4679 goto Exit; 4680 } 4681 4682 // add client to list 4683 { // scope for mLock 4684 Mutex::Autolock _l(mLock); 4685 thread = checkRecordThread_l(input); 4686 if (thread == NULL) { 4687 lStatus = BAD_VALUE; 4688 goto Exit; 4689 } 4690 4691 client = registerPid_l(pid); 4692 4693 // If no audio session id is provided, create one here 4694 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4695 lSessionId = *sessionId; 4696 } else { 4697 lSessionId = nextUniqueId(); 4698 if (sessionId != NULL) { 4699 *sessionId = lSessionId; 4700 } 4701 } 4702 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4703 recordTrack = thread->createRecordTrack_l(client, 4704 sampleRate, 4705 format, 4706 channelMask, 4707 frameCount, 4708 flags, 4709 lSessionId, 4710 &lStatus); 4711 } 4712 if (lStatus != NO_ERROR) { 4713 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4714 // destructor is called by the TrackBase destructor with mLock held 4715 client.clear(); 4716 recordTrack.clear(); 4717 goto Exit; 4718 } 4719 4720 // return to handle to client 4721 recordHandle = new RecordHandle(recordTrack); 4722 lStatus = NO_ERROR; 4723 4724Exit: 4725 if (status) { 4726 *status = lStatus; 4727 } 4728 return recordHandle; 4729} 4730 4731// ---------------------------------------------------------------------------- 4732 4733AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4734 : BnAudioRecord(), 4735 mRecordTrack(recordTrack) 4736{ 4737} 4738 4739AudioFlinger::RecordHandle::~RecordHandle() { 4740 stop(); 4741} 4742 4743sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4744 return mRecordTrack->getCblk(); 4745} 4746 4747status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4748 ALOGV("RecordHandle::start()"); 4749 return mRecordTrack->start(tid); 4750} 4751 4752void AudioFlinger::RecordHandle::stop() { 4753 ALOGV("RecordHandle::stop()"); 4754 mRecordTrack->stop(); 4755} 4756 4757status_t AudioFlinger::RecordHandle::onTransact( 4758 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4759{ 4760 return BnAudioRecord::onTransact(code, data, reply, flags); 4761} 4762 4763// ---------------------------------------------------------------------------- 4764 4765AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4766 AudioStreamIn *input, 4767 uint32_t sampleRate, 4768 uint32_t channels, 4769 audio_io_handle_t id, 4770 uint32_t device) : 4771 ThreadBase(audioFlinger, id, device, RECORD), 4772 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4773 // mRsmpInIndex and mInputBytes set by readInputParameters() 4774 mReqChannelCount(popcount(channels)), 4775 mReqSampleRate(sampleRate) 4776 // mBytesRead is only meaningful while active, and so is cleared in start() 4777 // (but might be better to also clear here for dump?) 4778{ 4779 snprintf(mName, kNameLength, "AudioIn_%d", id); 4780 4781 readInputParameters(); 4782} 4783 4784 4785AudioFlinger::RecordThread::~RecordThread() 4786{ 4787 delete[] mRsmpInBuffer; 4788 delete mResampler; 4789 delete[] mRsmpOutBuffer; 4790} 4791 4792void AudioFlinger::RecordThread::onFirstRef() 4793{ 4794 run(mName, PRIORITY_URGENT_AUDIO); 4795} 4796 4797status_t AudioFlinger::RecordThread::readyToRun() 4798{ 4799 status_t status = initCheck(); 4800 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4801 return status; 4802} 4803 4804bool AudioFlinger::RecordThread::threadLoop() 4805{ 4806 AudioBufferProvider::Buffer buffer; 4807 sp<RecordTrack> activeTrack; 4808 Vector< sp<EffectChain> > effectChains; 4809 4810 nsecs_t lastWarning = 0; 4811 4812 acquireWakeLock(); 4813 4814 // start recording 4815 while (!exitPending()) { 4816 4817 processConfigEvents(); 4818 4819 { // scope for mLock 4820 Mutex::Autolock _l(mLock); 4821 checkForNewParameters_l(); 4822 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4823 if (!mStandby) { 4824 mInput->stream->common.standby(&mInput->stream->common); 4825 mStandby = true; 4826 } 4827 4828 if (exitPending()) break; 4829 4830 releaseWakeLock_l(); 4831 ALOGV("RecordThread: loop stopping"); 4832 // go to sleep 4833 mWaitWorkCV.wait(mLock); 4834 ALOGV("RecordThread: loop starting"); 4835 acquireWakeLock_l(); 4836 continue; 4837 } 4838 if (mActiveTrack != 0) { 4839 if (mActiveTrack->mState == TrackBase::PAUSING) { 4840 if (!mStandby) { 4841 mInput->stream->common.standby(&mInput->stream->common); 4842 mStandby = true; 4843 } 4844 mActiveTrack.clear(); 4845 mStartStopCond.broadcast(); 4846 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4847 if (mReqChannelCount != mActiveTrack->channelCount()) { 4848 mActiveTrack.clear(); 4849 mStartStopCond.broadcast(); 4850 } else if (mBytesRead != 0) { 4851 // record start succeeds only if first read from audio input 4852 // succeeds 4853 if (mBytesRead > 0) { 4854 mActiveTrack->mState = TrackBase::ACTIVE; 4855 } else { 4856 mActiveTrack.clear(); 4857 } 4858 mStartStopCond.broadcast(); 4859 } 4860 mStandby = false; 4861 } 4862 } 4863 lockEffectChains_l(effectChains); 4864 } 4865 4866 if (mActiveTrack != 0) { 4867 if (mActiveTrack->mState != TrackBase::ACTIVE && 4868 mActiveTrack->mState != TrackBase::RESUMING) { 4869 unlockEffectChains(effectChains); 4870 usleep(kRecordThreadSleepUs); 4871 continue; 4872 } 4873 for (size_t i = 0; i < effectChains.size(); i ++) { 4874 effectChains[i]->process_l(); 4875 } 4876 4877 buffer.frameCount = mFrameCount; 4878 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4879 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4880 size_t framesOut = buffer.frameCount; 4881 if (mResampler == NULL) { 4882 // no resampling 4883 while (framesOut) { 4884 size_t framesIn = mFrameCount - mRsmpInIndex; 4885 if (framesIn) { 4886 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4887 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4888 if (framesIn > framesOut) 4889 framesIn = framesOut; 4890 mRsmpInIndex += framesIn; 4891 framesOut -= framesIn; 4892 if ((int)mChannelCount == mReqChannelCount || 4893 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4894 memcpy(dst, src, framesIn * mFrameSize); 4895 } else { 4896 int16_t *src16 = (int16_t *)src; 4897 int16_t *dst16 = (int16_t *)dst; 4898 if (mChannelCount == 1) { 4899 while (framesIn--) { 4900 *dst16++ = *src16; 4901 *dst16++ = *src16++; 4902 } 4903 } else { 4904 while (framesIn--) { 4905 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4906 src16 += 2; 4907 } 4908 } 4909 } 4910 } 4911 if (framesOut && mFrameCount == mRsmpInIndex) { 4912 if (framesOut == mFrameCount && 4913 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4914 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4915 framesOut = 0; 4916 } else { 4917 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4918 mRsmpInIndex = 0; 4919 } 4920 if (mBytesRead < 0) { 4921 ALOGE("Error reading audio input"); 4922 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4923 // Force input into standby so that it tries to 4924 // recover at next read attempt 4925 mInput->stream->common.standby(&mInput->stream->common); 4926 usleep(kRecordThreadSleepUs); 4927 } 4928 mRsmpInIndex = mFrameCount; 4929 framesOut = 0; 4930 buffer.frameCount = 0; 4931 } 4932 } 4933 } 4934 } else { 4935 // resampling 4936 4937 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4938 // alter output frame count as if we were expecting stereo samples 4939 if (mChannelCount == 1 && mReqChannelCount == 1) { 4940 framesOut >>= 1; 4941 } 4942 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4943 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4944 // are 32 bit aligned which should be always true. 4945 if (mChannelCount == 2 && mReqChannelCount == 1) { 4946 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4947 // the resampler always outputs stereo samples: do post stereo to mono conversion 4948 int16_t *src = (int16_t *)mRsmpOutBuffer; 4949 int16_t *dst = buffer.i16; 4950 while (framesOut--) { 4951 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4952 src += 2; 4953 } 4954 } else { 4955 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4956 } 4957 4958 } 4959 mActiveTrack->releaseBuffer(&buffer); 4960 mActiveTrack->overflow(); 4961 } 4962 // client isn't retrieving buffers fast enough 4963 else { 4964 if (!mActiveTrack->setOverflow()) { 4965 nsecs_t now = systemTime(); 4966 if ((now - lastWarning) > kWarningThrottleNs) { 4967 ALOGW("RecordThread: buffer overflow"); 4968 lastWarning = now; 4969 } 4970 } 4971 // Release the processor for a while before asking for a new buffer. 4972 // This will give the application more chance to read from the buffer and 4973 // clear the overflow. 4974 usleep(kRecordThreadSleepUs); 4975 } 4976 } 4977 // enable changes in effect chain 4978 unlockEffectChains(effectChains); 4979 effectChains.clear(); 4980 } 4981 4982 if (!mStandby) { 4983 mInput->stream->common.standby(&mInput->stream->common); 4984 } 4985 mActiveTrack.clear(); 4986 4987 mStartStopCond.broadcast(); 4988 4989 releaseWakeLock(); 4990 4991 ALOGV("RecordThread %p exiting", this); 4992 return false; 4993} 4994 4995 4996sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4997 const sp<AudioFlinger::Client>& client, 4998 uint32_t sampleRate, 4999 audio_format_t format, 5000 int channelMask, 5001 int frameCount, 5002 uint32_t flags, 5003 int sessionId, 5004 status_t *status) 5005{ 5006 sp<RecordTrack> track; 5007 status_t lStatus; 5008 5009 lStatus = initCheck(); 5010 if (lStatus != NO_ERROR) { 5011 ALOGE("Audio driver not initialized."); 5012 goto Exit; 5013 } 5014 5015 { // scope for mLock 5016 Mutex::Autolock _l(mLock); 5017 5018 track = new RecordTrack(this, client, sampleRate, 5019 format, channelMask, frameCount, flags, sessionId); 5020 5021 if (track->getCblk() == 0) { 5022 lStatus = NO_MEMORY; 5023 goto Exit; 5024 } 5025 5026 mTrack = track.get(); 5027 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5028 bool suspend = audio_is_bluetooth_sco_device( 5029 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5030 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5031 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5032 } 5033 lStatus = NO_ERROR; 5034 5035Exit: 5036 if (status) { 5037 *status = lStatus; 5038 } 5039 return track; 5040} 5041 5042status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5043{ 5044 ALOGV("RecordThread::start tid=%d", tid); 5045 sp <ThreadBase> strongMe = this; 5046 status_t status = NO_ERROR; 5047 { 5048 AutoMutex lock(mLock); 5049 if (mActiveTrack != 0) { 5050 if (recordTrack != mActiveTrack.get()) { 5051 status = -EBUSY; 5052 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5053 mActiveTrack->mState = TrackBase::ACTIVE; 5054 } 5055 return status; 5056 } 5057 5058 recordTrack->mState = TrackBase::IDLE; 5059 mActiveTrack = recordTrack; 5060 mLock.unlock(); 5061 status_t status = AudioSystem::startInput(mId); 5062 mLock.lock(); 5063 if (status != NO_ERROR) { 5064 mActiveTrack.clear(); 5065 return status; 5066 } 5067 mRsmpInIndex = mFrameCount; 5068 mBytesRead = 0; 5069 if (mResampler != NULL) { 5070 mResampler->reset(); 5071 } 5072 mActiveTrack->mState = TrackBase::RESUMING; 5073 // signal thread to start 5074 ALOGV("Signal record thread"); 5075 mWaitWorkCV.signal(); 5076 // do not wait for mStartStopCond if exiting 5077 if (exitPending()) { 5078 mActiveTrack.clear(); 5079 status = INVALID_OPERATION; 5080 goto startError; 5081 } 5082 mStartStopCond.wait(mLock); 5083 if (mActiveTrack == 0) { 5084 ALOGV("Record failed to start"); 5085 status = BAD_VALUE; 5086 goto startError; 5087 } 5088 ALOGV("Record started OK"); 5089 return status; 5090 } 5091startError: 5092 AudioSystem::stopInput(mId); 5093 return status; 5094} 5095 5096void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5097 ALOGV("RecordThread::stop"); 5098 sp <ThreadBase> strongMe = this; 5099 { 5100 AutoMutex lock(mLock); 5101 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5102 mActiveTrack->mState = TrackBase::PAUSING; 5103 // do not wait for mStartStopCond if exiting 5104 if (exitPending()) { 5105 return; 5106 } 5107 mStartStopCond.wait(mLock); 5108 // if we have been restarted, recordTrack == mActiveTrack.get() here 5109 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5110 mLock.unlock(); 5111 AudioSystem::stopInput(mId); 5112 mLock.lock(); 5113 ALOGV("Record stopped OK"); 5114 } 5115 } 5116 } 5117} 5118 5119status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5120{ 5121 const size_t SIZE = 256; 5122 char buffer[SIZE]; 5123 String8 result; 5124 5125 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5126 result.append(buffer); 5127 5128 if (mActiveTrack != 0) { 5129 result.append("Active Track:\n"); 5130 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5131 mActiveTrack->dump(buffer, SIZE); 5132 result.append(buffer); 5133 5134 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5135 result.append(buffer); 5136 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5137 result.append(buffer); 5138 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5139 result.append(buffer); 5140 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5141 result.append(buffer); 5142 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5143 result.append(buffer); 5144 5145 5146 } else { 5147 result.append("No record client\n"); 5148 } 5149 write(fd, result.string(), result.size()); 5150 5151 dumpBase(fd, args); 5152 dumpEffectChains(fd, args); 5153 5154 return NO_ERROR; 5155} 5156 5157status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5158{ 5159 size_t framesReq = buffer->frameCount; 5160 size_t framesReady = mFrameCount - mRsmpInIndex; 5161 int channelCount; 5162 5163 if (framesReady == 0) { 5164 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5165 if (mBytesRead < 0) { 5166 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5167 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5168 // Force input into standby so that it tries to 5169 // recover at next read attempt 5170 mInput->stream->common.standby(&mInput->stream->common); 5171 usleep(kRecordThreadSleepUs); 5172 } 5173 buffer->raw = NULL; 5174 buffer->frameCount = 0; 5175 return NOT_ENOUGH_DATA; 5176 } 5177 mRsmpInIndex = 0; 5178 framesReady = mFrameCount; 5179 } 5180 5181 if (framesReq > framesReady) { 5182 framesReq = framesReady; 5183 } 5184 5185 if (mChannelCount == 1 && mReqChannelCount == 2) { 5186 channelCount = 1; 5187 } else { 5188 channelCount = 2; 5189 } 5190 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5191 buffer->frameCount = framesReq; 5192 return NO_ERROR; 5193} 5194 5195void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5196{ 5197 mRsmpInIndex += buffer->frameCount; 5198 buffer->frameCount = 0; 5199} 5200 5201bool AudioFlinger::RecordThread::checkForNewParameters_l() 5202{ 5203 bool reconfig = false; 5204 5205 while (!mNewParameters.isEmpty()) { 5206 status_t status = NO_ERROR; 5207 String8 keyValuePair = mNewParameters[0]; 5208 AudioParameter param = AudioParameter(keyValuePair); 5209 int value; 5210 audio_format_t reqFormat = mFormat; 5211 int reqSamplingRate = mReqSampleRate; 5212 int reqChannelCount = mReqChannelCount; 5213 5214 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5215 reqSamplingRate = value; 5216 reconfig = true; 5217 } 5218 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5219 reqFormat = (audio_format_t) value; 5220 reconfig = true; 5221 } 5222 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5223 reqChannelCount = popcount(value); 5224 reconfig = true; 5225 } 5226 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5227 // do not accept frame count changes if tracks are open as the track buffer 5228 // size depends on frame count and correct behavior would not be guaranteed 5229 // if frame count is changed after track creation 5230 if (mActiveTrack != 0) { 5231 status = INVALID_OPERATION; 5232 } else { 5233 reconfig = true; 5234 } 5235 } 5236 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5237 // forward device change to effects that have requested to be 5238 // aware of attached audio device. 5239 for (size_t i = 0; i < mEffectChains.size(); i++) { 5240 mEffectChains[i]->setDevice_l(value); 5241 } 5242 // store input device and output device but do not forward output device to audio HAL. 5243 // Note that status is ignored by the caller for output device 5244 // (see AudioFlinger::setParameters() 5245 if (value & AUDIO_DEVICE_OUT_ALL) { 5246 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5247 status = BAD_VALUE; 5248 } else { 5249 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5250 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5251 if (mTrack != NULL) { 5252 bool suspend = audio_is_bluetooth_sco_device( 5253 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5254 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5255 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5256 } 5257 } 5258 mDevice |= (uint32_t)value; 5259 } 5260 if (status == NO_ERROR) { 5261 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5262 if (status == INVALID_OPERATION) { 5263 mInput->stream->common.standby(&mInput->stream->common); 5264 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5265 } 5266 if (reconfig) { 5267 if (status == BAD_VALUE && 5268 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5269 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5270 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5271 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5272 (reqChannelCount < 3)) { 5273 status = NO_ERROR; 5274 } 5275 if (status == NO_ERROR) { 5276 readInputParameters(); 5277 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5278 } 5279 } 5280 } 5281 5282 mNewParameters.removeAt(0); 5283 5284 mParamStatus = status; 5285 mParamCond.signal(); 5286 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5287 // already timed out waiting for the status and will never signal the condition. 5288 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5289 } 5290 return reconfig; 5291} 5292 5293String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5294{ 5295 char *s; 5296 String8 out_s8 = String8(); 5297 5298 Mutex::Autolock _l(mLock); 5299 if (initCheck() != NO_ERROR) { 5300 return out_s8; 5301 } 5302 5303 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5304 out_s8 = String8(s); 5305 free(s); 5306 return out_s8; 5307} 5308 5309void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5310 AudioSystem::OutputDescriptor desc; 5311 void *param2 = NULL; 5312 5313 switch (event) { 5314 case AudioSystem::INPUT_OPENED: 5315 case AudioSystem::INPUT_CONFIG_CHANGED: 5316 desc.channels = mChannelMask; 5317 desc.samplingRate = mSampleRate; 5318 desc.format = mFormat; 5319 desc.frameCount = mFrameCount; 5320 desc.latency = 0; 5321 param2 = &desc; 5322 break; 5323 5324 case AudioSystem::INPUT_CLOSED: 5325 default: 5326 break; 5327 } 5328 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5329} 5330 5331void AudioFlinger::RecordThread::readInputParameters() 5332{ 5333 delete mRsmpInBuffer; 5334 // mRsmpInBuffer is always assigned a new[] below 5335 delete mRsmpOutBuffer; 5336 mRsmpOutBuffer = NULL; 5337 delete mResampler; 5338 mResampler = NULL; 5339 5340 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5341 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5342 mChannelCount = (uint16_t)popcount(mChannelMask); 5343 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5344 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5345 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5346 mFrameCount = mInputBytes / mFrameSize; 5347 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5348 5349 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5350 { 5351 int channelCount; 5352 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5353 // stereo to mono post process as the resampler always outputs stereo. 5354 if (mChannelCount == 1 && mReqChannelCount == 2) { 5355 channelCount = 1; 5356 } else { 5357 channelCount = 2; 5358 } 5359 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5360 mResampler->setSampleRate(mSampleRate); 5361 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5362 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5363 5364 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5365 if (mChannelCount == 1 && mReqChannelCount == 1) { 5366 mFrameCount >>= 1; 5367 } 5368 5369 } 5370 mRsmpInIndex = mFrameCount; 5371} 5372 5373unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5374{ 5375 Mutex::Autolock _l(mLock); 5376 if (initCheck() != NO_ERROR) { 5377 return 0; 5378 } 5379 5380 return mInput->stream->get_input_frames_lost(mInput->stream); 5381} 5382 5383uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5384{ 5385 Mutex::Autolock _l(mLock); 5386 uint32_t result = 0; 5387 if (getEffectChain_l(sessionId) != 0) { 5388 result = EFFECT_SESSION; 5389 } 5390 5391 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5392 result |= TRACK_SESSION; 5393 } 5394 5395 return result; 5396} 5397 5398AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5399{ 5400 Mutex::Autolock _l(mLock); 5401 return mTrack; 5402} 5403 5404AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5405{ 5406 Mutex::Autolock _l(mLock); 5407 return mInput; 5408} 5409 5410AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5411{ 5412 Mutex::Autolock _l(mLock); 5413 AudioStreamIn *input = mInput; 5414 mInput = NULL; 5415 return input; 5416} 5417 5418// this method must always be called either with ThreadBase mLock held or inside the thread loop 5419audio_stream_t* AudioFlinger::RecordThread::stream() 5420{ 5421 if (mInput == NULL) { 5422 return NULL; 5423 } 5424 return &mInput->stream->common; 5425} 5426 5427 5428// ---------------------------------------------------------------------------- 5429 5430audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5431 uint32_t *pSamplingRate, 5432 audio_format_t *pFormat, 5433 uint32_t *pChannels, 5434 uint32_t *pLatencyMs, 5435 uint32_t flags) 5436{ 5437 status_t status; 5438 PlaybackThread *thread = NULL; 5439 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5440 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5441 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5442 uint32_t channels = pChannels ? *pChannels : 0; 5443 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5444 audio_stream_out_t *outStream; 5445 audio_hw_device_t *outHwDev; 5446 5447 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5448 pDevices ? *pDevices : 0, 5449 samplingRate, 5450 format, 5451 channels, 5452 flags); 5453 5454 if (pDevices == NULL || *pDevices == 0) { 5455 return 0; 5456 } 5457 5458 Mutex::Autolock _l(mLock); 5459 5460 outHwDev = findSuitableHwDev_l(*pDevices); 5461 if (outHwDev == NULL) 5462 return 0; 5463 5464 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5465 &channels, &samplingRate, &outStream); 5466 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5467 outStream, 5468 samplingRate, 5469 format, 5470 channels, 5471 status); 5472 5473 mHardwareStatus = AUDIO_HW_IDLE; 5474 if (outStream != NULL) { 5475 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5476 audio_io_handle_t id = nextUniqueId(); 5477 5478 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5479 (format != AUDIO_FORMAT_PCM_16_BIT) || 5480 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5481 thread = new DirectOutputThread(this, output, id, *pDevices); 5482 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5483 } else { 5484 thread = new MixerThread(this, output, id, *pDevices); 5485 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5486 } 5487 mPlaybackThreads.add(id, thread); 5488 5489 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5490 if (pFormat != NULL) *pFormat = format; 5491 if (pChannels != NULL) *pChannels = channels; 5492 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5493 5494 // notify client processes of the new output creation 5495 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5496 return id; 5497 } 5498 5499 return 0; 5500} 5501 5502audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5503 audio_io_handle_t output2) 5504{ 5505 Mutex::Autolock _l(mLock); 5506 MixerThread *thread1 = checkMixerThread_l(output1); 5507 MixerThread *thread2 = checkMixerThread_l(output2); 5508 5509 if (thread1 == NULL || thread2 == NULL) { 5510 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5511 return 0; 5512 } 5513 5514 audio_io_handle_t id = nextUniqueId(); 5515 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5516 thread->addOutputTrack(thread2); 5517 mPlaybackThreads.add(id, thread); 5518 // notify client processes of the new output creation 5519 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5520 return id; 5521} 5522 5523status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5524{ 5525 // keep strong reference on the playback thread so that 5526 // it is not destroyed while exit() is executed 5527 sp <PlaybackThread> thread; 5528 { 5529 Mutex::Autolock _l(mLock); 5530 thread = checkPlaybackThread_l(output); 5531 if (thread == NULL) { 5532 return BAD_VALUE; 5533 } 5534 5535 ALOGV("closeOutput() %d", output); 5536 5537 if (thread->type() == ThreadBase::MIXER) { 5538 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5539 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5540 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5541 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5542 } 5543 } 5544 } 5545 void *param2 = NULL; 5546 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5547 mPlaybackThreads.removeItem(output); 5548 } 5549 thread->exit(); 5550 // The thread entity (active unit of execution) is no longer running here, 5551 // but the ThreadBase container still exists. 5552 5553 if (thread->type() != ThreadBase::DUPLICATING) { 5554 AudioStreamOut *out = thread->clearOutput(); 5555 assert(out != NULL); 5556 // from now on thread->mOutput is NULL 5557 out->hwDev->close_output_stream(out->hwDev, out->stream); 5558 delete out; 5559 } 5560 return NO_ERROR; 5561} 5562 5563status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5564{ 5565 Mutex::Autolock _l(mLock); 5566 PlaybackThread *thread = checkPlaybackThread_l(output); 5567 5568 if (thread == NULL) { 5569 return BAD_VALUE; 5570 } 5571 5572 ALOGV("suspendOutput() %d", output); 5573 thread->suspend(); 5574 5575 return NO_ERROR; 5576} 5577 5578status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5579{ 5580 Mutex::Autolock _l(mLock); 5581 PlaybackThread *thread = checkPlaybackThread_l(output); 5582 5583 if (thread == NULL) { 5584 return BAD_VALUE; 5585 } 5586 5587 ALOGV("restoreOutput() %d", output); 5588 5589 thread->restore(); 5590 5591 return NO_ERROR; 5592} 5593 5594audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5595 uint32_t *pSamplingRate, 5596 audio_format_t *pFormat, 5597 uint32_t *pChannels, 5598 audio_in_acoustics_t acoustics) 5599{ 5600 status_t status; 5601 RecordThread *thread = NULL; 5602 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5603 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5604 uint32_t channels = pChannels ? *pChannels : 0; 5605 uint32_t reqSamplingRate = samplingRate; 5606 audio_format_t reqFormat = format; 5607 uint32_t reqChannels = channels; 5608 audio_stream_in_t *inStream; 5609 audio_hw_device_t *inHwDev; 5610 5611 if (pDevices == NULL || *pDevices == 0) { 5612 return 0; 5613 } 5614 5615 Mutex::Autolock _l(mLock); 5616 5617 inHwDev = findSuitableHwDev_l(*pDevices); 5618 if (inHwDev == NULL) 5619 return 0; 5620 5621 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5622 &channels, &samplingRate, 5623 acoustics, 5624 &inStream); 5625 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5626 inStream, 5627 samplingRate, 5628 format, 5629 channels, 5630 acoustics, 5631 status); 5632 5633 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5634 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5635 // or stereo to mono conversions on 16 bit PCM inputs. 5636 if (inStream == NULL && status == BAD_VALUE && 5637 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5638 (samplingRate <= 2 * reqSamplingRate) && 5639 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5640 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5641 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5642 &channels, &samplingRate, 5643 acoustics, 5644 &inStream); 5645 } 5646 5647 if (inStream != NULL) { 5648 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5649 5650 audio_io_handle_t id = nextUniqueId(); 5651 // Start record thread 5652 // RecorThread require both input and output device indication to forward to audio 5653 // pre processing modules 5654 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5655 thread = new RecordThread(this, 5656 input, 5657 reqSamplingRate, 5658 reqChannels, 5659 id, 5660 device); 5661 mRecordThreads.add(id, thread); 5662 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5663 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5664 if (pFormat != NULL) *pFormat = format; 5665 if (pChannels != NULL) *pChannels = reqChannels; 5666 5667 input->stream->common.standby(&input->stream->common); 5668 5669 // notify client processes of the new input creation 5670 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5671 return id; 5672 } 5673 5674 return 0; 5675} 5676 5677status_t AudioFlinger::closeInput(audio_io_handle_t input) 5678{ 5679 // keep strong reference on the record thread so that 5680 // it is not destroyed while exit() is executed 5681 sp <RecordThread> thread; 5682 { 5683 Mutex::Autolock _l(mLock); 5684 thread = checkRecordThread_l(input); 5685 if (thread == NULL) { 5686 return BAD_VALUE; 5687 } 5688 5689 ALOGV("closeInput() %d", input); 5690 void *param2 = NULL; 5691 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5692 mRecordThreads.removeItem(input); 5693 } 5694 thread->exit(); 5695 // The thread entity (active unit of execution) is no longer running here, 5696 // but the ThreadBase container still exists. 5697 5698 AudioStreamIn *in = thread->clearInput(); 5699 assert(in != NULL); 5700 // from now on thread->mInput is NULL 5701 in->hwDev->close_input_stream(in->hwDev, in->stream); 5702 delete in; 5703 5704 return NO_ERROR; 5705} 5706 5707status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5708{ 5709 Mutex::Autolock _l(mLock); 5710 MixerThread *dstThread = checkMixerThread_l(output); 5711 if (dstThread == NULL) { 5712 ALOGW("setStreamOutput() bad output id %d", output); 5713 return BAD_VALUE; 5714 } 5715 5716 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5717 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5718 5719 dstThread->setStreamValid(stream, true); 5720 5721 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5722 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5723 if (thread != dstThread && 5724 thread->type() != ThreadBase::DIRECT) { 5725 MixerThread *srcThread = (MixerThread *)thread; 5726 srcThread->setStreamValid(stream, false); 5727 srcThread->invalidateTracks(stream); 5728 } 5729 } 5730 5731 return NO_ERROR; 5732} 5733 5734 5735int AudioFlinger::newAudioSessionId() 5736{ 5737 return nextUniqueId(); 5738} 5739 5740void AudioFlinger::acquireAudioSessionId(int audioSession) 5741{ 5742 Mutex::Autolock _l(mLock); 5743 pid_t caller = IPCThreadState::self()->getCallingPid(); 5744 ALOGV("acquiring %d from %d", audioSession, caller); 5745 size_t num = mAudioSessionRefs.size(); 5746 for (size_t i = 0; i< num; i++) { 5747 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5748 if (ref->sessionid == audioSession && ref->pid == caller) { 5749 ref->cnt++; 5750 ALOGV(" incremented refcount to %d", ref->cnt); 5751 return; 5752 } 5753 } 5754 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5755 ALOGV(" added new entry for %d", audioSession); 5756} 5757 5758void AudioFlinger::releaseAudioSessionId(int audioSession) 5759{ 5760 Mutex::Autolock _l(mLock); 5761 pid_t caller = IPCThreadState::self()->getCallingPid(); 5762 ALOGV("releasing %d from %d", audioSession, caller); 5763 size_t num = mAudioSessionRefs.size(); 5764 for (size_t i = 0; i< num; i++) { 5765 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5766 if (ref->sessionid == audioSession && ref->pid == caller) { 5767 ref->cnt--; 5768 ALOGV(" decremented refcount to %d", ref->cnt); 5769 if (ref->cnt == 0) { 5770 mAudioSessionRefs.removeAt(i); 5771 delete ref; 5772 purgeStaleEffects_l(); 5773 } 5774 return; 5775 } 5776 } 5777 ALOGW("session id %d not found for pid %d", audioSession, caller); 5778} 5779 5780void AudioFlinger::purgeStaleEffects_l() { 5781 5782 ALOGV("purging stale effects"); 5783 5784 Vector< sp<EffectChain> > chains; 5785 5786 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5787 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5788 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5789 sp<EffectChain> ec = t->mEffectChains[j]; 5790 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5791 chains.push(ec); 5792 } 5793 } 5794 } 5795 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5796 sp<RecordThread> t = mRecordThreads.valueAt(i); 5797 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5798 sp<EffectChain> ec = t->mEffectChains[j]; 5799 chains.push(ec); 5800 } 5801 } 5802 5803 for (size_t i = 0; i < chains.size(); i++) { 5804 sp<EffectChain> ec = chains[i]; 5805 int sessionid = ec->sessionId(); 5806 sp<ThreadBase> t = ec->mThread.promote(); 5807 if (t == 0) { 5808 continue; 5809 } 5810 size_t numsessionrefs = mAudioSessionRefs.size(); 5811 bool found = false; 5812 for (size_t k = 0; k < numsessionrefs; k++) { 5813 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5814 if (ref->sessionid == sessionid) { 5815 ALOGV(" session %d still exists for %d with %d refs", 5816 sessionid, ref->pid, ref->cnt); 5817 found = true; 5818 break; 5819 } 5820 } 5821 if (!found) { 5822 // remove all effects from the chain 5823 while (ec->mEffects.size()) { 5824 sp<EffectModule> effect = ec->mEffects[0]; 5825 effect->unPin(); 5826 Mutex::Autolock _l (t->mLock); 5827 t->removeEffect_l(effect); 5828 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5829 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5830 if (handle != 0) { 5831 handle->mEffect.clear(); 5832 if (handle->mHasControl && handle->mEnabled) { 5833 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5834 } 5835 } 5836 } 5837 AudioSystem::unregisterEffect(effect->id()); 5838 } 5839 } 5840 } 5841 return; 5842} 5843 5844// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5845AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5846{ 5847 PlaybackThread *thread = NULL; 5848 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5849 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5850 } 5851 return thread; 5852} 5853 5854// checkMixerThread_l() must be called with AudioFlinger::mLock held 5855AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5856{ 5857 PlaybackThread *thread = checkPlaybackThread_l(output); 5858 if (thread != NULL) { 5859 if (thread->type() == ThreadBase::DIRECT) { 5860 thread = NULL; 5861 } 5862 } 5863 return (MixerThread *)thread; 5864} 5865 5866// checkRecordThread_l() must be called with AudioFlinger::mLock held 5867AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5868{ 5869 RecordThread *thread = NULL; 5870 if (mRecordThreads.indexOfKey(input) >= 0) { 5871 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5872 } 5873 return thread; 5874} 5875 5876uint32_t AudioFlinger::nextUniqueId() 5877{ 5878 return android_atomic_inc(&mNextUniqueId); 5879} 5880 5881AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5882{ 5883 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5884 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5885 AudioStreamOut *output = thread->getOutput(); 5886 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5887 return thread; 5888 } 5889 } 5890 return NULL; 5891} 5892 5893uint32_t AudioFlinger::primaryOutputDevice_l() 5894{ 5895 PlaybackThread *thread = primaryPlaybackThread_l(); 5896 5897 if (thread == NULL) { 5898 return 0; 5899 } 5900 5901 return thread->device(); 5902} 5903 5904 5905// ---------------------------------------------------------------------------- 5906// Effect management 5907// ---------------------------------------------------------------------------- 5908 5909 5910status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5911{ 5912 Mutex::Autolock _l(mLock); 5913 return EffectQueryNumberEffects(numEffects); 5914} 5915 5916status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5917{ 5918 Mutex::Autolock _l(mLock); 5919 return EffectQueryEffect(index, descriptor); 5920} 5921 5922status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5923 effect_descriptor_t *descriptor) const 5924{ 5925 Mutex::Autolock _l(mLock); 5926 return EffectGetDescriptor(pUuid, descriptor); 5927} 5928 5929 5930sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5931 effect_descriptor_t *pDesc, 5932 const sp<IEffectClient>& effectClient, 5933 int32_t priority, 5934 audio_io_handle_t io, 5935 int sessionId, 5936 status_t *status, 5937 int *id, 5938 int *enabled) 5939{ 5940 status_t lStatus = NO_ERROR; 5941 sp<EffectHandle> handle; 5942 effect_descriptor_t desc; 5943 5944 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5945 pid, effectClient.get(), priority, sessionId, io); 5946 5947 if (pDesc == NULL) { 5948 lStatus = BAD_VALUE; 5949 goto Exit; 5950 } 5951 5952 // check audio settings permission for global effects 5953 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5954 lStatus = PERMISSION_DENIED; 5955 goto Exit; 5956 } 5957 5958 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5959 // that can only be created by audio policy manager (running in same process) 5960 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5961 lStatus = PERMISSION_DENIED; 5962 goto Exit; 5963 } 5964 5965 if (io == 0) { 5966 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5967 // output must be specified by AudioPolicyManager when using session 5968 // AUDIO_SESSION_OUTPUT_STAGE 5969 lStatus = BAD_VALUE; 5970 goto Exit; 5971 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5972 // if the output returned by getOutputForEffect() is removed before we lock the 5973 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5974 // and we will exit safely 5975 io = AudioSystem::getOutputForEffect(&desc); 5976 } 5977 } 5978 5979 { 5980 Mutex::Autolock _l(mLock); 5981 5982 5983 if (!EffectIsNullUuid(&pDesc->uuid)) { 5984 // if uuid is specified, request effect descriptor 5985 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5986 if (lStatus < 0) { 5987 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5988 goto Exit; 5989 } 5990 } else { 5991 // if uuid is not specified, look for an available implementation 5992 // of the required type in effect factory 5993 if (EffectIsNullUuid(&pDesc->type)) { 5994 ALOGW("createEffect() no effect type"); 5995 lStatus = BAD_VALUE; 5996 goto Exit; 5997 } 5998 uint32_t numEffects = 0; 5999 effect_descriptor_t d; 6000 d.flags = 0; // prevent compiler warning 6001 bool found = false; 6002 6003 lStatus = EffectQueryNumberEffects(&numEffects); 6004 if (lStatus < 0) { 6005 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6006 goto Exit; 6007 } 6008 for (uint32_t i = 0; i < numEffects; i++) { 6009 lStatus = EffectQueryEffect(i, &desc); 6010 if (lStatus < 0) { 6011 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6012 continue; 6013 } 6014 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6015 // If matching type found save effect descriptor. If the session is 6016 // 0 and the effect is not auxiliary, continue enumeration in case 6017 // an auxiliary version of this effect type is available 6018 found = true; 6019 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6020 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6021 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6022 break; 6023 } 6024 } 6025 } 6026 if (!found) { 6027 lStatus = BAD_VALUE; 6028 ALOGW("createEffect() effect not found"); 6029 goto Exit; 6030 } 6031 // For same effect type, chose auxiliary version over insert version if 6032 // connect to output mix (Compliance to OpenSL ES) 6033 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6034 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6035 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6036 } 6037 } 6038 6039 // Do not allow auxiliary effects on a session different from 0 (output mix) 6040 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6041 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6042 lStatus = INVALID_OPERATION; 6043 goto Exit; 6044 } 6045 6046 // check recording permission for visualizer 6047 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6048 !recordingAllowed()) { 6049 lStatus = PERMISSION_DENIED; 6050 goto Exit; 6051 } 6052 6053 // return effect descriptor 6054 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6055 6056 // If output is not specified try to find a matching audio session ID in one of the 6057 // output threads. 6058 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6059 // because of code checking output when entering the function. 6060 // Note: io is never 0 when creating an effect on an input 6061 if (io == 0) { 6062 // look for the thread where the specified audio session is present 6063 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6064 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6065 io = mPlaybackThreads.keyAt(i); 6066 break; 6067 } 6068 } 6069 if (io == 0) { 6070 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6071 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6072 io = mRecordThreads.keyAt(i); 6073 break; 6074 } 6075 } 6076 } 6077 // If no output thread contains the requested session ID, default to 6078 // first output. The effect chain will be moved to the correct output 6079 // thread when a track with the same session ID is created 6080 if (io == 0 && mPlaybackThreads.size()) { 6081 io = mPlaybackThreads.keyAt(0); 6082 } 6083 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6084 } 6085 ThreadBase *thread = checkRecordThread_l(io); 6086 if (thread == NULL) { 6087 thread = checkPlaybackThread_l(io); 6088 if (thread == NULL) { 6089 ALOGE("createEffect() unknown output thread"); 6090 lStatus = BAD_VALUE; 6091 goto Exit; 6092 } 6093 } 6094 6095 sp<Client> client = registerPid_l(pid); 6096 6097 // create effect on selected output thread 6098 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6099 &desc, enabled, &lStatus); 6100 if (handle != 0 && id != NULL) { 6101 *id = handle->id(); 6102 } 6103 } 6104 6105Exit: 6106 if(status) { 6107 *status = lStatus; 6108 } 6109 return handle; 6110} 6111 6112status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6113 audio_io_handle_t dstOutput) 6114{ 6115 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6116 sessionId, srcOutput, dstOutput); 6117 Mutex::Autolock _l(mLock); 6118 if (srcOutput == dstOutput) { 6119 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6120 return NO_ERROR; 6121 } 6122 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6123 if (srcThread == NULL) { 6124 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6125 return BAD_VALUE; 6126 } 6127 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6128 if (dstThread == NULL) { 6129 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6130 return BAD_VALUE; 6131 } 6132 6133 Mutex::Autolock _dl(dstThread->mLock); 6134 Mutex::Autolock _sl(srcThread->mLock); 6135 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6136 6137 return NO_ERROR; 6138} 6139 6140// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6141status_t AudioFlinger::moveEffectChain_l(int sessionId, 6142 AudioFlinger::PlaybackThread *srcThread, 6143 AudioFlinger::PlaybackThread *dstThread, 6144 bool reRegister) 6145{ 6146 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6147 sessionId, srcThread, dstThread); 6148 6149 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6150 if (chain == 0) { 6151 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6152 sessionId, srcThread); 6153 return INVALID_OPERATION; 6154 } 6155 6156 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6157 // so that a new chain is created with correct parameters when first effect is added. This is 6158 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6159 // removed. 6160 srcThread->removeEffectChain_l(chain); 6161 6162 // transfer all effects one by one so that new effect chain is created on new thread with 6163 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6164 audio_io_handle_t dstOutput = dstThread->id(); 6165 sp<EffectChain> dstChain; 6166 uint32_t strategy = 0; // prevent compiler warning 6167 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6168 while (effect != 0) { 6169 srcThread->removeEffect_l(effect); 6170 dstThread->addEffect_l(effect); 6171 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6172 if (effect->state() == EffectModule::ACTIVE || 6173 effect->state() == EffectModule::STOPPING) { 6174 effect->start(); 6175 } 6176 // if the move request is not received from audio policy manager, the effect must be 6177 // re-registered with the new strategy and output 6178 if (dstChain == 0) { 6179 dstChain = effect->chain().promote(); 6180 if (dstChain == 0) { 6181 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6182 srcThread->addEffect_l(effect); 6183 return NO_INIT; 6184 } 6185 strategy = dstChain->strategy(); 6186 } 6187 if (reRegister) { 6188 AudioSystem::unregisterEffect(effect->id()); 6189 AudioSystem::registerEffect(&effect->desc(), 6190 dstOutput, 6191 strategy, 6192 sessionId, 6193 effect->id()); 6194 } 6195 effect = chain->getEffectFromId_l(0); 6196 } 6197 6198 return NO_ERROR; 6199} 6200 6201 6202// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6203sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6204 const sp<AudioFlinger::Client>& client, 6205 const sp<IEffectClient>& effectClient, 6206 int32_t priority, 6207 int sessionId, 6208 effect_descriptor_t *desc, 6209 int *enabled, 6210 status_t *status 6211 ) 6212{ 6213 sp<EffectModule> effect; 6214 sp<EffectHandle> handle; 6215 status_t lStatus; 6216 sp<EffectChain> chain; 6217 bool chainCreated = false; 6218 bool effectCreated = false; 6219 bool effectRegistered = false; 6220 6221 lStatus = initCheck(); 6222 if (lStatus != NO_ERROR) { 6223 ALOGW("createEffect_l() Audio driver not initialized."); 6224 goto Exit; 6225 } 6226 6227 // Do not allow effects with session ID 0 on direct output or duplicating threads 6228 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6229 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6230 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6231 desc->name, sessionId); 6232 lStatus = BAD_VALUE; 6233 goto Exit; 6234 } 6235 // Only Pre processor effects are allowed on input threads and only on input threads 6236 if ((mType == RECORD && 6237 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 6238 (mType != RECORD && 6239 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6240 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6241 desc->name, desc->flags, mType); 6242 lStatus = BAD_VALUE; 6243 goto Exit; 6244 } 6245 6246 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6247 6248 { // scope for mLock 6249 Mutex::Autolock _l(mLock); 6250 6251 // check for existing effect chain with the requested audio session 6252 chain = getEffectChain_l(sessionId); 6253 if (chain == 0) { 6254 // create a new chain for this session 6255 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6256 chain = new EffectChain(this, sessionId); 6257 addEffectChain_l(chain); 6258 chain->setStrategy(getStrategyForSession_l(sessionId)); 6259 chainCreated = true; 6260 } else { 6261 effect = chain->getEffectFromDesc_l(desc); 6262 } 6263 6264 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6265 6266 if (effect == 0) { 6267 int id = mAudioFlinger->nextUniqueId(); 6268 // Check CPU and memory usage 6269 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6270 if (lStatus != NO_ERROR) { 6271 goto Exit; 6272 } 6273 effectRegistered = true; 6274 // create a new effect module if none present in the chain 6275 effect = new EffectModule(this, chain, desc, id, sessionId); 6276 lStatus = effect->status(); 6277 if (lStatus != NO_ERROR) { 6278 goto Exit; 6279 } 6280 lStatus = chain->addEffect_l(effect); 6281 if (lStatus != NO_ERROR) { 6282 goto Exit; 6283 } 6284 effectCreated = true; 6285 6286 effect->setDevice(mDevice); 6287 effect->setMode(mAudioFlinger->getMode()); 6288 } 6289 // create effect handle and connect it to effect module 6290 handle = new EffectHandle(effect, client, effectClient, priority); 6291 lStatus = effect->addHandle(handle); 6292 if (enabled != NULL) { 6293 *enabled = (int)effect->isEnabled(); 6294 } 6295 } 6296 6297Exit: 6298 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6299 Mutex::Autolock _l(mLock); 6300 if (effectCreated) { 6301 chain->removeEffect_l(effect); 6302 } 6303 if (effectRegistered) { 6304 AudioSystem::unregisterEffect(effect->id()); 6305 } 6306 if (chainCreated) { 6307 removeEffectChain_l(chain); 6308 } 6309 handle.clear(); 6310 } 6311 6312 if(status) { 6313 *status = lStatus; 6314 } 6315 return handle; 6316} 6317 6318sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6319{ 6320 sp<EffectChain> chain = getEffectChain_l(sessionId); 6321 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6322} 6323 6324// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6325// PlaybackThread::mLock held 6326status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6327{ 6328 // check for existing effect chain with the requested audio session 6329 int sessionId = effect->sessionId(); 6330 sp<EffectChain> chain = getEffectChain_l(sessionId); 6331 bool chainCreated = false; 6332 6333 if (chain == 0) { 6334 // create a new chain for this session 6335 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6336 chain = new EffectChain(this, sessionId); 6337 addEffectChain_l(chain); 6338 chain->setStrategy(getStrategyForSession_l(sessionId)); 6339 chainCreated = true; 6340 } 6341 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6342 6343 if (chain->getEffectFromId_l(effect->id()) != 0) { 6344 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6345 this, effect->desc().name, chain.get()); 6346 return BAD_VALUE; 6347 } 6348 6349 status_t status = chain->addEffect_l(effect); 6350 if (status != NO_ERROR) { 6351 if (chainCreated) { 6352 removeEffectChain_l(chain); 6353 } 6354 return status; 6355 } 6356 6357 effect->setDevice(mDevice); 6358 effect->setMode(mAudioFlinger->getMode()); 6359 return NO_ERROR; 6360} 6361 6362void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6363 6364 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6365 effect_descriptor_t desc = effect->desc(); 6366 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6367 detachAuxEffect_l(effect->id()); 6368 } 6369 6370 sp<EffectChain> chain = effect->chain().promote(); 6371 if (chain != 0) { 6372 // remove effect chain if removing last effect 6373 if (chain->removeEffect_l(effect) == 0) { 6374 removeEffectChain_l(chain); 6375 } 6376 } else { 6377 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6378 } 6379} 6380 6381void AudioFlinger::ThreadBase::lockEffectChains_l( 6382 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6383{ 6384 effectChains = mEffectChains; 6385 for (size_t i = 0; i < mEffectChains.size(); i++) { 6386 mEffectChains[i]->lock(); 6387 } 6388} 6389 6390void AudioFlinger::ThreadBase::unlockEffectChains( 6391 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6392{ 6393 for (size_t i = 0; i < effectChains.size(); i++) { 6394 effectChains[i]->unlock(); 6395 } 6396} 6397 6398sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6399{ 6400 Mutex::Autolock _l(mLock); 6401 return getEffectChain_l(sessionId); 6402} 6403 6404sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6405{ 6406 size_t size = mEffectChains.size(); 6407 for (size_t i = 0; i < size; i++) { 6408 if (mEffectChains[i]->sessionId() == sessionId) { 6409 return mEffectChains[i]; 6410 } 6411 } 6412 return 0; 6413} 6414 6415void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6416{ 6417 Mutex::Autolock _l(mLock); 6418 size_t size = mEffectChains.size(); 6419 for (size_t i = 0; i < size; i++) { 6420 mEffectChains[i]->setMode_l(mode); 6421 } 6422} 6423 6424void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6425 const wp<EffectHandle>& handle, 6426 bool unpinIfLast) { 6427 6428 Mutex::Autolock _l(mLock); 6429 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6430 // delete the effect module if removing last handle on it 6431 if (effect->removeHandle(handle) == 0) { 6432 if (!effect->isPinned() || unpinIfLast) { 6433 removeEffect_l(effect); 6434 AudioSystem::unregisterEffect(effect->id()); 6435 } 6436 } 6437} 6438 6439status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6440{ 6441 int session = chain->sessionId(); 6442 int16_t *buffer = mMixBuffer; 6443 bool ownsBuffer = false; 6444 6445 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6446 if (session > 0) { 6447 // Only one effect chain can be present in direct output thread and it uses 6448 // the mix buffer as input 6449 if (mType != DIRECT) { 6450 size_t numSamples = mFrameCount * mChannelCount; 6451 buffer = new int16_t[numSamples]; 6452 memset(buffer, 0, numSamples * sizeof(int16_t)); 6453 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6454 ownsBuffer = true; 6455 } 6456 6457 // Attach all tracks with same session ID to this chain. 6458 for (size_t i = 0; i < mTracks.size(); ++i) { 6459 sp<Track> track = mTracks[i]; 6460 if (session == track->sessionId()) { 6461 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6462 track->setMainBuffer(buffer); 6463 chain->incTrackCnt(); 6464 } 6465 } 6466 6467 // indicate all active tracks in the chain 6468 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6469 sp<Track> track = mActiveTracks[i].promote(); 6470 if (track == 0) continue; 6471 if (session == track->sessionId()) { 6472 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6473 chain->incActiveTrackCnt(); 6474 } 6475 } 6476 } 6477 6478 chain->setInBuffer(buffer, ownsBuffer); 6479 chain->setOutBuffer(mMixBuffer); 6480 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6481 // chains list in order to be processed last as it contains output stage effects 6482 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6483 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6484 // after track specific effects and before output stage 6485 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6486 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6487 // Effect chain for other sessions are inserted at beginning of effect 6488 // chains list to be processed before output mix effects. Relative order between other 6489 // sessions is not important 6490 size_t size = mEffectChains.size(); 6491 size_t i = 0; 6492 for (i = 0; i < size; i++) { 6493 if (mEffectChains[i]->sessionId() < session) break; 6494 } 6495 mEffectChains.insertAt(chain, i); 6496 checkSuspendOnAddEffectChain_l(chain); 6497 6498 return NO_ERROR; 6499} 6500 6501size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6502{ 6503 int session = chain->sessionId(); 6504 6505 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6506 6507 for (size_t i = 0; i < mEffectChains.size(); i++) { 6508 if (chain == mEffectChains[i]) { 6509 mEffectChains.removeAt(i); 6510 // detach all active tracks from the chain 6511 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6512 sp<Track> track = mActiveTracks[i].promote(); 6513 if (track == 0) continue; 6514 if (session == track->sessionId()) { 6515 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6516 chain.get(), session); 6517 chain->decActiveTrackCnt(); 6518 } 6519 } 6520 6521 // detach all tracks with same session ID from this chain 6522 for (size_t i = 0; i < mTracks.size(); ++i) { 6523 sp<Track> track = mTracks[i]; 6524 if (session == track->sessionId()) { 6525 track->setMainBuffer(mMixBuffer); 6526 chain->decTrackCnt(); 6527 } 6528 } 6529 break; 6530 } 6531 } 6532 return mEffectChains.size(); 6533} 6534 6535status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6536 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6537{ 6538 Mutex::Autolock _l(mLock); 6539 return attachAuxEffect_l(track, EffectId); 6540} 6541 6542status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6543 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6544{ 6545 status_t status = NO_ERROR; 6546 6547 if (EffectId == 0) { 6548 track->setAuxBuffer(0, NULL); 6549 } else { 6550 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6551 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6552 if (effect != 0) { 6553 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6554 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6555 } else { 6556 status = INVALID_OPERATION; 6557 } 6558 } else { 6559 status = BAD_VALUE; 6560 } 6561 } 6562 return status; 6563} 6564 6565void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6566{ 6567 for (size_t i = 0; i < mTracks.size(); ++i) { 6568 sp<Track> track = mTracks[i]; 6569 if (track->auxEffectId() == effectId) { 6570 attachAuxEffect_l(track, 0); 6571 } 6572 } 6573} 6574 6575status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6576{ 6577 // only one chain per input thread 6578 if (mEffectChains.size() != 0) { 6579 return INVALID_OPERATION; 6580 } 6581 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6582 6583 chain->setInBuffer(NULL); 6584 chain->setOutBuffer(NULL); 6585 6586 checkSuspendOnAddEffectChain_l(chain); 6587 6588 mEffectChains.add(chain); 6589 6590 return NO_ERROR; 6591} 6592 6593size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6594{ 6595 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6596 ALOGW_IF(mEffectChains.size() != 1, 6597 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6598 chain.get(), mEffectChains.size(), this); 6599 if (mEffectChains.size() == 1) { 6600 mEffectChains.removeAt(0); 6601 } 6602 return 0; 6603} 6604 6605// ---------------------------------------------------------------------------- 6606// EffectModule implementation 6607// ---------------------------------------------------------------------------- 6608 6609#undef LOG_TAG 6610#define LOG_TAG "AudioFlinger::EffectModule" 6611 6612AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6613 const wp<AudioFlinger::EffectChain>& chain, 6614 effect_descriptor_t *desc, 6615 int id, 6616 int sessionId) 6617 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6618 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6619{ 6620 ALOGV("Constructor %p", this); 6621 int lStatus; 6622 sp<ThreadBase> thread = mThread.promote(); 6623 if (thread == 0) { 6624 return; 6625 } 6626 6627 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6628 6629 // create effect engine from effect factory 6630 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6631 6632 if (mStatus != NO_ERROR) { 6633 return; 6634 } 6635 lStatus = init(); 6636 if (lStatus < 0) { 6637 mStatus = lStatus; 6638 goto Error; 6639 } 6640 6641 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6642 mPinned = true; 6643 } 6644 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6645 return; 6646Error: 6647 EffectRelease(mEffectInterface); 6648 mEffectInterface = NULL; 6649 ALOGV("Constructor Error %d", mStatus); 6650} 6651 6652AudioFlinger::EffectModule::~EffectModule() 6653{ 6654 ALOGV("Destructor %p", this); 6655 if (mEffectInterface != NULL) { 6656 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6657 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6658 sp<ThreadBase> thread = mThread.promote(); 6659 if (thread != 0) { 6660 audio_stream_t *stream = thread->stream(); 6661 if (stream != NULL) { 6662 stream->remove_audio_effect(stream, mEffectInterface); 6663 } 6664 } 6665 } 6666 // release effect engine 6667 EffectRelease(mEffectInterface); 6668 } 6669} 6670 6671status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6672{ 6673 status_t status; 6674 6675 Mutex::Autolock _l(mLock); 6676 int priority = handle->priority(); 6677 size_t size = mHandles.size(); 6678 sp<EffectHandle> h; 6679 size_t i; 6680 for (i = 0; i < size; i++) { 6681 h = mHandles[i].promote(); 6682 if (h == 0) continue; 6683 if (h->priority() <= priority) break; 6684 } 6685 // if inserted in first place, move effect control from previous owner to this handle 6686 if (i == 0) { 6687 bool enabled = false; 6688 if (h != 0) { 6689 enabled = h->enabled(); 6690 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6691 } 6692 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6693 status = NO_ERROR; 6694 } else { 6695 status = ALREADY_EXISTS; 6696 } 6697 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6698 mHandles.insertAt(handle, i); 6699 return status; 6700} 6701 6702size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6703{ 6704 Mutex::Autolock _l(mLock); 6705 size_t size = mHandles.size(); 6706 size_t i; 6707 for (i = 0; i < size; i++) { 6708 if (mHandles[i] == handle) break; 6709 } 6710 if (i == size) { 6711 return size; 6712 } 6713 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6714 6715 bool enabled = false; 6716 EffectHandle *hdl = handle.unsafe_get(); 6717 if (hdl != NULL) { 6718 ALOGV("removeHandle() unsafe_get OK"); 6719 enabled = hdl->enabled(); 6720 } 6721 mHandles.removeAt(i); 6722 size = mHandles.size(); 6723 // if removed from first place, move effect control from this handle to next in line 6724 if (i == 0 && size != 0) { 6725 sp<EffectHandle> h = mHandles[0].promote(); 6726 if (h != 0) { 6727 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6728 } 6729 } 6730 6731 // Prevent calls to process() and other functions on effect interface from now on. 6732 // The effect engine will be released by the destructor when the last strong reference on 6733 // this object is released which can happen after next process is called. 6734 if (size == 0 && !mPinned) { 6735 mState = DESTROYED; 6736 } 6737 6738 return size; 6739} 6740 6741sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6742{ 6743 Mutex::Autolock _l(mLock); 6744 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6745} 6746 6747void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6748{ 6749 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6750 // keep a strong reference on this EffectModule to avoid calling the 6751 // destructor before we exit 6752 sp<EffectModule> keep(this); 6753 { 6754 sp<ThreadBase> thread = mThread.promote(); 6755 if (thread != 0) { 6756 thread->disconnectEffect(keep, handle, unpinIfLast); 6757 } 6758 } 6759} 6760 6761void AudioFlinger::EffectModule::updateState() { 6762 Mutex::Autolock _l(mLock); 6763 6764 switch (mState) { 6765 case RESTART: 6766 reset_l(); 6767 // FALL THROUGH 6768 6769 case STARTING: 6770 // clear auxiliary effect input buffer for next accumulation 6771 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6772 memset(mConfig.inputCfg.buffer.raw, 6773 0, 6774 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6775 } 6776 start_l(); 6777 mState = ACTIVE; 6778 break; 6779 case STOPPING: 6780 stop_l(); 6781 mDisableWaitCnt = mMaxDisableWaitCnt; 6782 mState = STOPPED; 6783 break; 6784 case STOPPED: 6785 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6786 // turn off sequence. 6787 if (--mDisableWaitCnt == 0) { 6788 reset_l(); 6789 mState = IDLE; 6790 } 6791 break; 6792 default: //IDLE , ACTIVE, DESTROYED 6793 break; 6794 } 6795} 6796 6797void AudioFlinger::EffectModule::process() 6798{ 6799 Mutex::Autolock _l(mLock); 6800 6801 if (mState == DESTROYED || mEffectInterface == NULL || 6802 mConfig.inputCfg.buffer.raw == NULL || 6803 mConfig.outputCfg.buffer.raw == NULL) { 6804 return; 6805 } 6806 6807 if (isProcessEnabled()) { 6808 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6809 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6810 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6811 mConfig.inputCfg.buffer.s32, 6812 mConfig.inputCfg.buffer.frameCount/2); 6813 } 6814 6815 // do the actual processing in the effect engine 6816 int ret = (*mEffectInterface)->process(mEffectInterface, 6817 &mConfig.inputCfg.buffer, 6818 &mConfig.outputCfg.buffer); 6819 6820 // force transition to IDLE state when engine is ready 6821 if (mState == STOPPED && ret == -ENODATA) { 6822 mDisableWaitCnt = 1; 6823 } 6824 6825 // clear auxiliary effect input buffer for next accumulation 6826 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6827 memset(mConfig.inputCfg.buffer.raw, 0, 6828 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6829 } 6830 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6831 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6832 // If an insert effect is idle and input buffer is different from output buffer, 6833 // accumulate input onto output 6834 sp<EffectChain> chain = mChain.promote(); 6835 if (chain != 0 && chain->activeTrackCnt() != 0) { 6836 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6837 int16_t *in = mConfig.inputCfg.buffer.s16; 6838 int16_t *out = mConfig.outputCfg.buffer.s16; 6839 for (size_t i = 0; i < frameCnt; i++) { 6840 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6841 } 6842 } 6843 } 6844} 6845 6846void AudioFlinger::EffectModule::reset_l() 6847{ 6848 if (mEffectInterface == NULL) { 6849 return; 6850 } 6851 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6852} 6853 6854status_t AudioFlinger::EffectModule::configure() 6855{ 6856 uint32_t channels; 6857 if (mEffectInterface == NULL) { 6858 return NO_INIT; 6859 } 6860 6861 sp<ThreadBase> thread = mThread.promote(); 6862 if (thread == 0) { 6863 return DEAD_OBJECT; 6864 } 6865 6866 // TODO: handle configuration of effects replacing track process 6867 if (thread->channelCount() == 1) { 6868 channels = AUDIO_CHANNEL_OUT_MONO; 6869 } else { 6870 channels = AUDIO_CHANNEL_OUT_STEREO; 6871 } 6872 6873 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6874 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6875 } else { 6876 mConfig.inputCfg.channels = channels; 6877 } 6878 mConfig.outputCfg.channels = channels; 6879 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6880 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6881 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6882 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6883 mConfig.inputCfg.bufferProvider.cookie = NULL; 6884 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6885 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6886 mConfig.outputCfg.bufferProvider.cookie = NULL; 6887 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6888 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6889 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6890 // Insert effect: 6891 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6892 // always overwrites output buffer: input buffer == output buffer 6893 // - in other sessions: 6894 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6895 // other effect: overwrites output buffer: input buffer == output buffer 6896 // Auxiliary effect: 6897 // accumulates in output buffer: input buffer != output buffer 6898 // Therefore: accumulate <=> input buffer != output buffer 6899 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6900 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6901 } else { 6902 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6903 } 6904 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6905 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6906 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6907 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6908 6909 ALOGV("configure() %p thread %p buffer %p framecount %d", 6910 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6911 6912 status_t cmdStatus; 6913 uint32_t size = sizeof(int); 6914 status_t status = (*mEffectInterface)->command(mEffectInterface, 6915 EFFECT_CMD_SET_CONFIG, 6916 sizeof(effect_config_t), 6917 &mConfig, 6918 &size, 6919 &cmdStatus); 6920 if (status == 0) { 6921 status = cmdStatus; 6922 } 6923 6924 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6925 (1000 * mConfig.outputCfg.buffer.frameCount); 6926 6927 return status; 6928} 6929 6930status_t AudioFlinger::EffectModule::init() 6931{ 6932 Mutex::Autolock _l(mLock); 6933 if (mEffectInterface == NULL) { 6934 return NO_INIT; 6935 } 6936 status_t cmdStatus; 6937 uint32_t size = sizeof(status_t); 6938 status_t status = (*mEffectInterface)->command(mEffectInterface, 6939 EFFECT_CMD_INIT, 6940 0, 6941 NULL, 6942 &size, 6943 &cmdStatus); 6944 if (status == 0) { 6945 status = cmdStatus; 6946 } 6947 return status; 6948} 6949 6950status_t AudioFlinger::EffectModule::start() 6951{ 6952 Mutex::Autolock _l(mLock); 6953 return start_l(); 6954} 6955 6956status_t AudioFlinger::EffectModule::start_l() 6957{ 6958 if (mEffectInterface == NULL) { 6959 return NO_INIT; 6960 } 6961 status_t cmdStatus; 6962 uint32_t size = sizeof(status_t); 6963 status_t status = (*mEffectInterface)->command(mEffectInterface, 6964 EFFECT_CMD_ENABLE, 6965 0, 6966 NULL, 6967 &size, 6968 &cmdStatus); 6969 if (status == 0) { 6970 status = cmdStatus; 6971 } 6972 if (status == 0 && 6973 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6974 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6975 sp<ThreadBase> thread = mThread.promote(); 6976 if (thread != 0) { 6977 audio_stream_t *stream = thread->stream(); 6978 if (stream != NULL) { 6979 stream->add_audio_effect(stream, mEffectInterface); 6980 } 6981 } 6982 } 6983 return status; 6984} 6985 6986status_t AudioFlinger::EffectModule::stop() 6987{ 6988 Mutex::Autolock _l(mLock); 6989 return stop_l(); 6990} 6991 6992status_t AudioFlinger::EffectModule::stop_l() 6993{ 6994 if (mEffectInterface == NULL) { 6995 return NO_INIT; 6996 } 6997 status_t cmdStatus; 6998 uint32_t size = sizeof(status_t); 6999 status_t status = (*mEffectInterface)->command(mEffectInterface, 7000 EFFECT_CMD_DISABLE, 7001 0, 7002 NULL, 7003 &size, 7004 &cmdStatus); 7005 if (status == 0) { 7006 status = cmdStatus; 7007 } 7008 if (status == 0 && 7009 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7010 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7011 sp<ThreadBase> thread = mThread.promote(); 7012 if (thread != 0) { 7013 audio_stream_t *stream = thread->stream(); 7014 if (stream != NULL) { 7015 stream->remove_audio_effect(stream, mEffectInterface); 7016 } 7017 } 7018 } 7019 return status; 7020} 7021 7022status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7023 uint32_t cmdSize, 7024 void *pCmdData, 7025 uint32_t *replySize, 7026 void *pReplyData) 7027{ 7028 Mutex::Autolock _l(mLock); 7029// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7030 7031 if (mState == DESTROYED || mEffectInterface == NULL) { 7032 return NO_INIT; 7033 } 7034 status_t status = (*mEffectInterface)->command(mEffectInterface, 7035 cmdCode, 7036 cmdSize, 7037 pCmdData, 7038 replySize, 7039 pReplyData); 7040 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7041 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7042 for (size_t i = 1; i < mHandles.size(); i++) { 7043 sp<EffectHandle> h = mHandles[i].promote(); 7044 if (h != 0) { 7045 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7046 } 7047 } 7048 } 7049 return status; 7050} 7051 7052status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7053{ 7054 7055 Mutex::Autolock _l(mLock); 7056 ALOGV("setEnabled %p enabled %d", this, enabled); 7057 7058 if (enabled != isEnabled()) { 7059 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7060 if (enabled && status != NO_ERROR) { 7061 return status; 7062 } 7063 7064 switch (mState) { 7065 // going from disabled to enabled 7066 case IDLE: 7067 mState = STARTING; 7068 break; 7069 case STOPPED: 7070 mState = RESTART; 7071 break; 7072 case STOPPING: 7073 mState = ACTIVE; 7074 break; 7075 7076 // going from enabled to disabled 7077 case RESTART: 7078 mState = STOPPED; 7079 break; 7080 case STARTING: 7081 mState = IDLE; 7082 break; 7083 case ACTIVE: 7084 mState = STOPPING; 7085 break; 7086 case DESTROYED: 7087 return NO_ERROR; // simply ignore as we are being destroyed 7088 } 7089 for (size_t i = 1; i < mHandles.size(); i++) { 7090 sp<EffectHandle> h = mHandles[i].promote(); 7091 if (h != 0) { 7092 h->setEnabled(enabled); 7093 } 7094 } 7095 } 7096 return NO_ERROR; 7097} 7098 7099bool AudioFlinger::EffectModule::isEnabled() const 7100{ 7101 switch (mState) { 7102 case RESTART: 7103 case STARTING: 7104 case ACTIVE: 7105 return true; 7106 case IDLE: 7107 case STOPPING: 7108 case STOPPED: 7109 case DESTROYED: 7110 default: 7111 return false; 7112 } 7113} 7114 7115bool AudioFlinger::EffectModule::isProcessEnabled() const 7116{ 7117 switch (mState) { 7118 case RESTART: 7119 case ACTIVE: 7120 case STOPPING: 7121 case STOPPED: 7122 return true; 7123 case IDLE: 7124 case STARTING: 7125 case DESTROYED: 7126 default: 7127 return false; 7128 } 7129} 7130 7131status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7132{ 7133 Mutex::Autolock _l(mLock); 7134 status_t status = NO_ERROR; 7135 7136 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7137 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7138 if (isProcessEnabled() && 7139 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7140 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7141 status_t cmdStatus; 7142 uint32_t volume[2]; 7143 uint32_t *pVolume = NULL; 7144 uint32_t size = sizeof(volume); 7145 volume[0] = *left; 7146 volume[1] = *right; 7147 if (controller) { 7148 pVolume = volume; 7149 } 7150 status = (*mEffectInterface)->command(mEffectInterface, 7151 EFFECT_CMD_SET_VOLUME, 7152 size, 7153 volume, 7154 &size, 7155 pVolume); 7156 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7157 *left = volume[0]; 7158 *right = volume[1]; 7159 } 7160 } 7161 return status; 7162} 7163 7164status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7165{ 7166 Mutex::Autolock _l(mLock); 7167 status_t status = NO_ERROR; 7168 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7169 // audio pre processing modules on RecordThread can receive both output and 7170 // input device indication in the same call 7171 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7172 if (dev) { 7173 status_t cmdStatus; 7174 uint32_t size = sizeof(status_t); 7175 7176 status = (*mEffectInterface)->command(mEffectInterface, 7177 EFFECT_CMD_SET_DEVICE, 7178 sizeof(uint32_t), 7179 &dev, 7180 &size, 7181 &cmdStatus); 7182 if (status == NO_ERROR) { 7183 status = cmdStatus; 7184 } 7185 } 7186 dev = device & AUDIO_DEVICE_IN_ALL; 7187 if (dev) { 7188 status_t cmdStatus; 7189 uint32_t size = sizeof(status_t); 7190 7191 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7192 EFFECT_CMD_SET_INPUT_DEVICE, 7193 sizeof(uint32_t), 7194 &dev, 7195 &size, 7196 &cmdStatus); 7197 if (status2 == NO_ERROR) { 7198 status2 = cmdStatus; 7199 } 7200 if (status == NO_ERROR) { 7201 status = status2; 7202 } 7203 } 7204 } 7205 return status; 7206} 7207 7208status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7209{ 7210 Mutex::Autolock _l(mLock); 7211 status_t status = NO_ERROR; 7212 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7213 status_t cmdStatus; 7214 uint32_t size = sizeof(status_t); 7215 status = (*mEffectInterface)->command(mEffectInterface, 7216 EFFECT_CMD_SET_AUDIO_MODE, 7217 sizeof(audio_mode_t), 7218 &mode, 7219 &size, 7220 &cmdStatus); 7221 if (status == NO_ERROR) { 7222 status = cmdStatus; 7223 } 7224 } 7225 return status; 7226} 7227 7228void AudioFlinger::EffectModule::setSuspended(bool suspended) 7229{ 7230 Mutex::Autolock _l(mLock); 7231 mSuspended = suspended; 7232} 7233 7234bool AudioFlinger::EffectModule::suspended() const 7235{ 7236 Mutex::Autolock _l(mLock); 7237 return mSuspended; 7238} 7239 7240status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7241{ 7242 const size_t SIZE = 256; 7243 char buffer[SIZE]; 7244 String8 result; 7245 7246 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7247 result.append(buffer); 7248 7249 bool locked = tryLock(mLock); 7250 // failed to lock - AudioFlinger is probably deadlocked 7251 if (!locked) { 7252 result.append("\t\tCould not lock Fx mutex:\n"); 7253 } 7254 7255 result.append("\t\tSession Status State Engine:\n"); 7256 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7257 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7258 result.append(buffer); 7259 7260 result.append("\t\tDescriptor:\n"); 7261 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7262 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7263 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7264 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7265 result.append(buffer); 7266 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7267 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7268 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7269 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7270 result.append(buffer); 7271 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7272 mDescriptor.apiVersion, 7273 mDescriptor.flags); 7274 result.append(buffer); 7275 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7276 mDescriptor.name); 7277 result.append(buffer); 7278 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7279 mDescriptor.implementor); 7280 result.append(buffer); 7281 7282 result.append("\t\t- Input 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.inputCfg.buffer.raw, 7286 mConfig.inputCfg.buffer.frameCount, 7287 mConfig.inputCfg.samplingRate, 7288 mConfig.inputCfg.channels, 7289 mConfig.inputCfg.format); 7290 result.append(buffer); 7291 7292 result.append("\t\t- Output configuration:\n"); 7293 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7294 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7295 (uint32_t)mConfig.outputCfg.buffer.raw, 7296 mConfig.outputCfg.buffer.frameCount, 7297 mConfig.outputCfg.samplingRate, 7298 mConfig.outputCfg.channels, 7299 mConfig.outputCfg.format); 7300 result.append(buffer); 7301 7302 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7303 result.append(buffer); 7304 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7305 for (size_t i = 0; i < mHandles.size(); ++i) { 7306 sp<EffectHandle> handle = mHandles[i].promote(); 7307 if (handle != 0) { 7308 handle->dump(buffer, SIZE); 7309 result.append(buffer); 7310 } 7311 } 7312 7313 result.append("\n"); 7314 7315 write(fd, result.string(), result.length()); 7316 7317 if (locked) { 7318 mLock.unlock(); 7319 } 7320 7321 return NO_ERROR; 7322} 7323 7324// ---------------------------------------------------------------------------- 7325// EffectHandle implementation 7326// ---------------------------------------------------------------------------- 7327 7328#undef LOG_TAG 7329#define LOG_TAG "AudioFlinger::EffectHandle" 7330 7331AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7332 const sp<AudioFlinger::Client>& client, 7333 const sp<IEffectClient>& effectClient, 7334 int32_t priority) 7335 : BnEffect(), 7336 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7337 mPriority(priority), mHasControl(false), mEnabled(false) 7338{ 7339 ALOGV("constructor %p", this); 7340 7341 if (client == 0) { 7342 return; 7343 } 7344 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7345 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7346 if (mCblkMemory != 0) { 7347 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7348 7349 if (mCblk != NULL) { 7350 new(mCblk) effect_param_cblk_t(); 7351 mBuffer = (uint8_t *)mCblk + bufOffset; 7352 } 7353 } else { 7354 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7355 return; 7356 } 7357} 7358 7359AudioFlinger::EffectHandle::~EffectHandle() 7360{ 7361 ALOGV("Destructor %p", this); 7362 disconnect(false); 7363 ALOGV("Destructor DONE %p", this); 7364} 7365 7366status_t AudioFlinger::EffectHandle::enable() 7367{ 7368 ALOGV("enable %p", this); 7369 if (!mHasControl) return INVALID_OPERATION; 7370 if (mEffect == 0) return DEAD_OBJECT; 7371 7372 if (mEnabled) { 7373 return NO_ERROR; 7374 } 7375 7376 mEnabled = true; 7377 7378 sp<ThreadBase> thread = mEffect->thread().promote(); 7379 if (thread != 0) { 7380 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7381 } 7382 7383 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7384 if (mEffect->suspended()) { 7385 return NO_ERROR; 7386 } 7387 7388 status_t status = mEffect->setEnabled(true); 7389 if (status != NO_ERROR) { 7390 if (thread != 0) { 7391 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7392 } 7393 mEnabled = false; 7394 } 7395 return status; 7396} 7397 7398status_t AudioFlinger::EffectHandle::disable() 7399{ 7400 ALOGV("disable %p", this); 7401 if (!mHasControl) return INVALID_OPERATION; 7402 if (mEffect == 0) return DEAD_OBJECT; 7403 7404 if (!mEnabled) { 7405 return NO_ERROR; 7406 } 7407 mEnabled = false; 7408 7409 if (mEffect->suspended()) { 7410 return NO_ERROR; 7411 } 7412 7413 status_t status = mEffect->setEnabled(false); 7414 7415 sp<ThreadBase> thread = mEffect->thread().promote(); 7416 if (thread != 0) { 7417 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7418 } 7419 7420 return status; 7421} 7422 7423void AudioFlinger::EffectHandle::disconnect() 7424{ 7425 disconnect(true); 7426} 7427 7428void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7429{ 7430 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7431 if (mEffect == 0) { 7432 return; 7433 } 7434 mEffect->disconnect(this, unpinIfLast); 7435 7436 if (mHasControl && mEnabled) { 7437 sp<ThreadBase> thread = mEffect->thread().promote(); 7438 if (thread != 0) { 7439 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7440 } 7441 } 7442 7443 // release sp on module => module destructor can be called now 7444 mEffect.clear(); 7445 if (mClient != 0) { 7446 if (mCblk != NULL) { 7447 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7448 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7449 } 7450 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7451 // Client destructor must run with AudioFlinger mutex locked 7452 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7453 mClient.clear(); 7454 } 7455} 7456 7457status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7458 uint32_t cmdSize, 7459 void *pCmdData, 7460 uint32_t *replySize, 7461 void *pReplyData) 7462{ 7463// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7464// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7465 7466 // only get parameter command is permitted for applications not controlling the effect 7467 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7468 return INVALID_OPERATION; 7469 } 7470 if (mEffect == 0) return DEAD_OBJECT; 7471 if (mClient == 0) return INVALID_OPERATION; 7472 7473 // handle commands that are not forwarded transparently to effect engine 7474 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7475 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7476 // no risk to block the whole media server process or mixer threads is we are stuck here 7477 Mutex::Autolock _l(mCblk->lock); 7478 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7479 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7480 mCblk->serverIndex = 0; 7481 mCblk->clientIndex = 0; 7482 return BAD_VALUE; 7483 } 7484 status_t status = NO_ERROR; 7485 while (mCblk->serverIndex < mCblk->clientIndex) { 7486 int reply; 7487 uint32_t rsize = sizeof(int); 7488 int *p = (int *)(mBuffer + mCblk->serverIndex); 7489 int size = *p++; 7490 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7491 ALOGW("command(): invalid parameter block size"); 7492 break; 7493 } 7494 effect_param_t *param = (effect_param_t *)p; 7495 if (param->psize == 0 || param->vsize == 0) { 7496 ALOGW("command(): null parameter or value size"); 7497 mCblk->serverIndex += size; 7498 continue; 7499 } 7500 uint32_t psize = sizeof(effect_param_t) + 7501 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7502 param->vsize; 7503 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7504 psize, 7505 p, 7506 &rsize, 7507 &reply); 7508 // stop at first error encountered 7509 if (ret != NO_ERROR) { 7510 status = ret; 7511 *(int *)pReplyData = reply; 7512 break; 7513 } else if (reply != NO_ERROR) { 7514 *(int *)pReplyData = reply; 7515 break; 7516 } 7517 mCblk->serverIndex += size; 7518 } 7519 mCblk->serverIndex = 0; 7520 mCblk->clientIndex = 0; 7521 return status; 7522 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7523 *(int *)pReplyData = NO_ERROR; 7524 return enable(); 7525 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7526 *(int *)pReplyData = NO_ERROR; 7527 return disable(); 7528 } 7529 7530 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7531} 7532 7533void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7534{ 7535 ALOGV("setControl %p control %d", this, hasControl); 7536 7537 mHasControl = hasControl; 7538 mEnabled = enabled; 7539 7540 if (signal && mEffectClient != 0) { 7541 mEffectClient->controlStatusChanged(hasControl); 7542 } 7543} 7544 7545void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7546 uint32_t cmdSize, 7547 void *pCmdData, 7548 uint32_t replySize, 7549 void *pReplyData) 7550{ 7551 if (mEffectClient != 0) { 7552 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7553 } 7554} 7555 7556 7557 7558void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7559{ 7560 if (mEffectClient != 0) { 7561 mEffectClient->enableStatusChanged(enabled); 7562 } 7563} 7564 7565status_t AudioFlinger::EffectHandle::onTransact( 7566 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7567{ 7568 return BnEffect::onTransact(code, data, reply, flags); 7569} 7570 7571 7572void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7573{ 7574 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7575 7576 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7577 (mClient == 0) ? getpid_cached : mClient->pid(), 7578 mPriority, 7579 mHasControl, 7580 !locked, 7581 mCblk ? mCblk->clientIndex : 0, 7582 mCblk ? mCblk->serverIndex : 0 7583 ); 7584 7585 if (locked) { 7586 mCblk->lock.unlock(); 7587 } 7588} 7589 7590#undef LOG_TAG 7591#define LOG_TAG "AudioFlinger::EffectChain" 7592 7593AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7594 int sessionId) 7595 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7596 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7597 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7598{ 7599 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7600 sp<ThreadBase> thread = mThread.promote(); 7601 if (thread == 0) { 7602 return; 7603 } 7604 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7605 thread->frameCount(); 7606} 7607 7608AudioFlinger::EffectChain::~EffectChain() 7609{ 7610 if (mOwnInBuffer) { 7611 delete mInBuffer; 7612 } 7613 7614} 7615 7616// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7617sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7618{ 7619 size_t size = mEffects.size(); 7620 7621 for (size_t i = 0; i < size; i++) { 7622 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7623 return mEffects[i]; 7624 } 7625 } 7626 return 0; 7627} 7628 7629// getEffectFromId_l() must be called with ThreadBase::mLock held 7630sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7631{ 7632 size_t size = mEffects.size(); 7633 7634 for (size_t i = 0; i < size; i++) { 7635 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7636 if (id == 0 || mEffects[i]->id() == id) { 7637 return mEffects[i]; 7638 } 7639 } 7640 return 0; 7641} 7642 7643// getEffectFromType_l() must be called with ThreadBase::mLock held 7644sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7645 const effect_uuid_t *type) 7646{ 7647 size_t size = mEffects.size(); 7648 7649 for (size_t i = 0; i < size; i++) { 7650 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7651 return mEffects[i]; 7652 } 7653 } 7654 return 0; 7655} 7656 7657// Must be called with EffectChain::mLock locked 7658void AudioFlinger::EffectChain::process_l() 7659{ 7660 sp<ThreadBase> thread = mThread.promote(); 7661 if (thread == 0) { 7662 ALOGW("process_l(): cannot promote mixer thread"); 7663 return; 7664 } 7665 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7666 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7667 // always process effects unless no more tracks are on the session and the effect tail 7668 // has been rendered 7669 bool doProcess = true; 7670 if (!isGlobalSession) { 7671 bool tracksOnSession = (trackCnt() != 0); 7672 7673 if (!tracksOnSession && mTailBufferCount == 0) { 7674 doProcess = false; 7675 } 7676 7677 if (activeTrackCnt() == 0) { 7678 // if no track is active and the effect tail has not been rendered, 7679 // the input buffer must be cleared here as the mixer process will not do it 7680 if (tracksOnSession || mTailBufferCount > 0) { 7681 size_t numSamples = thread->frameCount() * thread->channelCount(); 7682 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7683 if (mTailBufferCount > 0) { 7684 mTailBufferCount--; 7685 } 7686 } 7687 } 7688 } 7689 7690 size_t size = mEffects.size(); 7691 if (doProcess) { 7692 for (size_t i = 0; i < size; i++) { 7693 mEffects[i]->process(); 7694 } 7695 } 7696 for (size_t i = 0; i < size; i++) { 7697 mEffects[i]->updateState(); 7698 } 7699} 7700 7701// addEffect_l() must be called with PlaybackThread::mLock held 7702status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7703{ 7704 effect_descriptor_t desc = effect->desc(); 7705 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7706 7707 Mutex::Autolock _l(mLock); 7708 effect->setChain(this); 7709 sp<ThreadBase> thread = mThread.promote(); 7710 if (thread == 0) { 7711 return NO_INIT; 7712 } 7713 effect->setThread(thread); 7714 7715 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7716 // Auxiliary effects are inserted at the beginning of mEffects vector as 7717 // they are processed first and accumulated in chain input buffer 7718 mEffects.insertAt(effect, 0); 7719 7720 // the input buffer for auxiliary effect contains mono samples in 7721 // 32 bit format. This is to avoid saturation in AudoMixer 7722 // accumulation stage. Saturation is done in EffectModule::process() before 7723 // calling the process in effect engine 7724 size_t numSamples = thread->frameCount(); 7725 int32_t *buffer = new int32_t[numSamples]; 7726 memset(buffer, 0, numSamples * sizeof(int32_t)); 7727 effect->setInBuffer((int16_t *)buffer); 7728 // auxiliary effects output samples to chain input buffer for further processing 7729 // by insert effects 7730 effect->setOutBuffer(mInBuffer); 7731 } else { 7732 // Insert effects are inserted at the end of mEffects vector as they are processed 7733 // after track and auxiliary effects. 7734 // Insert effect order as a function of indicated preference: 7735 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7736 // another effect is present 7737 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7738 // last effect claiming first position 7739 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7740 // first effect claiming last position 7741 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7742 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7743 // already present 7744 7745 size_t size = mEffects.size(); 7746 size_t idx_insert = size; 7747 ssize_t idx_insert_first = -1; 7748 ssize_t idx_insert_last = -1; 7749 7750 for (size_t i = 0; i < size; i++) { 7751 effect_descriptor_t d = mEffects[i]->desc(); 7752 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7753 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7754 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7755 // check invalid effect chaining combinations 7756 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7757 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7758 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7759 return INVALID_OPERATION; 7760 } 7761 // remember position of first insert effect and by default 7762 // select this as insert position for new effect 7763 if (idx_insert == size) { 7764 idx_insert = i; 7765 } 7766 // remember position of last insert effect claiming 7767 // first position 7768 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7769 idx_insert_first = i; 7770 } 7771 // remember position of first insert effect claiming 7772 // last position 7773 if (iPref == EFFECT_FLAG_INSERT_LAST && 7774 idx_insert_last == -1) { 7775 idx_insert_last = i; 7776 } 7777 } 7778 } 7779 7780 // modify idx_insert from first position if needed 7781 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7782 if (idx_insert_last != -1) { 7783 idx_insert = idx_insert_last; 7784 } else { 7785 idx_insert = size; 7786 } 7787 } else { 7788 if (idx_insert_first != -1) { 7789 idx_insert = idx_insert_first + 1; 7790 } 7791 } 7792 7793 // always read samples from chain input buffer 7794 effect->setInBuffer(mInBuffer); 7795 7796 // if last effect in the chain, output samples to chain 7797 // output buffer, otherwise to chain input buffer 7798 if (idx_insert == size) { 7799 if (idx_insert != 0) { 7800 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7801 mEffects[idx_insert-1]->configure(); 7802 } 7803 effect->setOutBuffer(mOutBuffer); 7804 } else { 7805 effect->setOutBuffer(mInBuffer); 7806 } 7807 mEffects.insertAt(effect, idx_insert); 7808 7809 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7810 } 7811 effect->configure(); 7812 return NO_ERROR; 7813} 7814 7815// removeEffect_l() must be called with PlaybackThread::mLock held 7816size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7817{ 7818 Mutex::Autolock _l(mLock); 7819 size_t size = mEffects.size(); 7820 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7821 7822 for (size_t i = 0; i < size; i++) { 7823 if (effect == mEffects[i]) { 7824 // calling stop here will remove pre-processing effect from the audio HAL. 7825 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7826 // the middle of a read from audio HAL 7827 if (mEffects[i]->state() == EffectModule::ACTIVE || 7828 mEffects[i]->state() == EffectModule::STOPPING) { 7829 mEffects[i]->stop(); 7830 } 7831 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7832 delete[] effect->inBuffer(); 7833 } else { 7834 if (i == size - 1 && i != 0) { 7835 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7836 mEffects[i - 1]->configure(); 7837 } 7838 } 7839 mEffects.removeAt(i); 7840 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7841 break; 7842 } 7843 } 7844 7845 return mEffects.size(); 7846} 7847 7848// setDevice_l() must be called with PlaybackThread::mLock held 7849void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7850{ 7851 size_t size = mEffects.size(); 7852 for (size_t i = 0; i < size; i++) { 7853 mEffects[i]->setDevice(device); 7854 } 7855} 7856 7857// setMode_l() must be called with PlaybackThread::mLock held 7858void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7859{ 7860 size_t size = mEffects.size(); 7861 for (size_t i = 0; i < size; i++) { 7862 mEffects[i]->setMode(mode); 7863 } 7864} 7865 7866// setVolume_l() must be called with PlaybackThread::mLock held 7867bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7868{ 7869 uint32_t newLeft = *left; 7870 uint32_t newRight = *right; 7871 bool hasControl = false; 7872 int ctrlIdx = -1; 7873 size_t size = mEffects.size(); 7874 7875 // first update volume controller 7876 for (size_t i = size; i > 0; i--) { 7877 if (mEffects[i - 1]->isProcessEnabled() && 7878 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7879 ctrlIdx = i - 1; 7880 hasControl = true; 7881 break; 7882 } 7883 } 7884 7885 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7886 if (hasControl) { 7887 *left = mNewLeftVolume; 7888 *right = mNewRightVolume; 7889 } 7890 return hasControl; 7891 } 7892 7893 mVolumeCtrlIdx = ctrlIdx; 7894 mLeftVolume = newLeft; 7895 mRightVolume = newRight; 7896 7897 // second get volume update from volume controller 7898 if (ctrlIdx >= 0) { 7899 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7900 mNewLeftVolume = newLeft; 7901 mNewRightVolume = newRight; 7902 } 7903 // then indicate volume to all other effects in chain. 7904 // Pass altered volume to effects before volume controller 7905 // and requested volume to effects after controller 7906 uint32_t lVol = newLeft; 7907 uint32_t rVol = newRight; 7908 7909 for (size_t i = 0; i < size; i++) { 7910 if ((int)i == ctrlIdx) continue; 7911 // this also works for ctrlIdx == -1 when there is no volume controller 7912 if ((int)i > ctrlIdx) { 7913 lVol = *left; 7914 rVol = *right; 7915 } 7916 mEffects[i]->setVolume(&lVol, &rVol, false); 7917 } 7918 *left = newLeft; 7919 *right = newRight; 7920 7921 return hasControl; 7922} 7923 7924status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7925{ 7926 const size_t SIZE = 256; 7927 char buffer[SIZE]; 7928 String8 result; 7929 7930 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7931 result.append(buffer); 7932 7933 bool locked = tryLock(mLock); 7934 // failed to lock - AudioFlinger is probably deadlocked 7935 if (!locked) { 7936 result.append("\tCould not lock mutex:\n"); 7937 } 7938 7939 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7940 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7941 mEffects.size(), 7942 (uint32_t)mInBuffer, 7943 (uint32_t)mOutBuffer, 7944 mActiveTrackCnt); 7945 result.append(buffer); 7946 write(fd, result.string(), result.size()); 7947 7948 for (size_t i = 0; i < mEffects.size(); ++i) { 7949 sp<EffectModule> effect = mEffects[i]; 7950 if (effect != 0) { 7951 effect->dump(fd, args); 7952 } 7953 } 7954 7955 if (locked) { 7956 mLock.unlock(); 7957 } 7958 7959 return NO_ERROR; 7960} 7961 7962// must be called with ThreadBase::mLock held 7963void AudioFlinger::EffectChain::setEffectSuspended_l( 7964 const effect_uuid_t *type, bool suspend) 7965{ 7966 sp<SuspendedEffectDesc> desc; 7967 // use effect type UUID timelow as key as there is no real risk of identical 7968 // timeLow fields among effect type UUIDs. 7969 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7970 if (suspend) { 7971 if (index >= 0) { 7972 desc = mSuspendedEffects.valueAt(index); 7973 } else { 7974 desc = new SuspendedEffectDesc(); 7975 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7976 mSuspendedEffects.add(type->timeLow, desc); 7977 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7978 } 7979 if (desc->mRefCount++ == 0) { 7980 sp<EffectModule> effect = getEffectIfEnabled(type); 7981 if (effect != 0) { 7982 desc->mEffect = effect; 7983 effect->setSuspended(true); 7984 effect->setEnabled(false); 7985 } 7986 } 7987 } else { 7988 if (index < 0) { 7989 return; 7990 } 7991 desc = mSuspendedEffects.valueAt(index); 7992 if (desc->mRefCount <= 0) { 7993 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7994 desc->mRefCount = 1; 7995 } 7996 if (--desc->mRefCount == 0) { 7997 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7998 if (desc->mEffect != 0) { 7999 sp<EffectModule> effect = desc->mEffect.promote(); 8000 if (effect != 0) { 8001 effect->setSuspended(false); 8002 sp<EffectHandle> handle = effect->controlHandle(); 8003 if (handle != 0) { 8004 effect->setEnabled(handle->enabled()); 8005 } 8006 } 8007 desc->mEffect.clear(); 8008 } 8009 mSuspendedEffects.removeItemsAt(index); 8010 } 8011 } 8012} 8013 8014// must be called with ThreadBase::mLock held 8015void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8016{ 8017 sp<SuspendedEffectDesc> desc; 8018 8019 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8020 if (suspend) { 8021 if (index >= 0) { 8022 desc = mSuspendedEffects.valueAt(index); 8023 } else { 8024 desc = new SuspendedEffectDesc(); 8025 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8026 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8027 } 8028 if (desc->mRefCount++ == 0) { 8029 Vector< sp<EffectModule> > effects; 8030 getSuspendEligibleEffects(effects); 8031 for (size_t i = 0; i < effects.size(); i++) { 8032 setEffectSuspended_l(&effects[i]->desc().type, true); 8033 } 8034 } 8035 } else { 8036 if (index < 0) { 8037 return; 8038 } 8039 desc = mSuspendedEffects.valueAt(index); 8040 if (desc->mRefCount <= 0) { 8041 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8042 desc->mRefCount = 1; 8043 } 8044 if (--desc->mRefCount == 0) { 8045 Vector<const effect_uuid_t *> types; 8046 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8047 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8048 continue; 8049 } 8050 types.add(&mSuspendedEffects.valueAt(i)->mType); 8051 } 8052 for (size_t i = 0; i < types.size(); i++) { 8053 setEffectSuspended_l(types[i], false); 8054 } 8055 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8056 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8057 } 8058 } 8059} 8060 8061 8062// The volume effect is used for automated tests only 8063#ifndef OPENSL_ES_H_ 8064static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8065 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8066const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8067#endif //OPENSL_ES_H_ 8068 8069bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8070{ 8071 // auxiliary effects and visualizer are never suspended on output mix 8072 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8073 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8074 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8075 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8076 return false; 8077 } 8078 return true; 8079} 8080 8081void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8082{ 8083 effects.clear(); 8084 for (size_t i = 0; i < mEffects.size(); i++) { 8085 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8086 effects.add(mEffects[i]); 8087 } 8088 } 8089} 8090 8091sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8092 const effect_uuid_t *type) 8093{ 8094 sp<EffectModule> effect = getEffectFromType_l(type); 8095 return effect != 0 && effect->isEnabled() ? effect : 0; 8096} 8097 8098void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8099 bool enabled) 8100{ 8101 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8102 if (enabled) { 8103 if (index < 0) { 8104 // if the effect is not suspend check if all effects are suspended 8105 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8106 if (index < 0) { 8107 return; 8108 } 8109 if (!isEffectEligibleForSuspend(effect->desc())) { 8110 return; 8111 } 8112 setEffectSuspended_l(&effect->desc().type, enabled); 8113 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8114 if (index < 0) { 8115 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8116 return; 8117 } 8118 } 8119 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8120 effect->desc().type.timeLow); 8121 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8122 // if effect is requested to suspended but was not yet enabled, supend it now. 8123 if (desc->mEffect == 0) { 8124 desc->mEffect = effect; 8125 effect->setEnabled(false); 8126 effect->setSuspended(true); 8127 } 8128 } else { 8129 if (index < 0) { 8130 return; 8131 } 8132 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8133 effect->desc().type.timeLow); 8134 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8135 desc->mEffect.clear(); 8136 effect->setSuspended(false); 8137 } 8138} 8139 8140#undef LOG_TAG 8141#define LOG_TAG "AudioFlinger" 8142 8143// ---------------------------------------------------------------------------- 8144 8145status_t AudioFlinger::onTransact( 8146 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8147{ 8148 return BnAudioFlinger::onTransact(code, data, reply, flags); 8149} 8150 8151}; // namespace android 8152