AudioFlinger.cpp revision 843a12d146bd64642bf85a4e56c274246e3893a6
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_IDLE; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_IDLE; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 assert(NULL != mPrimaryHardwareDev); 727 assert(NULL != mPrimaryHardwareDev->get_master_volume); 728 729 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 730 mHardwareStatus = AUDIO_HW_IDLE; 731 return ret_val; 732 } 733 734 return mMasterVolume; 735} 736 737status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 738 audio_io_handle_t output) 739{ 740 // check calling permissions 741 if (!settingsAllowed()) { 742 return PERMISSION_DENIED; 743 } 744 745 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 746 ALOGE("setStreamVolume() invalid stream %d", stream); 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 PlaybackThread *thread = NULL; 752 if (output) { 753 thread = checkPlaybackThread_l(output); 754 if (thread == NULL) { 755 return BAD_VALUE; 756 } 757 } 758 759 mStreamTypes[stream].volume = value; 760 761 if (thread == NULL) { 762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 763 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 764 } 765 } else { 766 thread->setStreamVolume(stream, value); 767 } 768 769 return NO_ERROR; 770} 771 772status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 773{ 774 // check calling permissions 775 if (!settingsAllowed()) { 776 return PERMISSION_DENIED; 777 } 778 779 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 780 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 781 ALOGE("setStreamMute() invalid stream %d", stream); 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 mStreamTypes[stream].mute = muted; 787 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 789 790 return NO_ERROR; 791} 792 793float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 794{ 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 return 0.0f; 797 } 798 799 AutoMutex lock(mLock); 800 float volume; 801 if (output) { 802 PlaybackThread *thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return 0.0f; 805 } 806 volume = thread->streamVolume(stream); 807 } else { 808 volume = streamVolume_l(stream); 809 } 810 811 return volume; 812} 813 814bool AudioFlinger::streamMute(audio_stream_type_t stream) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return true; 818 } 819 820 AutoMutex lock(mLock); 821 return streamMute_l(stream); 822} 823 824status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 825{ 826 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 827 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 828 // check calling permissions 829 if (!settingsAllowed()) { 830 return PERMISSION_DENIED; 831 } 832 833 // ioHandle == 0 means the parameters are global to the audio hardware interface 834 if (ioHandle == 0) { 835 status_t final_result = NO_ERROR; 836 { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs[i]; 841 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 842 final_result = result ?: final_result; 843 } 844 mHardwareStatus = AUDIO_HW_IDLE; 845 } 846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 847 AudioParameter param = AudioParameter(keyValuePairs); 848 String8 value; 849 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 850 Mutex::Autolock _l(mLock); 851 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 852 if (mBtNrecIsOff != btNrecIsOff) { 853 for (size_t i = 0; i < mRecordThreads.size(); i++) { 854 sp<RecordThread> thread = mRecordThreads.valueAt(i); 855 RecordThread::RecordTrack *track = thread->track(); 856 if (track != NULL) { 857 audio_devices_t device = (audio_devices_t)( 858 thread->device() & AUDIO_DEVICE_IN_ALL); 859 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 860 thread->setEffectSuspended(FX_IID_AEC, 861 suspend, 862 track->sessionId()); 863 thread->setEffectSuspended(FX_IID_NS, 864 suspend, 865 track->sessionId()); 866 } 867 } 868 mBtNrecIsOff = btNrecIsOff; 869 } 870 } 871 return final_result; 872 } 873 874 // hold a strong ref on thread in case closeOutput() or closeInput() is called 875 // and the thread is exited once the lock is released 876 sp<ThreadBase> thread; 877 { 878 Mutex::Autolock _l(mLock); 879 thread = checkPlaybackThread_l(ioHandle); 880 if (thread == NULL) { 881 thread = checkRecordThread_l(ioHandle); 882 } else if (thread == primaryPlaybackThread_l()) { 883 // indicate output device change to all input threads for pre processing 884 AudioParameter param = AudioParameter(keyValuePairs); 885 int value; 886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 887 for (size_t i = 0; i < mRecordThreads.size(); i++) { 888 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 889 } 890 } 891 } 892 } 893 if (thread != 0) { 894 return thread->setParameters(keyValuePairs); 895 } 896 return BAD_VALUE; 897} 898 899String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 900{ 901// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 902// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 903 904 if (ioHandle == 0) { 905 String8 out_s8; 906 907 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 908 char *s; 909 { 910 AutoMutex lock(mHardwareLock); 911 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 912 audio_hw_device_t *dev = mAudioHwDevs[i]; 913 s = dev->get_parameters(dev, keys.string()); 914 mHardwareStatus = AUDIO_HW_IDLE; 915 } 916 out_s8 += String8(s ? s : ""); 917 free(s); 918 } 919 return out_s8; 920 } 921 922 Mutex::Autolock _l(mLock); 923 924 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 925 if (playbackThread != NULL) { 926 return playbackThread->getParameters(keys); 927 } 928 RecordThread *recordThread = checkRecordThread_l(ioHandle); 929 if (recordThread != NULL) { 930 return recordThread->getParameters(keys); 931 } 932 return String8(""); 933} 934 935size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 936{ 937 status_t ret = initCheck(); 938 if (ret != NO_ERROR) { 939 return 0; 940 } 941 942 AutoMutex lock(mHardwareLock); 943 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 944 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 945 mHardwareStatus = AUDIO_HW_IDLE; 946 return size; 947} 948 949unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 950{ 951 if (ioHandle == 0) { 952 return 0; 953 } 954 955 Mutex::Autolock _l(mLock); 956 957 RecordThread *recordThread = checkRecordThread_l(ioHandle); 958 if (recordThread != NULL) { 959 return recordThread->getInputFramesLost(); 960 } 961 return 0; 962} 963 964status_t AudioFlinger::setVoiceVolume(float value) 965{ 966 status_t ret = initCheck(); 967 if (ret != NO_ERROR) { 968 return ret; 969 } 970 971 // check calling permissions 972 if (!settingsAllowed()) { 973 return PERMISSION_DENIED; 974 } 975 976 AutoMutex lock(mHardwareLock); 977 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 978 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 979 mHardwareStatus = AUDIO_HW_IDLE; 980 981 return ret; 982} 983 984status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 985 audio_io_handle_t output) const 986{ 987 status_t status; 988 989 Mutex::Autolock _l(mLock); 990 991 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 992 if (playbackThread != NULL) { 993 return playbackThread->getRenderPosition(halFrames, dspFrames); 994 } 995 996 return BAD_VALUE; 997} 998 999void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1000{ 1001 1002 Mutex::Autolock _l(mLock); 1003 1004 pid_t pid = IPCThreadState::self()->getCallingPid(); 1005 if (mNotificationClients.indexOfKey(pid) < 0) { 1006 sp<NotificationClient> notificationClient = new NotificationClient(this, 1007 client, 1008 pid); 1009 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1010 1011 mNotificationClients.add(pid, notificationClient); 1012 1013 sp<IBinder> binder = client->asBinder(); 1014 binder->linkToDeath(notificationClient); 1015 1016 // the config change is always sent from playback or record threads to avoid deadlock 1017 // with AudioSystem::gLock 1018 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1019 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1020 } 1021 1022 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1023 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1024 } 1025 } 1026} 1027 1028void AudioFlinger::removeNotificationClient(pid_t pid) 1029{ 1030 Mutex::Autolock _l(mLock); 1031 1032 ssize_t index = mNotificationClients.indexOfKey(pid); 1033 if (index >= 0) { 1034 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 1035 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 1036 mNotificationClients.removeItem(pid); 1037 } 1038 1039 ALOGV("%d died, releasing its sessions", pid); 1040 size_t num = mAudioSessionRefs.size(); 1041 bool removed = false; 1042 for (size_t i = 0; i< num; ) { 1043 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1044 ALOGV(" pid %d @ %d", ref->pid, i); 1045 if (ref->pid == pid) { 1046 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1047 mAudioSessionRefs.removeAt(i); 1048 delete ref; 1049 removed = true; 1050 num--; 1051 } else { 1052 i++; 1053 } 1054 } 1055 if (removed) { 1056 purgeStaleEffects_l(); 1057 } 1058} 1059 1060// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1061void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1062{ 1063 size_t size = mNotificationClients.size(); 1064 for (size_t i = 0; i < size; i++) { 1065 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1066 param2); 1067 } 1068} 1069 1070// removeClient_l() must be called with AudioFlinger::mLock held 1071void AudioFlinger::removeClient_l(pid_t pid) 1072{ 1073 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1074 mClients.removeItem(pid); 1075} 1076 1077 1078// ---------------------------------------------------------------------------- 1079 1080AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1081 uint32_t device, type_t type) 1082 : Thread(false), 1083 mType(type), 1084 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1085 // mChannelMask 1086 mChannelCount(0), 1087 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1088 mParamStatus(NO_ERROR), 1089 mStandby(false), mId(id), 1090 mDevice(device), 1091 mDeathRecipient(new PMDeathRecipient(this)) 1092{ 1093} 1094 1095AudioFlinger::ThreadBase::~ThreadBase() 1096{ 1097 mParamCond.broadcast(); 1098 // do not lock the mutex in destructor 1099 releaseWakeLock_l(); 1100 if (mPowerManager != 0) { 1101 sp<IBinder> binder = mPowerManager->asBinder(); 1102 binder->unlinkToDeath(mDeathRecipient); 1103 } 1104} 1105 1106void AudioFlinger::ThreadBase::exit() 1107{ 1108 ALOGV("ThreadBase::exit"); 1109 { 1110 // This lock prevents the following race in thread (uniprocessor for illustration): 1111 // if (!exitPending()) { 1112 // // context switch from here to exit() 1113 // // exit() calls requestExit(), what exitPending() observes 1114 // // exit() calls signal(), which is dropped since no waiters 1115 // // context switch back from exit() to here 1116 // mWaitWorkCV.wait(...); 1117 // // now thread is hung 1118 // } 1119 AutoMutex lock(mLock); 1120 requestExit(); 1121 mWaitWorkCV.signal(); 1122 } 1123 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1124 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1125 requestExitAndWait(); 1126} 1127 1128status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1129{ 1130 status_t status; 1131 1132 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1133 Mutex::Autolock _l(mLock); 1134 1135 mNewParameters.add(keyValuePairs); 1136 mWaitWorkCV.signal(); 1137 // wait condition with timeout in case the thread loop has exited 1138 // before the request could be processed 1139 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1140 status = mParamStatus; 1141 mWaitWorkCV.signal(); 1142 } else { 1143 status = TIMED_OUT; 1144 } 1145 return status; 1146} 1147 1148void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1149{ 1150 Mutex::Autolock _l(mLock); 1151 sendConfigEvent_l(event, param); 1152} 1153 1154// sendConfigEvent_l() must be called with ThreadBase::mLock held 1155void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1156{ 1157 ConfigEvent configEvent; 1158 configEvent.mEvent = event; 1159 configEvent.mParam = param; 1160 mConfigEvents.add(configEvent); 1161 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1162 mWaitWorkCV.signal(); 1163} 1164 1165void AudioFlinger::ThreadBase::processConfigEvents() 1166{ 1167 mLock.lock(); 1168 while(!mConfigEvents.isEmpty()) { 1169 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1170 ConfigEvent configEvent = mConfigEvents[0]; 1171 mConfigEvents.removeAt(0); 1172 // release mLock before locking AudioFlinger mLock: lock order is always 1173 // AudioFlinger then ThreadBase to avoid cross deadlock 1174 mLock.unlock(); 1175 mAudioFlinger->mLock.lock(); 1176 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1177 mAudioFlinger->mLock.unlock(); 1178 mLock.lock(); 1179 } 1180 mLock.unlock(); 1181} 1182 1183status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1184{ 1185 const size_t SIZE = 256; 1186 char buffer[SIZE]; 1187 String8 result; 1188 1189 bool locked = tryLock(mLock); 1190 if (!locked) { 1191 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1192 write(fd, buffer, strlen(buffer)); 1193 } 1194 1195 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1196 result.append(buffer); 1197 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1202 result.append(buffer); 1203 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1204 result.append(buffer); 1205 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1206 result.append(buffer); 1207 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1208 result.append(buffer); 1209 1210 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1211 result.append(buffer); 1212 result.append(" Index Command"); 1213 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1214 snprintf(buffer, SIZE, "\n %02d ", i); 1215 result.append(buffer); 1216 result.append(mNewParameters[i]); 1217 } 1218 1219 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1220 result.append(buffer); 1221 snprintf(buffer, SIZE, " Index event param\n"); 1222 result.append(buffer); 1223 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1224 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1225 result.append(buffer); 1226 } 1227 result.append("\n"); 1228 1229 write(fd, result.string(), result.size()); 1230 1231 if (locked) { 1232 mLock.unlock(); 1233 } 1234 return NO_ERROR; 1235} 1236 1237status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1238{ 1239 const size_t SIZE = 256; 1240 char buffer[SIZE]; 1241 String8 result; 1242 1243 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1244 write(fd, buffer, strlen(buffer)); 1245 1246 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1247 sp<EffectChain> chain = mEffectChains[i]; 1248 if (chain != 0) { 1249 chain->dump(fd, args); 1250 } 1251 } 1252 return NO_ERROR; 1253} 1254 1255void AudioFlinger::ThreadBase::acquireWakeLock() 1256{ 1257 Mutex::Autolock _l(mLock); 1258 acquireWakeLock_l(); 1259} 1260 1261void AudioFlinger::ThreadBase::acquireWakeLock_l() 1262{ 1263 if (mPowerManager == 0) { 1264 // use checkService() to avoid blocking if power service is not up yet 1265 sp<IBinder> binder = 1266 defaultServiceManager()->checkService(String16("power")); 1267 if (binder == 0) { 1268 ALOGW("Thread %s cannot connect to the power manager service", mName); 1269 } else { 1270 mPowerManager = interface_cast<IPowerManager>(binder); 1271 binder->linkToDeath(mDeathRecipient); 1272 } 1273 } 1274 if (mPowerManager != 0) { 1275 sp<IBinder> binder = new BBinder(); 1276 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1277 binder, 1278 String16(mName)); 1279 if (status == NO_ERROR) { 1280 mWakeLockToken = binder; 1281 } 1282 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1283 } 1284} 1285 1286void AudioFlinger::ThreadBase::releaseWakeLock() 1287{ 1288 Mutex::Autolock _l(mLock); 1289 releaseWakeLock_l(); 1290} 1291 1292void AudioFlinger::ThreadBase::releaseWakeLock_l() 1293{ 1294 if (mWakeLockToken != 0) { 1295 ALOGV("releaseWakeLock_l() %s", mName); 1296 if (mPowerManager != 0) { 1297 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1298 } 1299 mWakeLockToken.clear(); 1300 } 1301} 1302 1303void AudioFlinger::ThreadBase::clearPowerManager() 1304{ 1305 Mutex::Autolock _l(mLock); 1306 releaseWakeLock_l(); 1307 mPowerManager.clear(); 1308} 1309 1310void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1311{ 1312 sp<ThreadBase> thread = mThread.promote(); 1313 if (thread != 0) { 1314 thread->clearPowerManager(); 1315 } 1316 ALOGW("power manager service died !!!"); 1317} 1318 1319void AudioFlinger::ThreadBase::setEffectSuspended( 1320 const effect_uuid_t *type, bool suspend, int sessionId) 1321{ 1322 Mutex::Autolock _l(mLock); 1323 setEffectSuspended_l(type, suspend, sessionId); 1324} 1325 1326void AudioFlinger::ThreadBase::setEffectSuspended_l( 1327 const effect_uuid_t *type, bool suspend, int sessionId) 1328{ 1329 sp<EffectChain> chain = getEffectChain_l(sessionId); 1330 if (chain != 0) { 1331 if (type != NULL) { 1332 chain->setEffectSuspended_l(type, suspend); 1333 } else { 1334 chain->setEffectSuspendedAll_l(suspend); 1335 } 1336 } 1337 1338 updateSuspendedSessions_l(type, suspend, sessionId); 1339} 1340 1341void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1342{ 1343 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1344 if (index < 0) { 1345 return; 1346 } 1347 1348 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1349 mSuspendedSessions.editValueAt(index); 1350 1351 for (size_t i = 0; i < sessionEffects.size(); i++) { 1352 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1353 for (int j = 0; j < desc->mRefCount; j++) { 1354 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1355 chain->setEffectSuspendedAll_l(true); 1356 } else { 1357 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1358 desc->mType.timeLow); 1359 chain->setEffectSuspended_l(&desc->mType, true); 1360 } 1361 } 1362 } 1363} 1364 1365void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1366 bool suspend, 1367 int sessionId) 1368{ 1369 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1370 1371 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1372 1373 if (suspend) { 1374 if (index >= 0) { 1375 sessionEffects = mSuspendedSessions.editValueAt(index); 1376 } else { 1377 mSuspendedSessions.add(sessionId, sessionEffects); 1378 } 1379 } else { 1380 if (index < 0) { 1381 return; 1382 } 1383 sessionEffects = mSuspendedSessions.editValueAt(index); 1384 } 1385 1386 1387 int key = EffectChain::kKeyForSuspendAll; 1388 if (type != NULL) { 1389 key = type->timeLow; 1390 } 1391 index = sessionEffects.indexOfKey(key); 1392 1393 sp <SuspendedSessionDesc> desc; 1394 if (suspend) { 1395 if (index >= 0) { 1396 desc = sessionEffects.valueAt(index); 1397 } else { 1398 desc = new SuspendedSessionDesc(); 1399 if (type != NULL) { 1400 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1401 } 1402 sessionEffects.add(key, desc); 1403 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1404 } 1405 desc->mRefCount++; 1406 } else { 1407 if (index < 0) { 1408 return; 1409 } 1410 desc = sessionEffects.valueAt(index); 1411 if (--desc->mRefCount == 0) { 1412 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1413 sessionEffects.removeItemsAt(index); 1414 if (sessionEffects.isEmpty()) { 1415 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1416 sessionId); 1417 mSuspendedSessions.removeItem(sessionId); 1418 } 1419 } 1420 } 1421 if (!sessionEffects.isEmpty()) { 1422 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1423 } 1424} 1425 1426void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1427 bool enabled, 1428 int sessionId) 1429{ 1430 Mutex::Autolock _l(mLock); 1431 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1432} 1433 1434void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1435 bool enabled, 1436 int sessionId) 1437{ 1438 if (mType != RECORD) { 1439 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1440 // another session. This gives the priority to well behaved effect control panels 1441 // and applications not using global effects. 1442 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1443 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1444 } 1445 } 1446 1447 sp<EffectChain> chain = getEffectChain_l(sessionId); 1448 if (chain != 0) { 1449 chain->checkSuspendOnEffectEnabled(effect, enabled); 1450 } 1451} 1452 1453// ---------------------------------------------------------------------------- 1454 1455AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1456 AudioStreamOut* output, 1457 audio_io_handle_t id, 1458 uint32_t device, 1459 type_t type) 1460 : ThreadBase(audioFlinger, id, device, type), 1461 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1462 // Assumes constructor is called by AudioFlinger with it's mLock held, 1463 // but it would be safer to explicitly pass initial masterMute as parameter 1464 mMasterMute(audioFlinger->masterMute_l()), 1465 // mStreamTypes[] initialized in constructor body 1466 mOutput(output), 1467 // Assumes constructor is called by AudioFlinger with it's mLock held, 1468 // but it would be safer to explicitly pass initial masterVolume as parameter 1469 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1470 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1471{ 1472 snprintf(mName, kNameLength, "AudioOut_%d", id); 1473 1474 readOutputParameters(); 1475 1476 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1477 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1478 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1479 stream = (audio_stream_type_t) (stream + 1)) { 1480 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1481 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1482 // initialized by stream_type_t default constructor 1483 // mStreamTypes[stream].valid = true; 1484 } 1485 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1486 // because mAudioFlinger doesn't have one to copy from 1487} 1488 1489AudioFlinger::PlaybackThread::~PlaybackThread() 1490{ 1491 delete [] mMixBuffer; 1492} 1493 1494status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1495{ 1496 dumpInternals(fd, args); 1497 dumpTracks(fd, args); 1498 dumpEffectChains(fd, args); 1499 return NO_ERROR; 1500} 1501 1502status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1503{ 1504 const size_t SIZE = 256; 1505 char buffer[SIZE]; 1506 String8 result; 1507 1508 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1509 result.append(buffer); 1510 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1511 for (size_t i = 0; i < mTracks.size(); ++i) { 1512 sp<Track> track = mTracks[i]; 1513 if (track != 0) { 1514 track->dump(buffer, SIZE); 1515 result.append(buffer); 1516 } 1517 } 1518 1519 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1520 result.append(buffer); 1521 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1522 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1523 sp<Track> track = mActiveTracks[i].promote(); 1524 if (track != 0) { 1525 track->dump(buffer, SIZE); 1526 result.append(buffer); 1527 } 1528 } 1529 write(fd, result.string(), result.size()); 1530 return NO_ERROR; 1531} 1532 1533status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1534{ 1535 const size_t SIZE = 256; 1536 char buffer[SIZE]; 1537 String8 result; 1538 1539 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1540 result.append(buffer); 1541 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1542 result.append(buffer); 1543 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1544 result.append(buffer); 1545 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1546 result.append(buffer); 1547 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1548 result.append(buffer); 1549 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1550 result.append(buffer); 1551 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1552 result.append(buffer); 1553 write(fd, result.string(), result.size()); 1554 1555 dumpBase(fd, args); 1556 1557 return NO_ERROR; 1558} 1559 1560// Thread virtuals 1561status_t AudioFlinger::PlaybackThread::readyToRun() 1562{ 1563 status_t status = initCheck(); 1564 if (status == NO_ERROR) { 1565 ALOGI("AudioFlinger's thread %p ready to run", this); 1566 } else { 1567 ALOGE("No working audio driver found."); 1568 } 1569 return status; 1570} 1571 1572void AudioFlinger::PlaybackThread::onFirstRef() 1573{ 1574 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1575} 1576 1577// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1578sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1579 const sp<AudioFlinger::Client>& client, 1580 audio_stream_type_t streamType, 1581 uint32_t sampleRate, 1582 audio_format_t format, 1583 uint32_t channelMask, 1584 int frameCount, 1585 const sp<IMemory>& sharedBuffer, 1586 int sessionId, 1587 bool isTimed, 1588 status_t *status) 1589{ 1590 sp<Track> track; 1591 status_t lStatus; 1592 1593 if (mType == DIRECT) { 1594 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1595 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1596 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1597 "for output %p with format %d", 1598 sampleRate, format, channelMask, mOutput, mFormat); 1599 lStatus = BAD_VALUE; 1600 goto Exit; 1601 } 1602 } 1603 } else { 1604 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1605 if (sampleRate > mSampleRate*2) { 1606 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1607 lStatus = BAD_VALUE; 1608 goto Exit; 1609 } 1610 } 1611 1612 lStatus = initCheck(); 1613 if (lStatus != NO_ERROR) { 1614 ALOGE("Audio driver not initialized."); 1615 goto Exit; 1616 } 1617 1618 { // scope for mLock 1619 Mutex::Autolock _l(mLock); 1620 1621 // all tracks in same audio session must share the same routing strategy otherwise 1622 // conflicts will happen when tracks are moved from one output to another by audio policy 1623 // manager 1624 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1625 for (size_t i = 0; i < mTracks.size(); ++i) { 1626 sp<Track> t = mTracks[i]; 1627 if (t != 0) { 1628 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1629 if (sessionId == t->sessionId() && strategy != actual) { 1630 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1631 strategy, actual); 1632 lStatus = BAD_VALUE; 1633 goto Exit; 1634 } 1635 } 1636 } 1637 1638 if (!isTimed) { 1639 track = new Track(this, client, streamType, sampleRate, format, 1640 channelMask, frameCount, sharedBuffer, sessionId); 1641 } else { 1642 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1643 channelMask, frameCount, sharedBuffer, sessionId); 1644 } 1645 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1646 lStatus = NO_MEMORY; 1647 goto Exit; 1648 } 1649 mTracks.add(track); 1650 1651 sp<EffectChain> chain = getEffectChain_l(sessionId); 1652 if (chain != 0) { 1653 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1654 track->setMainBuffer(chain->inBuffer()); 1655 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1656 chain->incTrackCnt(); 1657 } 1658 1659 // invalidate track immediately if the stream type was moved to another thread since 1660 // createTrack() was called by the client process. 1661 if (!mStreamTypes[streamType].valid) { 1662 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1663 this, streamType); 1664 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1665 } 1666 } 1667 lStatus = NO_ERROR; 1668 1669Exit: 1670 if(status) { 1671 *status = lStatus; 1672 } 1673 return track; 1674} 1675 1676uint32_t AudioFlinger::PlaybackThread::latency() const 1677{ 1678 Mutex::Autolock _l(mLock); 1679 if (initCheck() == NO_ERROR) { 1680 return mOutput->stream->get_latency(mOutput->stream); 1681 } else { 1682 return 0; 1683 } 1684} 1685 1686void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1687{ 1688 Mutex::Autolock _l(mLock); 1689 mMasterVolume = value; 1690} 1691 1692void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1693{ 1694 Mutex::Autolock _l(mLock); 1695 setMasterMute_l(muted); 1696} 1697 1698void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1699{ 1700 Mutex::Autolock _l(mLock); 1701 mStreamTypes[stream].volume = value; 1702} 1703 1704void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1705{ 1706 Mutex::Autolock _l(mLock); 1707 mStreamTypes[stream].mute = muted; 1708} 1709 1710float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1711{ 1712 Mutex::Autolock _l(mLock); 1713 return mStreamTypes[stream].volume; 1714} 1715 1716// addTrack_l() must be called with ThreadBase::mLock held 1717status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1718{ 1719 status_t status = ALREADY_EXISTS; 1720 1721 // set retry count for buffer fill 1722 track->mRetryCount = kMaxTrackStartupRetries; 1723 if (mActiveTracks.indexOf(track) < 0) { 1724 // the track is newly added, make sure it fills up all its 1725 // buffers before playing. This is to ensure the client will 1726 // effectively get the latency it requested. 1727 track->mFillingUpStatus = Track::FS_FILLING; 1728 track->mResetDone = false; 1729 mActiveTracks.add(track); 1730 if (track->mainBuffer() != mMixBuffer) { 1731 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1732 if (chain != 0) { 1733 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1734 chain->incActiveTrackCnt(); 1735 } 1736 } 1737 1738 status = NO_ERROR; 1739 } 1740 1741 ALOGV("mWaitWorkCV.broadcast"); 1742 mWaitWorkCV.broadcast(); 1743 1744 return status; 1745} 1746 1747// destroyTrack_l() must be called with ThreadBase::mLock held 1748void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1749{ 1750 track->mState = TrackBase::TERMINATED; 1751 if (mActiveTracks.indexOf(track) < 0) { 1752 removeTrack_l(track); 1753 } 1754} 1755 1756void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1757{ 1758 mTracks.remove(track); 1759 deleteTrackName_l(track->name()); 1760 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1761 if (chain != 0) { 1762 chain->decTrackCnt(); 1763 } 1764} 1765 1766String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1767{ 1768 String8 out_s8 = String8(""); 1769 char *s; 1770 1771 Mutex::Autolock _l(mLock); 1772 if (initCheck() != NO_ERROR) { 1773 return out_s8; 1774 } 1775 1776 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1777 out_s8 = String8(s); 1778 free(s); 1779 return out_s8; 1780} 1781 1782// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1783void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1784 AudioSystem::OutputDescriptor desc; 1785 void *param2 = NULL; 1786 1787 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1788 1789 switch (event) { 1790 case AudioSystem::OUTPUT_OPENED: 1791 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1792 desc.channels = mChannelMask; 1793 desc.samplingRate = mSampleRate; 1794 desc.format = mFormat; 1795 desc.frameCount = mFrameCount; 1796 desc.latency = latency(); 1797 param2 = &desc; 1798 break; 1799 1800 case AudioSystem::STREAM_CONFIG_CHANGED: 1801 param2 = ¶m; 1802 case AudioSystem::OUTPUT_CLOSED: 1803 default: 1804 break; 1805 } 1806 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1807} 1808 1809void AudioFlinger::PlaybackThread::readOutputParameters() 1810{ 1811 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1812 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1813 mChannelCount = (uint16_t)popcount(mChannelMask); 1814 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1815 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1816 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1817 1818 // FIXME - Current mixer implementation only supports stereo output: Always 1819 // Allocate a stereo buffer even if HW output is mono. 1820 delete[] mMixBuffer; 1821 mMixBuffer = new int16_t[mFrameCount * 2]; 1822 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1823 1824 // force reconfiguration of effect chains and engines to take new buffer size and audio 1825 // parameters into account 1826 // Note that mLock is not held when readOutputParameters() is called from the constructor 1827 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1828 // matter. 1829 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1830 Vector< sp<EffectChain> > effectChains = mEffectChains; 1831 for (size_t i = 0; i < effectChains.size(); i ++) { 1832 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1833 } 1834} 1835 1836status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1837{ 1838 if (halFrames == NULL || dspFrames == NULL) { 1839 return BAD_VALUE; 1840 } 1841 Mutex::Autolock _l(mLock); 1842 if (initCheck() != NO_ERROR) { 1843 return INVALID_OPERATION; 1844 } 1845 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1846 1847 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1848} 1849 1850uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1851{ 1852 Mutex::Autolock _l(mLock); 1853 uint32_t result = 0; 1854 if (getEffectChain_l(sessionId) != 0) { 1855 result = EFFECT_SESSION; 1856 } 1857 1858 for (size_t i = 0; i < mTracks.size(); ++i) { 1859 sp<Track> track = mTracks[i]; 1860 if (sessionId == track->sessionId() && 1861 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1862 result |= TRACK_SESSION; 1863 break; 1864 } 1865 } 1866 1867 return result; 1868} 1869 1870uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1871{ 1872 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1873 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1874 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1875 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1876 } 1877 for (size_t i = 0; i < mTracks.size(); i++) { 1878 sp<Track> track = mTracks[i]; 1879 if (sessionId == track->sessionId() && 1880 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1881 return AudioSystem::getStrategyForStream(track->streamType()); 1882 } 1883 } 1884 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1885} 1886 1887 1888AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1889{ 1890 Mutex::Autolock _l(mLock); 1891 return mOutput; 1892} 1893 1894AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1895{ 1896 Mutex::Autolock _l(mLock); 1897 AudioStreamOut *output = mOutput; 1898 mOutput = NULL; 1899 return output; 1900} 1901 1902// this method must always be called either with ThreadBase mLock held or inside the thread loop 1903audio_stream_t* AudioFlinger::PlaybackThread::stream() 1904{ 1905 if (mOutput == NULL) { 1906 return NULL; 1907 } 1908 return &mOutput->stream->common; 1909} 1910 1911uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1912{ 1913 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1914 // decoding and transfer time. So sleeping for half of the latency would likely cause 1915 // underruns 1916 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1917 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1918 } else { 1919 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1920 } 1921} 1922 1923// ---------------------------------------------------------------------------- 1924 1925AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1926 audio_io_handle_t id, uint32_t device, type_t type) 1927 : PlaybackThread(audioFlinger, output, id, device, type), 1928 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1929 mPrevMixerStatus(MIXER_IDLE) 1930{ 1931 // FIXME - Current mixer implementation only supports stereo output 1932 if (mChannelCount == 1) { 1933 ALOGE("Invalid audio hardware channel count"); 1934 } 1935} 1936 1937AudioFlinger::MixerThread::~MixerThread() 1938{ 1939 delete mAudioMixer; 1940} 1941 1942class CpuStats { 1943public: 1944 void sample(); 1945#ifdef DEBUG_CPU_USAGE 1946private: 1947 ThreadCpuUsage mCpu; 1948#endif 1949}; 1950 1951void CpuStats::sample() { 1952#ifdef DEBUG_CPU_USAGE 1953 const CentralTendencyStatistics& stats = mCpu.statistics(); 1954 mCpu.sampleAndEnable(); 1955 unsigned n = stats.n(); 1956 // mCpu.elapsed() is expensive, so don't call it every loop 1957 if ((n & 127) == 1) { 1958 long long elapsed = mCpu.elapsed(); 1959 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1960 double perLoop = elapsed / (double) n; 1961 double perLoop100 = perLoop * 0.01; 1962 double mean = stats.mean(); 1963 double stddev = stats.stddev(); 1964 double minimum = stats.minimum(); 1965 double maximum = stats.maximum(); 1966 mCpu.resetStatistics(); 1967 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", 1968 elapsed * .000000001, n, perLoop * .000001, 1969 mean * .001, 1970 stddev * .001, 1971 minimum * .001, 1972 maximum * .001, 1973 mean / perLoop100, 1974 stddev / perLoop100, 1975 minimum / perLoop100, 1976 maximum / perLoop100); 1977 } 1978 } 1979#endif 1980}; 1981 1982void AudioFlinger::PlaybackThread::checkSilentMode_l() 1983{ 1984 if (!mMasterMute) { 1985 char value[PROPERTY_VALUE_MAX]; 1986 if (property_get("ro.audio.silent", value, "0") > 0) { 1987 char *endptr; 1988 unsigned long ul = strtoul(value, &endptr, 0); 1989 if (*endptr == '\0' && ul != 0) { 1990 ALOGD("Silence is golden"); 1991 // The setprop command will not allow a property to be changed after 1992 // the first time it is set, so we don't have to worry about un-muting. 1993 setMasterMute_l(true); 1994 } 1995 } 1996 } 1997} 1998 1999bool AudioFlinger::MixerThread::threadLoop() 2000{ 2001 Vector< sp<Track> > tracksToRemove; 2002 nsecs_t standbyTime = systemTime(); 2003 size_t mixBufferSize = mFrameCount * mFrameSize; 2004 // FIXME: Relaxed timing because of a certain device that can't meet latency 2005 // Should be reduced to 2x after the vendor fixes the driver issue 2006 // increase threshold again due to low power audio mode. The way this warning threshold is 2007 // calculated and its usefulness should be reconsidered anyway. 2008 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2009 nsecs_t lastWarning = 0; 2010 bool longStandbyExit = false; 2011 uint32_t activeSleepTime = activeSleepTimeUs(); 2012 uint32_t idleSleepTime = idleSleepTimeUs(); 2013 uint32_t sleepTime = idleSleepTime; 2014 uint32_t sleepTimeShift = 0; 2015 Vector< sp<EffectChain> > effectChains; 2016 CpuStats cpuStats; 2017 2018 acquireWakeLock(); 2019 2020 while (!exitPending()) 2021 { 2022 cpuStats.sample(); 2023 processConfigEvents(); 2024 2025 mixer_state mixerStatus = MIXER_IDLE; 2026 { // scope for mLock 2027 2028 Mutex::Autolock _l(mLock); 2029 2030 if (checkForNewParameters_l()) { 2031 mixBufferSize = mFrameCount * mFrameSize; 2032 // FIXME: Relaxed timing because of a certain device that can't meet latency 2033 // Should be reduced to 2x after the vendor fixes the driver issue 2034 // increase threshold again due to low power audio mode. The way this warning 2035 // threshold is calculated and its usefulness should be reconsidered anyway. 2036 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2037 activeSleepTime = activeSleepTimeUs(); 2038 idleSleepTime = idleSleepTimeUs(); 2039 } 2040 2041 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2042 2043 // put audio hardware into standby after short delay 2044 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2045 mSuspended)) { 2046 if (!mStandby) { 2047 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2048 mOutput->stream->common.standby(&mOutput->stream->common); 2049 mStandby = true; 2050 mBytesWritten = 0; 2051 } 2052 2053 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2054 // we're about to wait, flush the binder command buffer 2055 IPCThreadState::self()->flushCommands(); 2056 2057 if (exitPending()) break; 2058 2059 releaseWakeLock_l(); 2060 // wait until we have something to do... 2061 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2062 mWaitWorkCV.wait(mLock); 2063 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2064 acquireWakeLock_l(); 2065 2066 mPrevMixerStatus = MIXER_IDLE; 2067 checkSilentMode_l(); 2068 2069 standbyTime = systemTime() + mStandbyTimeInNsecs; 2070 sleepTime = idleSleepTime; 2071 sleepTimeShift = 0; 2072 continue; 2073 } 2074 } 2075 2076 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2077 2078 // prevent any changes in effect chain list and in each effect chain 2079 // during mixing and effect process as the audio buffers could be deleted 2080 // or modified if an effect is created or deleted 2081 lockEffectChains_l(effectChains); 2082 } 2083 2084 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2085 // obtain the presentation timestamp of the next output buffer 2086 int64_t pts; 2087 status_t status = INVALID_OPERATION; 2088 2089 if (NULL != mOutput->stream->get_next_write_timestamp) { 2090 status = mOutput->stream->get_next_write_timestamp( 2091 mOutput->stream, &pts); 2092 } 2093 2094 if (status != NO_ERROR) { 2095 pts = AudioBufferProvider::kInvalidPTS; 2096 } 2097 2098 // mix buffers... 2099 mAudioMixer->process(pts); 2100 // increase sleep time progressively when application underrun condition clears. 2101 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2102 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2103 // such that we would underrun the audio HAL. 2104 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2105 sleepTimeShift--; 2106 } 2107 sleepTime = 0; 2108 standbyTime = systemTime() + mStandbyTimeInNsecs; 2109 //TODO: delay standby when effects have a tail 2110 } else { 2111 // If no tracks are ready, sleep once for the duration of an output 2112 // buffer size, then write 0s to the output 2113 if (sleepTime == 0) { 2114 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2115 sleepTime = activeSleepTime >> sleepTimeShift; 2116 if (sleepTime < kMinThreadSleepTimeUs) { 2117 sleepTime = kMinThreadSleepTimeUs; 2118 } 2119 // reduce sleep time in case of consecutive application underruns to avoid 2120 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2121 // duration we would end up writing less data than needed by the audio HAL if 2122 // the condition persists. 2123 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2124 sleepTimeShift++; 2125 } 2126 } else { 2127 sleepTime = idleSleepTime; 2128 } 2129 } else if (mBytesWritten != 0 || 2130 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2131 memset (mMixBuffer, 0, mixBufferSize); 2132 sleepTime = 0; 2133 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2134 } 2135 // TODO add standby time extension fct of effect tail 2136 } 2137 2138 if (mSuspended) { 2139 sleepTime = suspendSleepTimeUs(); 2140 } 2141 // sleepTime == 0 means we must write to audio hardware 2142 if (sleepTime == 0) { 2143 for (size_t i = 0; i < effectChains.size(); i ++) { 2144 effectChains[i]->process_l(); 2145 } 2146 // enable changes in effect chain 2147 unlockEffectChains(effectChains); 2148 mLastWriteTime = systemTime(); 2149 mInWrite = true; 2150 mBytesWritten += mixBufferSize; 2151 2152 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2153 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2154 mNumWrites++; 2155 mInWrite = false; 2156 nsecs_t now = systemTime(); 2157 nsecs_t delta = now - mLastWriteTime; 2158 if (!mStandby && delta > maxPeriod) { 2159 mNumDelayedWrites++; 2160 if ((now - lastWarning) > kWarningThrottleNs) { 2161 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2162 ns2ms(delta), mNumDelayedWrites, this); 2163 lastWarning = now; 2164 } 2165 if (mStandby) { 2166 longStandbyExit = true; 2167 } 2168 } 2169 mStandby = false; 2170 } else { 2171 // enable changes in effect chain 2172 unlockEffectChains(effectChains); 2173 usleep(sleepTime); 2174 } 2175 2176 // finally let go of all our tracks, without the lock held 2177 // since we can't guarantee the destructors won't acquire that 2178 // same lock. 2179 tracksToRemove.clear(); 2180 2181 // Effect chains will be actually deleted here if they were removed from 2182 // mEffectChains list during mixing or effects processing 2183 effectChains.clear(); 2184 } 2185 2186 if (!mStandby) { 2187 mOutput->stream->common.standby(&mOutput->stream->common); 2188 } 2189 2190 releaseWakeLock(); 2191 2192 ALOGV("Thread %p type %d exiting", this, mType); 2193 return false; 2194} 2195 2196// prepareTracks_l() must be called with ThreadBase::mLock held 2197AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2198 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2199{ 2200 2201 mixer_state mixerStatus = MIXER_IDLE; 2202 // find out which tracks need to be processed 2203 size_t count = activeTracks.size(); 2204 size_t mixedTracks = 0; 2205 size_t tracksWithEffect = 0; 2206 2207 float masterVolume = mMasterVolume; 2208 bool masterMute = mMasterMute; 2209 2210 if (masterMute) { 2211 masterVolume = 0; 2212 } 2213 // Delegate master volume control to effect in output mix effect chain if needed 2214 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2215 if (chain != 0) { 2216 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2217 chain->setVolume_l(&v, &v); 2218 masterVolume = (float)((v + (1 << 23)) >> 24); 2219 chain.clear(); 2220 } 2221 2222 for (size_t i=0 ; i<count ; i++) { 2223 sp<Track> t = activeTracks[i].promote(); 2224 if (t == 0) continue; 2225 2226 // this const just means the local variable doesn't change 2227 Track* const track = t.get(); 2228 audio_track_cblk_t* cblk = track->cblk(); 2229 2230 // The first time a track is added we wait 2231 // for all its buffers to be filled before processing it 2232 int name = track->name(); 2233 // make sure that we have enough frames to mix one full buffer. 2234 // enforce this condition only once to enable draining the buffer in case the client 2235 // app does not call stop() and relies on underrun to stop: 2236 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2237 // during last round 2238 uint32_t minFrames = 1; 2239 if (!track->isStopped() && !track->isPausing() && 2240 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2241 if (t->sampleRate() == (int)mSampleRate) { 2242 minFrames = mFrameCount; 2243 } else { 2244 // +1 for rounding and +1 for additional sample needed for interpolation 2245 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2246 // add frames already consumed but not yet released by the resampler 2247 // because cblk->framesReady() will include these frames 2248 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2249 // the minimum track buffer size is normally twice the number of frames necessary 2250 // to fill one buffer and the resampler should not leave more than one buffer worth 2251 // of unreleased frames after each pass, but just in case... 2252 ALOG_ASSERT(minFrames <= cblk->frameCount); 2253 } 2254 } 2255 if ((track->framesReady() >= minFrames) && track->isReady() && 2256 !track->isPaused() && !track->isTerminated()) 2257 { 2258 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2259 2260 mixedTracks++; 2261 2262 // track->mainBuffer() != mMixBuffer means there is an effect chain 2263 // connected to the track 2264 chain.clear(); 2265 if (track->mainBuffer() != mMixBuffer) { 2266 chain = getEffectChain_l(track->sessionId()); 2267 // Delegate volume control to effect in track effect chain if needed 2268 if (chain != 0) { 2269 tracksWithEffect++; 2270 } else { 2271 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2272 name, track->sessionId()); 2273 } 2274 } 2275 2276 2277 int param = AudioMixer::VOLUME; 2278 if (track->mFillingUpStatus == Track::FS_FILLED) { 2279 // no ramp for the first volume setting 2280 track->mFillingUpStatus = Track::FS_ACTIVE; 2281 if (track->mState == TrackBase::RESUMING) { 2282 track->mState = TrackBase::ACTIVE; 2283 param = AudioMixer::RAMP_VOLUME; 2284 } 2285 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2286 } else if (cblk->server != 0) { 2287 // If the track is stopped before the first frame was mixed, 2288 // do not apply ramp 2289 param = AudioMixer::RAMP_VOLUME; 2290 } 2291 2292 // compute volume for this track 2293 uint32_t vl, vr, va; 2294 if (track->isMuted() || track->isPausing() || 2295 mStreamTypes[track->streamType()].mute) { 2296 vl = vr = va = 0; 2297 if (track->isPausing()) { 2298 track->setPaused(); 2299 } 2300 } else { 2301 2302 // read original volumes with volume control 2303 float typeVolume = mStreamTypes[track->streamType()].volume; 2304 float v = masterVolume * typeVolume; 2305 uint32_t vlr = cblk->getVolumeLR(); 2306 vl = vlr & 0xFFFF; 2307 vr = vlr >> 16; 2308 // track volumes come from shared memory, so can't be trusted and must be clamped 2309 if (vl > MAX_GAIN_INT) { 2310 ALOGV("Track left volume out of range: %04X", vl); 2311 vl = MAX_GAIN_INT; 2312 } 2313 if (vr > MAX_GAIN_INT) { 2314 ALOGV("Track right volume out of range: %04X", vr); 2315 vr = MAX_GAIN_INT; 2316 } 2317 // now apply the master volume and stream type volume 2318 vl = (uint32_t)(v * vl) << 12; 2319 vr = (uint32_t)(v * vr) << 12; 2320 // assuming master volume and stream type volume each go up to 1.0, 2321 // vl and vr are now in 8.24 format 2322 2323 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2324 // send level comes from shared memory and so may be corrupt 2325 if (sendLevel > MAX_GAIN_INT) { 2326 ALOGV("Track send level out of range: %04X", sendLevel); 2327 sendLevel = MAX_GAIN_INT; 2328 } 2329 va = (uint32_t)(v * sendLevel); 2330 } 2331 // Delegate volume control to effect in track effect chain if needed 2332 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2333 // Do not ramp volume if volume is controlled by effect 2334 param = AudioMixer::VOLUME; 2335 track->mHasVolumeController = true; 2336 } else { 2337 // force no volume ramp when volume controller was just disabled or removed 2338 // from effect chain to avoid volume spike 2339 if (track->mHasVolumeController) { 2340 param = AudioMixer::VOLUME; 2341 } 2342 track->mHasVolumeController = false; 2343 } 2344 2345 // Convert volumes from 8.24 to 4.12 format 2346 // This additional clamping is needed in case chain->setVolume_l() overshot 2347 vl = (vl + (1 << 11)) >> 12; 2348 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2349 vr = (vr + (1 << 11)) >> 12; 2350 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2351 2352 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2353 2354 // XXX: these things DON'T need to be done each time 2355 mAudioMixer->setBufferProvider(name, track); 2356 mAudioMixer->enable(name); 2357 2358 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2359 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2360 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2361 mAudioMixer->setParameter( 2362 name, 2363 AudioMixer::TRACK, 2364 AudioMixer::FORMAT, (void *)track->format()); 2365 mAudioMixer->setParameter( 2366 name, 2367 AudioMixer::TRACK, 2368 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2369 mAudioMixer->setParameter( 2370 name, 2371 AudioMixer::RESAMPLE, 2372 AudioMixer::SAMPLE_RATE, 2373 (void *)(cblk->sampleRate)); 2374 mAudioMixer->setParameter( 2375 name, 2376 AudioMixer::TRACK, 2377 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2378 mAudioMixer->setParameter( 2379 name, 2380 AudioMixer::TRACK, 2381 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2382 2383 // reset retry count 2384 track->mRetryCount = kMaxTrackRetries; 2385 // If one track is ready, set the mixer ready if: 2386 // - the mixer was not ready during previous round OR 2387 // - no other track is not ready 2388 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2389 mixerStatus != MIXER_TRACKS_ENABLED) { 2390 mixerStatus = MIXER_TRACKS_READY; 2391 } 2392 } else { 2393 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2394 if (track->isStopped()) { 2395 track->reset(); 2396 } 2397 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2398 // We have consumed all the buffers of this track. 2399 // Remove it from the list of active tracks. 2400 tracksToRemove->add(track); 2401 } else { 2402 // No buffers for this track. Give it a few chances to 2403 // fill a buffer, then remove it from active list. 2404 if (--(track->mRetryCount) <= 0) { 2405 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2406 tracksToRemove->add(track); 2407 // indicate to client process that the track was disabled because of underrun 2408 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2409 // If one track is not ready, mark the mixer also not ready if: 2410 // - the mixer was ready during previous round OR 2411 // - no other track is ready 2412 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2413 mixerStatus != MIXER_TRACKS_READY) { 2414 mixerStatus = MIXER_TRACKS_ENABLED; 2415 } 2416 } 2417 mAudioMixer->disable(name); 2418 } 2419 } 2420 2421 // remove all the tracks that need to be... 2422 count = tracksToRemove->size(); 2423 if (CC_UNLIKELY(count)) { 2424 for (size_t i=0 ; i<count ; i++) { 2425 const sp<Track>& track = tracksToRemove->itemAt(i); 2426 mActiveTracks.remove(track); 2427 if (track->mainBuffer() != mMixBuffer) { 2428 chain = getEffectChain_l(track->sessionId()); 2429 if (chain != 0) { 2430 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2431 chain->decActiveTrackCnt(); 2432 } 2433 } 2434 if (track->isTerminated()) { 2435 removeTrack_l(track); 2436 } 2437 } 2438 } 2439 2440 // mix buffer must be cleared if all tracks are connected to an 2441 // effect chain as in this case the mixer will not write to 2442 // mix buffer and track effects will accumulate into it 2443 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2444 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2445 } 2446 2447 mPrevMixerStatus = mixerStatus; 2448 return mixerStatus; 2449} 2450 2451void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2452{ 2453 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2454 this, streamType, mTracks.size()); 2455 Mutex::Autolock _l(mLock); 2456 2457 size_t size = mTracks.size(); 2458 for (size_t i = 0; i < size; i++) { 2459 sp<Track> t = mTracks[i]; 2460 if (t->streamType() == streamType) { 2461 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2462 t->mCblk->cv.signal(); 2463 } 2464 } 2465} 2466 2467void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2468{ 2469 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2470 this, streamType, valid); 2471 Mutex::Autolock _l(mLock); 2472 2473 mStreamTypes[streamType].valid = valid; 2474} 2475 2476// getTrackName_l() must be called with ThreadBase::mLock held 2477int AudioFlinger::MixerThread::getTrackName_l() 2478{ 2479 return mAudioMixer->getTrackName(); 2480} 2481 2482// deleteTrackName_l() must be called with ThreadBase::mLock held 2483void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2484{ 2485 ALOGV("remove track (%d) and delete from mixer", name); 2486 mAudioMixer->deleteTrackName(name); 2487} 2488 2489// checkForNewParameters_l() must be called with ThreadBase::mLock held 2490bool AudioFlinger::MixerThread::checkForNewParameters_l() 2491{ 2492 bool reconfig = false; 2493 2494 while (!mNewParameters.isEmpty()) { 2495 status_t status = NO_ERROR; 2496 String8 keyValuePair = mNewParameters[0]; 2497 AudioParameter param = AudioParameter(keyValuePair); 2498 int value; 2499 2500 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2501 reconfig = true; 2502 } 2503 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2504 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2505 status = BAD_VALUE; 2506 } else { 2507 reconfig = true; 2508 } 2509 } 2510 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2511 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2512 status = BAD_VALUE; 2513 } else { 2514 reconfig = true; 2515 } 2516 } 2517 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2518 // do not accept frame count changes if tracks are open as the track buffer 2519 // size depends on frame count and correct behavior would not be guaranteed 2520 // if frame count is changed after track creation 2521 if (!mTracks.isEmpty()) { 2522 status = INVALID_OPERATION; 2523 } else { 2524 reconfig = true; 2525 } 2526 } 2527 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2528 // when changing the audio output device, call addBatteryData to notify 2529 // the change 2530 if ((int)mDevice != value) { 2531 uint32_t params = 0; 2532 // check whether speaker is on 2533 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2534 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2535 } 2536 2537 int deviceWithoutSpeaker 2538 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2539 // check if any other device (except speaker) is on 2540 if (value & deviceWithoutSpeaker ) { 2541 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2542 } 2543 2544 if (params != 0) { 2545 addBatteryData(params); 2546 } 2547 } 2548 2549 // forward device change to effects that have requested to be 2550 // aware of attached audio device. 2551 mDevice = (uint32_t)value; 2552 for (size_t i = 0; i < mEffectChains.size(); i++) { 2553 mEffectChains[i]->setDevice_l(mDevice); 2554 } 2555 } 2556 2557 if (status == NO_ERROR) { 2558 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2559 keyValuePair.string()); 2560 if (!mStandby && status == INVALID_OPERATION) { 2561 mOutput->stream->common.standby(&mOutput->stream->common); 2562 mStandby = true; 2563 mBytesWritten = 0; 2564 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2565 keyValuePair.string()); 2566 } 2567 if (status == NO_ERROR && reconfig) { 2568 delete mAudioMixer; 2569 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2570 mAudioMixer = NULL; 2571 readOutputParameters(); 2572 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2573 for (size_t i = 0; i < mTracks.size() ; i++) { 2574 int name = getTrackName_l(); 2575 if (name < 0) break; 2576 mTracks[i]->mName = name; 2577 // limit track sample rate to 2 x new output sample rate 2578 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2579 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2580 } 2581 } 2582 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2583 } 2584 } 2585 2586 mNewParameters.removeAt(0); 2587 2588 mParamStatus = status; 2589 mParamCond.signal(); 2590 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2591 // already timed out waiting for the status and will never signal the condition. 2592 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2593 } 2594 return reconfig; 2595} 2596 2597status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2598{ 2599 const size_t SIZE = 256; 2600 char buffer[SIZE]; 2601 String8 result; 2602 2603 PlaybackThread::dumpInternals(fd, args); 2604 2605 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2606 result.append(buffer); 2607 write(fd, result.string(), result.size()); 2608 return NO_ERROR; 2609} 2610 2611uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2612{ 2613 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2614} 2615 2616uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2617{ 2618 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2619} 2620 2621// ---------------------------------------------------------------------------- 2622AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2623 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2624 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2625 // mLeftVolFloat, mRightVolFloat 2626 // mLeftVolShort, mRightVolShort 2627{ 2628} 2629 2630AudioFlinger::DirectOutputThread::~DirectOutputThread() 2631{ 2632} 2633 2634void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2635{ 2636 // Do not apply volume on compressed audio 2637 if (!audio_is_linear_pcm(mFormat)) { 2638 return; 2639 } 2640 2641 // convert to signed 16 bit before volume calculation 2642 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2643 size_t count = mFrameCount * mChannelCount; 2644 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2645 int16_t *dst = mMixBuffer + count-1; 2646 while(count--) { 2647 *dst-- = (int16_t)(*src--^0x80) << 8; 2648 } 2649 } 2650 2651 size_t frameCount = mFrameCount; 2652 int16_t *out = mMixBuffer; 2653 if (ramp) { 2654 if (mChannelCount == 1) { 2655 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2656 int32_t vlInc = d / (int32_t)frameCount; 2657 int32_t vl = ((int32_t)mLeftVolShort << 16); 2658 do { 2659 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2660 out++; 2661 vl += vlInc; 2662 } while (--frameCount); 2663 2664 } else { 2665 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2666 int32_t vlInc = d / (int32_t)frameCount; 2667 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2668 int32_t vrInc = d / (int32_t)frameCount; 2669 int32_t vl = ((int32_t)mLeftVolShort << 16); 2670 int32_t vr = ((int32_t)mRightVolShort << 16); 2671 do { 2672 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2673 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2674 out += 2; 2675 vl += vlInc; 2676 vr += vrInc; 2677 } while (--frameCount); 2678 } 2679 } else { 2680 if (mChannelCount == 1) { 2681 do { 2682 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2683 out++; 2684 } while (--frameCount); 2685 } else { 2686 do { 2687 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2688 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2689 out += 2; 2690 } while (--frameCount); 2691 } 2692 } 2693 2694 // convert back to unsigned 8 bit after volume calculation 2695 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2696 size_t count = mFrameCount * mChannelCount; 2697 int16_t *src = mMixBuffer; 2698 uint8_t *dst = (uint8_t *)mMixBuffer; 2699 while(count--) { 2700 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2701 } 2702 } 2703 2704 mLeftVolShort = leftVol; 2705 mRightVolShort = rightVol; 2706} 2707 2708bool AudioFlinger::DirectOutputThread::threadLoop() 2709{ 2710 sp<Track> trackToRemove; 2711 sp<Track> activeTrack; 2712 nsecs_t standbyTime = systemTime(); 2713 size_t mixBufferSize = mFrameCount*mFrameSize; 2714 uint32_t activeSleepTime = activeSleepTimeUs(); 2715 uint32_t idleSleepTime = idleSleepTimeUs(); 2716 uint32_t sleepTime = idleSleepTime; 2717 // use shorter standby delay as on normal output to release 2718 // hardware resources as soon as possible 2719 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2720 2721 acquireWakeLock(); 2722 2723 while (!exitPending()) 2724 { 2725 bool rampVolume; 2726 uint16_t leftVol; 2727 uint16_t rightVol; 2728 Vector< sp<EffectChain> > effectChains; 2729 2730 processConfigEvents(); 2731 2732 mixer_state mixerStatus = MIXER_IDLE; 2733 { // scope for the mLock 2734 2735 Mutex::Autolock _l(mLock); 2736 2737 if (checkForNewParameters_l()) { 2738 mixBufferSize = mFrameCount*mFrameSize; 2739 activeSleepTime = activeSleepTimeUs(); 2740 idleSleepTime = idleSleepTimeUs(); 2741 standbyDelay = microseconds(activeSleepTime*2); 2742 } 2743 2744 // put audio hardware into standby after short delay 2745 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2746 mSuspended)) { 2747 // wait until we have something to do... 2748 if (!mStandby) { 2749 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2750 mOutput->stream->common.standby(&mOutput->stream->common); 2751 mStandby = true; 2752 mBytesWritten = 0; 2753 } 2754 2755 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2756 // we're about to wait, flush the binder command buffer 2757 IPCThreadState::self()->flushCommands(); 2758 2759 if (exitPending()) break; 2760 2761 releaseWakeLock_l(); 2762 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2763 mWaitWorkCV.wait(mLock); 2764 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2765 acquireWakeLock_l(); 2766 2767 checkSilentMode_l(); 2768 2769 standbyTime = systemTime() + standbyDelay; 2770 sleepTime = idleSleepTime; 2771 continue; 2772 } 2773 } 2774 2775 effectChains = mEffectChains; 2776 2777 // find out which tracks need to be processed 2778 if (mActiveTracks.size() != 0) { 2779 sp<Track> t = mActiveTracks[0].promote(); 2780 if (t == 0) continue; 2781 2782 Track* const track = t.get(); 2783 audio_track_cblk_t* cblk = track->cblk(); 2784 2785 // The first time a track is added we wait 2786 // for all its buffers to be filled before processing it 2787 if (cblk->framesReady() && track->isReady() && 2788 !track->isPaused() && !track->isTerminated()) 2789 { 2790 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2791 2792 if (track->mFillingUpStatus == Track::FS_FILLED) { 2793 track->mFillingUpStatus = Track::FS_ACTIVE; 2794 mLeftVolFloat = mRightVolFloat = 0; 2795 mLeftVolShort = mRightVolShort = 0; 2796 if (track->mState == TrackBase::RESUMING) { 2797 track->mState = TrackBase::ACTIVE; 2798 rampVolume = true; 2799 } 2800 } else if (cblk->server != 0) { 2801 // If the track is stopped before the first frame was mixed, 2802 // do not apply ramp 2803 rampVolume = true; 2804 } 2805 // compute volume for this track 2806 float left, right; 2807 if (track->isMuted() || mMasterMute || track->isPausing() || 2808 mStreamTypes[track->streamType()].mute) { 2809 left = right = 0; 2810 if (track->isPausing()) { 2811 track->setPaused(); 2812 } 2813 } else { 2814 float typeVolume = mStreamTypes[track->streamType()].volume; 2815 float v = mMasterVolume * typeVolume; 2816 uint32_t vlr = cblk->getVolumeLR(); 2817 float v_clamped = v * (vlr & 0xFFFF); 2818 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2819 left = v_clamped/MAX_GAIN; 2820 v_clamped = v * (vlr >> 16); 2821 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2822 right = v_clamped/MAX_GAIN; 2823 } 2824 2825 if (left != mLeftVolFloat || right != mRightVolFloat) { 2826 mLeftVolFloat = left; 2827 mRightVolFloat = right; 2828 2829 // If audio HAL implements volume control, 2830 // force software volume to nominal value 2831 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2832 left = 1.0f; 2833 right = 1.0f; 2834 } 2835 2836 // Convert volumes from float to 8.24 2837 uint32_t vl = (uint32_t)(left * (1 << 24)); 2838 uint32_t vr = (uint32_t)(right * (1 << 24)); 2839 2840 // Delegate volume control to effect in track effect chain if needed 2841 // only one effect chain can be present on DirectOutputThread, so if 2842 // there is one, the track is connected to it 2843 if (!effectChains.isEmpty()) { 2844 // Do not ramp volume if volume is controlled by effect 2845 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2846 rampVolume = false; 2847 } 2848 } 2849 2850 // Convert volumes from 8.24 to 4.12 format 2851 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2852 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2853 leftVol = (uint16_t)v_clamped; 2854 v_clamped = (vr + (1 << 11)) >> 12; 2855 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2856 rightVol = (uint16_t)v_clamped; 2857 } else { 2858 leftVol = mLeftVolShort; 2859 rightVol = mRightVolShort; 2860 rampVolume = false; 2861 } 2862 2863 // reset retry count 2864 track->mRetryCount = kMaxTrackRetriesDirect; 2865 activeTrack = t; 2866 mixerStatus = MIXER_TRACKS_READY; 2867 } else { 2868 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2869 if (track->isStopped()) { 2870 track->reset(); 2871 } 2872 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2873 // We have consumed all the buffers of this track. 2874 // Remove it from the list of active tracks. 2875 trackToRemove = track; 2876 } else { 2877 // No buffers for this track. Give it a few chances to 2878 // fill a buffer, then remove it from active list. 2879 if (--(track->mRetryCount) <= 0) { 2880 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2881 trackToRemove = track; 2882 } else { 2883 mixerStatus = MIXER_TRACKS_ENABLED; 2884 } 2885 } 2886 } 2887 } 2888 2889 // remove all the tracks that need to be... 2890 if (CC_UNLIKELY(trackToRemove != 0)) { 2891 mActiveTracks.remove(trackToRemove); 2892 if (!effectChains.isEmpty()) { 2893 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2894 trackToRemove->sessionId()); 2895 effectChains[0]->decActiveTrackCnt(); 2896 } 2897 if (trackToRemove->isTerminated()) { 2898 removeTrack_l(trackToRemove); 2899 } 2900 } 2901 2902 lockEffectChains_l(effectChains); 2903 } 2904 2905 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2906 AudioBufferProvider::Buffer buffer; 2907 size_t frameCount = mFrameCount; 2908 int8_t *curBuf = (int8_t *)mMixBuffer; 2909 // output audio to hardware 2910 while (frameCount) { 2911 buffer.frameCount = frameCount; 2912 activeTrack->getNextBuffer(&buffer, 2913 AudioBufferProvider::kInvalidPTS); 2914 if (CC_UNLIKELY(buffer.raw == NULL)) { 2915 memset(curBuf, 0, frameCount * mFrameSize); 2916 break; 2917 } 2918 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2919 frameCount -= buffer.frameCount; 2920 curBuf += buffer.frameCount * mFrameSize; 2921 activeTrack->releaseBuffer(&buffer); 2922 } 2923 sleepTime = 0; 2924 standbyTime = systemTime() + standbyDelay; 2925 } else { 2926 if (sleepTime == 0) { 2927 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2928 sleepTime = activeSleepTime; 2929 } else { 2930 sleepTime = idleSleepTime; 2931 } 2932 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2933 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2934 sleepTime = 0; 2935 } 2936 } 2937 2938 if (mSuspended) { 2939 sleepTime = suspendSleepTimeUs(); 2940 } 2941 // sleepTime == 0 means we must write to audio hardware 2942 if (sleepTime == 0) { 2943 if (mixerStatus == MIXER_TRACKS_READY) { 2944 applyVolume(leftVol, rightVol, rampVolume); 2945 } 2946 for (size_t i = 0; i < effectChains.size(); i ++) { 2947 effectChains[i]->process_l(); 2948 } 2949 unlockEffectChains(effectChains); 2950 2951 mLastWriteTime = systemTime(); 2952 mInWrite = true; 2953 mBytesWritten += mixBufferSize; 2954 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2955 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2956 mNumWrites++; 2957 mInWrite = false; 2958 mStandby = false; 2959 } else { 2960 unlockEffectChains(effectChains); 2961 usleep(sleepTime); 2962 } 2963 2964 // finally let go of removed track, without the lock held 2965 // since we can't guarantee the destructors won't acquire that 2966 // same lock. 2967 trackToRemove.clear(); 2968 activeTrack.clear(); 2969 2970 // Effect chains will be actually deleted here if they were removed from 2971 // mEffectChains list during mixing or effects processing 2972 effectChains.clear(); 2973 } 2974 2975 if (!mStandby) { 2976 mOutput->stream->common.standby(&mOutput->stream->common); 2977 } 2978 2979 releaseWakeLock(); 2980 2981 ALOGV("Thread %p type %d exiting", this, mType); 2982 return false; 2983} 2984 2985// getTrackName_l() must be called with ThreadBase::mLock held 2986int AudioFlinger::DirectOutputThread::getTrackName_l() 2987{ 2988 return 0; 2989} 2990 2991// deleteTrackName_l() must be called with ThreadBase::mLock held 2992void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2993{ 2994} 2995 2996// checkForNewParameters_l() must be called with ThreadBase::mLock held 2997bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2998{ 2999 bool reconfig = false; 3000 3001 while (!mNewParameters.isEmpty()) { 3002 status_t status = NO_ERROR; 3003 String8 keyValuePair = mNewParameters[0]; 3004 AudioParameter param = AudioParameter(keyValuePair); 3005 int value; 3006 3007 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3008 // do not accept frame count changes if tracks are open as the track buffer 3009 // size depends on frame count and correct behavior would not be garantied 3010 // if frame count is changed after track creation 3011 if (!mTracks.isEmpty()) { 3012 status = INVALID_OPERATION; 3013 } else { 3014 reconfig = true; 3015 } 3016 } 3017 if (status == NO_ERROR) { 3018 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3019 keyValuePair.string()); 3020 if (!mStandby && status == INVALID_OPERATION) { 3021 mOutput->stream->common.standby(&mOutput->stream->common); 3022 mStandby = true; 3023 mBytesWritten = 0; 3024 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3025 keyValuePair.string()); 3026 } 3027 if (status == NO_ERROR && reconfig) { 3028 readOutputParameters(); 3029 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3030 } 3031 } 3032 3033 mNewParameters.removeAt(0); 3034 3035 mParamStatus = status; 3036 mParamCond.signal(); 3037 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3038 // already timed out waiting for the status and will never signal the condition. 3039 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3040 } 3041 return reconfig; 3042} 3043 3044uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3045{ 3046 uint32_t time; 3047 if (audio_is_linear_pcm(mFormat)) { 3048 time = PlaybackThread::activeSleepTimeUs(); 3049 } else { 3050 time = 10000; 3051 } 3052 return time; 3053} 3054 3055uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3056{ 3057 uint32_t time; 3058 if (audio_is_linear_pcm(mFormat)) { 3059 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3060 } else { 3061 time = 10000; 3062 } 3063 return time; 3064} 3065 3066uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3067{ 3068 uint32_t time; 3069 if (audio_is_linear_pcm(mFormat)) { 3070 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3071 } else { 3072 time = 10000; 3073 } 3074 return time; 3075} 3076 3077 3078// ---------------------------------------------------------------------------- 3079 3080AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3081 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3082 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3083 mWaitTimeMs(UINT_MAX) 3084{ 3085 addOutputTrack(mainThread); 3086} 3087 3088AudioFlinger::DuplicatingThread::~DuplicatingThread() 3089{ 3090 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3091 mOutputTracks[i]->destroy(); 3092 } 3093} 3094 3095bool AudioFlinger::DuplicatingThread::threadLoop() 3096{ 3097 Vector< sp<Track> > tracksToRemove; 3098 nsecs_t standbyTime = systemTime(); 3099 size_t mixBufferSize = mFrameCount*mFrameSize; 3100 SortedVector< sp<OutputTrack> > outputTracks; 3101 uint32_t writeFrames = 0; 3102 uint32_t activeSleepTime = activeSleepTimeUs(); 3103 uint32_t idleSleepTime = idleSleepTimeUs(); 3104 uint32_t sleepTime = idleSleepTime; 3105 Vector< sp<EffectChain> > effectChains; 3106 3107 acquireWakeLock(); 3108 3109 while (!exitPending()) 3110 { 3111 processConfigEvents(); 3112 3113 mixer_state mixerStatus = MIXER_IDLE; 3114 { // scope for the mLock 3115 3116 Mutex::Autolock _l(mLock); 3117 3118 if (checkForNewParameters_l()) { 3119 mixBufferSize = mFrameCount*mFrameSize; 3120 updateWaitTime(); 3121 activeSleepTime = activeSleepTimeUs(); 3122 idleSleepTime = idleSleepTimeUs(); 3123 } 3124 3125 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3126 3127 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3128 outputTracks.add(mOutputTracks[i]); 3129 } 3130 3131 // put audio hardware into standby after short delay 3132 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3133 mSuspended)) { 3134 if (!mStandby) { 3135 for (size_t i = 0; i < outputTracks.size(); i++) { 3136 outputTracks[i]->stop(); 3137 } 3138 mStandby = true; 3139 mBytesWritten = 0; 3140 } 3141 3142 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3143 // we're about to wait, flush the binder command buffer 3144 IPCThreadState::self()->flushCommands(); 3145 outputTracks.clear(); 3146 3147 if (exitPending()) break; 3148 3149 releaseWakeLock_l(); 3150 // wait until we have something to do... 3151 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 3152 mWaitWorkCV.wait(mLock); 3153 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 3154 acquireWakeLock_l(); 3155 3156 checkSilentMode_l(); 3157 3158 standbyTime = systemTime() + mStandbyTimeInNsecs; 3159 sleepTime = idleSleepTime; 3160 continue; 3161 } 3162 } 3163 3164 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3165 3166 // prevent any changes in effect chain list and in each effect chain 3167 // during mixing and effect process as the audio buffers could be deleted 3168 // or modified if an effect is created or deleted 3169 lockEffectChains_l(effectChains); 3170 } 3171 3172 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3173 // mix buffers... 3174 if (outputsReady(outputTracks)) { 3175 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3176 } else { 3177 memset(mMixBuffer, 0, mixBufferSize); 3178 } 3179 sleepTime = 0; 3180 writeFrames = mFrameCount; 3181 } else { 3182 if (sleepTime == 0) { 3183 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3184 sleepTime = activeSleepTime; 3185 } else { 3186 sleepTime = idleSleepTime; 3187 } 3188 } else if (mBytesWritten != 0) { 3189 // flush remaining overflow buffers in output tracks 3190 for (size_t i = 0; i < outputTracks.size(); i++) { 3191 if (outputTracks[i]->isActive()) { 3192 sleepTime = 0; 3193 writeFrames = 0; 3194 memset(mMixBuffer, 0, mixBufferSize); 3195 break; 3196 } 3197 } 3198 } 3199 } 3200 3201 if (mSuspended) { 3202 sleepTime = suspendSleepTimeUs(); 3203 } 3204 // sleepTime == 0 means we must write to audio hardware 3205 if (sleepTime == 0) { 3206 for (size_t i = 0; i < effectChains.size(); i ++) { 3207 effectChains[i]->process_l(); 3208 } 3209 // enable changes in effect chain 3210 unlockEffectChains(effectChains); 3211 3212 standbyTime = systemTime() + mStandbyTimeInNsecs; 3213 for (size_t i = 0; i < outputTracks.size(); i++) { 3214 outputTracks[i]->write(mMixBuffer, writeFrames); 3215 } 3216 mStandby = false; 3217 mBytesWritten += mixBufferSize; 3218 } else { 3219 // enable changes in effect chain 3220 unlockEffectChains(effectChains); 3221 usleep(sleepTime); 3222 } 3223 3224 // finally let go of all our tracks, without the lock held 3225 // since we can't guarantee the destructors won't acquire that 3226 // same lock. 3227 tracksToRemove.clear(); 3228 outputTracks.clear(); 3229 3230 // Effect chains will be actually deleted here if they were removed from 3231 // mEffectChains list during mixing or effects processing 3232 effectChains.clear(); 3233 } 3234 3235 releaseWakeLock(); 3236 3237 ALOGV("Thread %p type %d exiting", this, mType); 3238 return false; 3239} 3240 3241void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3242{ 3243 // FIXME explain this formula 3244 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3245 OutputTrack *outputTrack = new OutputTrack(thread, 3246 this, 3247 mSampleRate, 3248 mFormat, 3249 mChannelMask, 3250 frameCount); 3251 if (outputTrack->cblk() != NULL) { 3252 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3253 mOutputTracks.add(outputTrack); 3254 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3255 updateWaitTime(); 3256 } 3257} 3258 3259void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3260{ 3261 Mutex::Autolock _l(mLock); 3262 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3263 if (mOutputTracks[i]->thread() == thread) { 3264 mOutputTracks[i]->destroy(); 3265 mOutputTracks.removeAt(i); 3266 updateWaitTime(); 3267 return; 3268 } 3269 } 3270 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3271} 3272 3273void AudioFlinger::DuplicatingThread::updateWaitTime() 3274{ 3275 mWaitTimeMs = UINT_MAX; 3276 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3277 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3278 if (strong != 0) { 3279 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3280 if (waitTimeMs < mWaitTimeMs) { 3281 mWaitTimeMs = waitTimeMs; 3282 } 3283 } 3284 } 3285} 3286 3287 3288bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3289{ 3290 for (size_t i = 0; i < outputTracks.size(); i++) { 3291 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3292 if (thread == 0) { 3293 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3294 return false; 3295 } 3296 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3297 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3298 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3299 return false; 3300 } 3301 } 3302 return true; 3303} 3304 3305uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3306{ 3307 return (mWaitTimeMs * 1000) / 2; 3308} 3309 3310// ---------------------------------------------------------------------------- 3311 3312// TrackBase constructor must be called with AudioFlinger::mLock held 3313AudioFlinger::ThreadBase::TrackBase::TrackBase( 3314 ThreadBase *thread, 3315 const sp<Client>& client, 3316 uint32_t sampleRate, 3317 audio_format_t format, 3318 uint32_t channelMask, 3319 int frameCount, 3320 const sp<IMemory>& sharedBuffer, 3321 int sessionId) 3322 : RefBase(), 3323 mThread(thread), 3324 mClient(client), 3325 mCblk(NULL), 3326 // mBuffer 3327 // mBufferEnd 3328 mFrameCount(0), 3329 mState(IDLE), 3330 mFormat(format), 3331 mStepServerFailed(false), 3332 mSessionId(sessionId) 3333 // mChannelCount 3334 // mChannelMask 3335{ 3336 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3337 3338 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3339 size_t size = sizeof(audio_track_cblk_t); 3340 uint8_t channelCount = popcount(channelMask); 3341 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3342 if (sharedBuffer == 0) { 3343 size += bufferSize; 3344 } 3345 3346 if (client != NULL) { 3347 mCblkMemory = client->heap()->allocate(size); 3348 if (mCblkMemory != 0) { 3349 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3350 if (mCblk != NULL) { // construct the shared structure in-place. 3351 new(mCblk) audio_track_cblk_t(); 3352 // clear all buffers 3353 mCblk->frameCount = frameCount; 3354 mCblk->sampleRate = sampleRate; 3355 mChannelCount = channelCount; 3356 mChannelMask = channelMask; 3357 if (sharedBuffer == 0) { 3358 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3359 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3360 // Force underrun condition to avoid false underrun callback until first data is 3361 // written to buffer (other flags are cleared) 3362 mCblk->flags = CBLK_UNDERRUN_ON; 3363 } else { 3364 mBuffer = sharedBuffer->pointer(); 3365 } 3366 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3367 } 3368 } else { 3369 ALOGE("not enough memory for AudioTrack size=%u", size); 3370 client->heap()->dump("AudioTrack"); 3371 return; 3372 } 3373 } else { 3374 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3375 // construct the shared structure in-place. 3376 new(mCblk) audio_track_cblk_t(); 3377 // clear all buffers 3378 mCblk->frameCount = frameCount; 3379 mCblk->sampleRate = sampleRate; 3380 mChannelCount = channelCount; 3381 mChannelMask = channelMask; 3382 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3383 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3384 // Force underrun condition to avoid false underrun callback until first data is 3385 // written to buffer (other flags are cleared) 3386 mCblk->flags = CBLK_UNDERRUN_ON; 3387 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3388 } 3389} 3390 3391AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3392{ 3393 if (mCblk != NULL) { 3394 if (mClient == 0) { 3395 delete mCblk; 3396 } else { 3397 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3398 } 3399 } 3400 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3401 if (mClient != 0) { 3402 // Client destructor must run with AudioFlinger mutex locked 3403 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3404 // If the client's reference count drops to zero, the associated destructor 3405 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3406 // relying on the automatic clear() at end of scope. 3407 mClient.clear(); 3408 } 3409} 3410 3411void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3412{ 3413 buffer->raw = NULL; 3414 mFrameCount = buffer->frameCount; 3415 step(); 3416 buffer->frameCount = 0; 3417} 3418 3419bool AudioFlinger::ThreadBase::TrackBase::step() { 3420 bool result; 3421 audio_track_cblk_t* cblk = this->cblk(); 3422 3423 result = cblk->stepServer(mFrameCount); 3424 if (!result) { 3425 ALOGV("stepServer failed acquiring cblk mutex"); 3426 mStepServerFailed = true; 3427 } 3428 return result; 3429} 3430 3431void AudioFlinger::ThreadBase::TrackBase::reset() { 3432 audio_track_cblk_t* cblk = this->cblk(); 3433 3434 cblk->user = 0; 3435 cblk->server = 0; 3436 cblk->userBase = 0; 3437 cblk->serverBase = 0; 3438 mStepServerFailed = false; 3439 ALOGV("TrackBase::reset"); 3440} 3441 3442int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3443 return (int)mCblk->sampleRate; 3444} 3445 3446void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3447 audio_track_cblk_t* cblk = this->cblk(); 3448 size_t frameSize = cblk->frameSize; 3449 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3450 int8_t *bufferEnd = bufferStart + frames * frameSize; 3451 3452 // Check validity of returned pointer in case the track control block would have been corrupted. 3453 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3454 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3455 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3456 server %d, serverBase %d, user %d, userBase %d", 3457 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3458 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3459 return NULL; 3460 } 3461 3462 return bufferStart; 3463} 3464 3465// ---------------------------------------------------------------------------- 3466 3467// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3468AudioFlinger::PlaybackThread::Track::Track( 3469 PlaybackThread *thread, 3470 const sp<Client>& client, 3471 audio_stream_type_t streamType, 3472 uint32_t sampleRate, 3473 audio_format_t format, 3474 uint32_t channelMask, 3475 int frameCount, 3476 const sp<IMemory>& sharedBuffer, 3477 int sessionId) 3478 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3479 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3480 mAuxEffectId(0), mHasVolumeController(false) 3481{ 3482 if (mCblk != NULL) { 3483 if (thread != NULL) { 3484 mName = thread->getTrackName_l(); 3485 mMainBuffer = thread->mixBuffer(); 3486 } 3487 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3488 if (mName < 0) { 3489 ALOGE("no more track names available"); 3490 } 3491 mStreamType = streamType; 3492 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3493 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3494 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3495 } 3496} 3497 3498AudioFlinger::PlaybackThread::Track::~Track() 3499{ 3500 ALOGV("PlaybackThread::Track destructor"); 3501 sp<ThreadBase> thread = mThread.promote(); 3502 if (thread != 0) { 3503 Mutex::Autolock _l(thread->mLock); 3504 mState = TERMINATED; 3505 } 3506} 3507 3508void AudioFlinger::PlaybackThread::Track::destroy() 3509{ 3510 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3511 // by removing it from mTracks vector, so there is a risk that this Tracks's 3512 // destructor is called. As the destructor needs to lock mLock, 3513 // we must acquire a strong reference on this Track before locking mLock 3514 // here so that the destructor is called only when exiting this function. 3515 // On the other hand, as long as Track::destroy() is only called by 3516 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3517 // this Track with its member mTrack. 3518 sp<Track> keep(this); 3519 { // scope for mLock 3520 sp<ThreadBase> thread = mThread.promote(); 3521 if (thread != 0) { 3522 if (!isOutputTrack()) { 3523 if (mState == ACTIVE || mState == RESUMING) { 3524 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3525 3526 // to track the speaker usage 3527 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3528 } 3529 AudioSystem::releaseOutput(thread->id()); 3530 } 3531 Mutex::Autolock _l(thread->mLock); 3532 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3533 playbackThread->destroyTrack_l(this); 3534 } 3535 } 3536} 3537 3538void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3539{ 3540 uint32_t vlr = mCblk->getVolumeLR(); 3541 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", 3542 mName - AudioMixer::TRACK0, 3543 (mClient == 0) ? getpid_cached : mClient->pid(), 3544 mStreamType, 3545 mFormat, 3546 mChannelMask, 3547 mSessionId, 3548 mFrameCount, 3549 mState, 3550 mMute, 3551 mFillingUpStatus, 3552 mCblk->sampleRate, 3553 vlr & 0xFFFF, 3554 vlr >> 16, 3555 mCblk->server, 3556 mCblk->user, 3557 (int)mMainBuffer, 3558 (int)mAuxBuffer); 3559} 3560 3561status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3562 AudioBufferProvider::Buffer* buffer, int64_t pts) 3563{ 3564 audio_track_cblk_t* cblk = this->cblk(); 3565 uint32_t framesReady; 3566 uint32_t framesReq = buffer->frameCount; 3567 3568 // Check if last stepServer failed, try to step now 3569 if (mStepServerFailed) { 3570 if (!step()) goto getNextBuffer_exit; 3571 ALOGV("stepServer recovered"); 3572 mStepServerFailed = false; 3573 } 3574 3575 framesReady = cblk->framesReady(); 3576 3577 if (CC_LIKELY(framesReady)) { 3578 uint32_t s = cblk->server; 3579 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3580 3581 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3582 if (framesReq > framesReady) { 3583 framesReq = framesReady; 3584 } 3585 if (s + framesReq > bufferEnd) { 3586 framesReq = bufferEnd - s; 3587 } 3588 3589 buffer->raw = getBuffer(s, framesReq); 3590 if (buffer->raw == NULL) goto getNextBuffer_exit; 3591 3592 buffer->frameCount = framesReq; 3593 return NO_ERROR; 3594 } 3595 3596getNextBuffer_exit: 3597 buffer->raw = NULL; 3598 buffer->frameCount = 0; 3599 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3600 return NOT_ENOUGH_DATA; 3601} 3602 3603uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3604 return mCblk->framesReady(); 3605} 3606 3607bool AudioFlinger::PlaybackThread::Track::isReady() const { 3608 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3609 3610 if (framesReady() >= mCblk->frameCount || 3611 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3612 mFillingUpStatus = FS_FILLED; 3613 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3614 return true; 3615 } 3616 return false; 3617} 3618 3619status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3620{ 3621 status_t status = NO_ERROR; 3622 ALOGV("start(%d), calling pid %d session %d tid %d", 3623 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3624 sp<ThreadBase> thread = mThread.promote(); 3625 if (thread != 0) { 3626 Mutex::Autolock _l(thread->mLock); 3627 track_state state = mState; 3628 // here the track could be either new, or restarted 3629 // in both cases "unstop" the track 3630 if (mState == PAUSED) { 3631 mState = TrackBase::RESUMING; 3632 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3633 } else { 3634 mState = TrackBase::ACTIVE; 3635 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3636 } 3637 3638 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3639 thread->mLock.unlock(); 3640 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3641 thread->mLock.lock(); 3642 3643 // to track the speaker usage 3644 if (status == NO_ERROR) { 3645 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3646 } 3647 } 3648 if (status == NO_ERROR) { 3649 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3650 playbackThread->addTrack_l(this); 3651 } else { 3652 mState = state; 3653 } 3654 } else { 3655 status = BAD_VALUE; 3656 } 3657 return status; 3658} 3659 3660void AudioFlinger::PlaybackThread::Track::stop() 3661{ 3662 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3663 sp<ThreadBase> thread = mThread.promote(); 3664 if (thread != 0) { 3665 Mutex::Autolock _l(thread->mLock); 3666 track_state state = mState; 3667 if (mState > STOPPED) { 3668 mState = STOPPED; 3669 // If the track is not active (PAUSED and buffers full), flush buffers 3670 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3671 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3672 reset(); 3673 } 3674 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3675 } 3676 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3677 thread->mLock.unlock(); 3678 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3679 thread->mLock.lock(); 3680 3681 // to track the speaker usage 3682 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3683 } 3684 } 3685} 3686 3687void AudioFlinger::PlaybackThread::Track::pause() 3688{ 3689 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3690 sp<ThreadBase> thread = mThread.promote(); 3691 if (thread != 0) { 3692 Mutex::Autolock _l(thread->mLock); 3693 if (mState == ACTIVE || mState == RESUMING) { 3694 mState = PAUSING; 3695 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3696 if (!isOutputTrack()) { 3697 thread->mLock.unlock(); 3698 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3699 thread->mLock.lock(); 3700 3701 // to track the speaker usage 3702 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3703 } 3704 } 3705 } 3706} 3707 3708void AudioFlinger::PlaybackThread::Track::flush() 3709{ 3710 ALOGV("flush(%d)", mName); 3711 sp<ThreadBase> thread = mThread.promote(); 3712 if (thread != 0) { 3713 Mutex::Autolock _l(thread->mLock); 3714 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3715 return; 3716 } 3717 // No point remaining in PAUSED state after a flush => go to 3718 // STOPPED state 3719 mState = STOPPED; 3720 3721 // do not reset the track if it is still in the process of being stopped or paused. 3722 // this will be done by prepareTracks_l() when the track is stopped. 3723 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3724 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3725 reset(); 3726 } 3727 } 3728} 3729 3730void AudioFlinger::PlaybackThread::Track::reset() 3731{ 3732 // Do not reset twice to avoid discarding data written just after a flush and before 3733 // the audioflinger thread detects the track is stopped. 3734 if (!mResetDone) { 3735 TrackBase::reset(); 3736 // Force underrun condition to avoid false underrun callback until first data is 3737 // written to buffer 3738 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3739 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3740 mFillingUpStatus = FS_FILLING; 3741 mResetDone = true; 3742 } 3743} 3744 3745void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3746{ 3747 mMute = muted; 3748} 3749 3750status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3751{ 3752 status_t status = DEAD_OBJECT; 3753 sp<ThreadBase> thread = mThread.promote(); 3754 if (thread != 0) { 3755 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3756 status = playbackThread->attachAuxEffect(this, EffectId); 3757 } 3758 return status; 3759} 3760 3761void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3762{ 3763 mAuxEffectId = EffectId; 3764 mAuxBuffer = buffer; 3765} 3766 3767// timed audio tracks 3768 3769sp<AudioFlinger::PlaybackThread::TimedTrack> 3770AudioFlinger::PlaybackThread::TimedTrack::create( 3771 PlaybackThread *thread, 3772 const sp<Client>& client, 3773 audio_stream_type_t streamType, 3774 uint32_t sampleRate, 3775 audio_format_t format, 3776 uint32_t channelMask, 3777 int frameCount, 3778 const sp<IMemory>& sharedBuffer, 3779 int sessionId) { 3780 if (!client->reserveTimedTrack()) 3781 return NULL; 3782 3783 sp<TimedTrack> track = new TimedTrack( 3784 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3785 sharedBuffer, sessionId); 3786 3787 if (track == NULL) { 3788 client->releaseTimedTrack(); 3789 return NULL; 3790 } 3791 3792 return track; 3793} 3794 3795AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3796 PlaybackThread *thread, 3797 const sp<Client>& client, 3798 audio_stream_type_t streamType, 3799 uint32_t sampleRate, 3800 audio_format_t format, 3801 uint32_t channelMask, 3802 int frameCount, 3803 const sp<IMemory>& sharedBuffer, 3804 int sessionId) 3805 : Track(thread, client, streamType, sampleRate, format, channelMask, 3806 frameCount, sharedBuffer, sessionId), 3807 mTimedSilenceBuffer(NULL), 3808 mTimedSilenceBufferSize(0), 3809 mTimedAudioOutputOnTime(false), 3810 mMediaTimeTransformValid(false) 3811{ 3812 LocalClock lc; 3813 mLocalTimeFreq = lc.getLocalFreq(); 3814 3815 mLocalTimeToSampleTransform.a_zero = 0; 3816 mLocalTimeToSampleTransform.b_zero = 0; 3817 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3818 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3819 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3820 &mLocalTimeToSampleTransform.a_to_b_denom); 3821} 3822 3823AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3824 mClient->releaseTimedTrack(); 3825 delete [] mTimedSilenceBuffer; 3826} 3827 3828status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3829 size_t size, sp<IMemory>* buffer) { 3830 3831 Mutex::Autolock _l(mTimedBufferQueueLock); 3832 3833 trimTimedBufferQueue_l(); 3834 3835 // lazily initialize the shared memory heap for timed buffers 3836 if (mTimedMemoryDealer == NULL) { 3837 const int kTimedBufferHeapSize = 512 << 10; 3838 3839 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3840 "AudioFlingerTimed"); 3841 if (mTimedMemoryDealer == NULL) 3842 return NO_MEMORY; 3843 } 3844 3845 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3846 if (newBuffer == NULL) { 3847 newBuffer = mTimedMemoryDealer->allocate(size); 3848 if (newBuffer == NULL) 3849 return NO_MEMORY; 3850 } 3851 3852 *buffer = newBuffer; 3853 return NO_ERROR; 3854} 3855 3856// caller must hold mTimedBufferQueueLock 3857void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3858 int64_t mediaTimeNow; 3859 { 3860 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3861 if (!mMediaTimeTransformValid) 3862 return; 3863 3864 int64_t targetTimeNow; 3865 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3866 ? mCCHelper.getCommonTime(&targetTimeNow) 3867 : mCCHelper.getLocalTime(&targetTimeNow); 3868 3869 if (OK != res) 3870 return; 3871 3872 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3873 &mediaTimeNow)) { 3874 return; 3875 } 3876 } 3877 3878 size_t trimIndex; 3879 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3880 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3881 break; 3882 } 3883 3884 if (trimIndex) { 3885 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3886 } 3887} 3888 3889status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3890 const sp<IMemory>& buffer, int64_t pts) { 3891 3892 { 3893 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3894 if (!mMediaTimeTransformValid) 3895 return INVALID_OPERATION; 3896 } 3897 3898 Mutex::Autolock _l(mTimedBufferQueueLock); 3899 3900 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3901 3902 return NO_ERROR; 3903} 3904 3905status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3906 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3907 3908 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3909 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3910 target); 3911 3912 if (!(target == TimedAudioTrack::LOCAL_TIME || 3913 target == TimedAudioTrack::COMMON_TIME)) { 3914 return BAD_VALUE; 3915 } 3916 3917 Mutex::Autolock lock(mMediaTimeTransformLock); 3918 mMediaTimeTransform = xform; 3919 mMediaTimeTransformTarget = target; 3920 mMediaTimeTransformValid = true; 3921 3922 return NO_ERROR; 3923} 3924 3925#define min(a, b) ((a) < (b) ? (a) : (b)) 3926 3927// implementation of getNextBuffer for tracks whose buffers have timestamps 3928status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3929 AudioBufferProvider::Buffer* buffer, int64_t pts) 3930{ 3931 if (pts == AudioBufferProvider::kInvalidPTS) { 3932 buffer->raw = 0; 3933 buffer->frameCount = 0; 3934 return INVALID_OPERATION; 3935 } 3936 3937 Mutex::Autolock _l(mTimedBufferQueueLock); 3938 3939 while (true) { 3940 3941 // if we have no timed buffers, then fail 3942 if (mTimedBufferQueue.isEmpty()) { 3943 buffer->raw = 0; 3944 buffer->frameCount = 0; 3945 return NOT_ENOUGH_DATA; 3946 } 3947 3948 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3949 3950 // calculate the PTS of the head of the timed buffer queue expressed in 3951 // local time 3952 int64_t headLocalPTS; 3953 { 3954 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3955 3956 assert(mMediaTimeTransformValid); 3957 3958 if (mMediaTimeTransform.a_to_b_denom == 0) { 3959 // the transform represents a pause, so yield silence 3960 timedYieldSilence(buffer->frameCount, buffer); 3961 return NO_ERROR; 3962 } 3963 3964 int64_t transformedPTS; 3965 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3966 &transformedPTS)) { 3967 // the transform failed. this shouldn't happen, but if it does 3968 // then just drop this buffer 3969 ALOGW("timedGetNextBuffer transform failed"); 3970 buffer->raw = 0; 3971 buffer->frameCount = 0; 3972 mTimedBufferQueue.removeAt(0); 3973 return NO_ERROR; 3974 } 3975 3976 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3977 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3978 &headLocalPTS)) { 3979 buffer->raw = 0; 3980 buffer->frameCount = 0; 3981 return INVALID_OPERATION; 3982 } 3983 } else { 3984 headLocalPTS = transformedPTS; 3985 } 3986 } 3987 3988 // adjust the head buffer's PTS to reflect the portion of the head buffer 3989 // that has already been consumed 3990 int64_t effectivePTS = headLocalPTS + 3991 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3992 3993 // Calculate the delta in samples between the head of the input buffer 3994 // queue and the start of the next output buffer that will be written. 3995 // If the transformation fails because of over or underflow, it means 3996 // that the sample's position in the output stream is so far out of 3997 // whack that it should just be dropped. 3998 int64_t sampleDelta; 3999 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4000 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4001 mTimedBufferQueue.removeAt(0); 4002 continue; 4003 } 4004 if (!mLocalTimeToSampleTransform.doForwardTransform( 4005 (effectivePTS - pts) << 32, &sampleDelta)) { 4006 ALOGV("*** too late during sample rate transform: dropped buffer"); 4007 mTimedBufferQueue.removeAt(0); 4008 continue; 4009 } 4010 4011 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4012 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4013 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4014 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4015 4016 // if the delta between the ideal placement for the next input sample and 4017 // the current output position is within this threshold, then we will 4018 // concatenate the next input samples to the previous output 4019 const int64_t kSampleContinuityThreshold = 4020 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4021 4022 // if this is the first buffer of audio that we're emitting from this track 4023 // then it should be almost exactly on time. 4024 const int64_t kSampleStartupThreshold = 1LL << 32; 4025 4026 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4027 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4028 // the next input is close enough to being on time, so concatenate it 4029 // with the last output 4030 timedYieldSamples(buffer); 4031 4032 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4033 return NO_ERROR; 4034 } else if (sampleDelta > 0) { 4035 // the gap between the current output position and the proper start of 4036 // the next input sample is too big, so fill it with silence 4037 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4038 4039 timedYieldSilence(framesUntilNextInput, buffer); 4040 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4041 return NO_ERROR; 4042 } else { 4043 // the next input sample is late 4044 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4045 size_t onTimeSamplePosition = 4046 head.position() + lateFrames * mCblk->frameSize; 4047 4048 if (onTimeSamplePosition > head.buffer()->size()) { 4049 // all the remaining samples in the head are too late, so 4050 // drop it and move on 4051 ALOGV("*** too late: dropped buffer"); 4052 mTimedBufferQueue.removeAt(0); 4053 continue; 4054 } else { 4055 // skip over the late samples 4056 head.setPosition(onTimeSamplePosition); 4057 4058 // yield the available samples 4059 timedYieldSamples(buffer); 4060 4061 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4062 return NO_ERROR; 4063 } 4064 } 4065 } 4066} 4067 4068// Yield samples from the timed buffer queue head up to the given output 4069// buffer's capacity. 4070// 4071// Caller must hold mTimedBufferQueueLock 4072void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4073 AudioBufferProvider::Buffer* buffer) { 4074 4075 const TimedBuffer& head = mTimedBufferQueue[0]; 4076 4077 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4078 head.position()); 4079 4080 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4081 mCblk->frameSize); 4082 size_t framesRequested = buffer->frameCount; 4083 buffer->frameCount = min(framesLeftInHead, framesRequested); 4084 4085 mTimedAudioOutputOnTime = true; 4086} 4087 4088// Yield samples of silence up to the given output buffer's capacity 4089// 4090// Caller must hold mTimedBufferQueueLock 4091void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4092 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4093 4094 // lazily allocate a buffer filled with silence 4095 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4096 delete [] mTimedSilenceBuffer; 4097 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4098 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4099 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4100 } 4101 4102 buffer->raw = mTimedSilenceBuffer; 4103 size_t framesRequested = buffer->frameCount; 4104 buffer->frameCount = min(numFrames, framesRequested); 4105 4106 mTimedAudioOutputOnTime = false; 4107} 4108 4109void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4110 AudioBufferProvider::Buffer* buffer) { 4111 4112 Mutex::Autolock _l(mTimedBufferQueueLock); 4113 4114 // If the buffer which was just released is part of the buffer at the head 4115 // of the queue, be sure to update the amt of the buffer which has been 4116 // consumed. If the buffer being returned is not part of the head of the 4117 // queue, its either because the buffer is part of the silence buffer, or 4118 // because the head of the timed queue was trimmed after the mixer called 4119 // getNextBuffer but before the mixer called releaseBuffer. 4120 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4121 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4122 4123 void* start = head.buffer()->pointer(); 4124 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4125 4126 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4127 head.setPosition(head.position() + 4128 (buffer->frameCount * mCblk->frameSize)); 4129 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4130 mTimedBufferQueue.removeAt(0); 4131 } 4132 } 4133 } 4134 4135 buffer->raw = 0; 4136 buffer->frameCount = 0; 4137} 4138 4139uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4140 Mutex::Autolock _l(mTimedBufferQueueLock); 4141 4142 uint32_t frames = 0; 4143 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4144 const TimedBuffer& tb = mTimedBufferQueue[i]; 4145 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4146 } 4147 4148 return frames; 4149} 4150 4151AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4152 : mPTS(0), mPosition(0) {} 4153 4154AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4155 const sp<IMemory>& buffer, int64_t pts) 4156 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4157 4158// ---------------------------------------------------------------------------- 4159 4160// RecordTrack constructor must be called with AudioFlinger::mLock held 4161AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4162 RecordThread *thread, 4163 const sp<Client>& client, 4164 uint32_t sampleRate, 4165 audio_format_t format, 4166 uint32_t channelMask, 4167 int frameCount, 4168 int sessionId) 4169 : TrackBase(thread, client, sampleRate, format, 4170 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4171 mOverflow(false) 4172{ 4173 if (mCblk != NULL) { 4174 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4175 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4176 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4177 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4178 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4179 } else { 4180 mCblk->frameSize = sizeof(int8_t); 4181 } 4182 } 4183} 4184 4185AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4186{ 4187 sp<ThreadBase> thread = mThread.promote(); 4188 if (thread != 0) { 4189 AudioSystem::releaseInput(thread->id()); 4190 } 4191} 4192 4193status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4194{ 4195 audio_track_cblk_t* cblk = this->cblk(); 4196 uint32_t framesAvail; 4197 uint32_t framesReq = buffer->frameCount; 4198 4199 // Check if last stepServer failed, try to step now 4200 if (mStepServerFailed) { 4201 if (!step()) goto getNextBuffer_exit; 4202 ALOGV("stepServer recovered"); 4203 mStepServerFailed = false; 4204 } 4205 4206 framesAvail = cblk->framesAvailable_l(); 4207 4208 if (CC_LIKELY(framesAvail)) { 4209 uint32_t s = cblk->server; 4210 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4211 4212 if (framesReq > framesAvail) { 4213 framesReq = framesAvail; 4214 } 4215 if (s + framesReq > bufferEnd) { 4216 framesReq = bufferEnd - s; 4217 } 4218 4219 buffer->raw = getBuffer(s, framesReq); 4220 if (buffer->raw == NULL) goto getNextBuffer_exit; 4221 4222 buffer->frameCount = framesReq; 4223 return NO_ERROR; 4224 } 4225 4226getNextBuffer_exit: 4227 buffer->raw = NULL; 4228 buffer->frameCount = 0; 4229 return NOT_ENOUGH_DATA; 4230} 4231 4232status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4233{ 4234 sp<ThreadBase> thread = mThread.promote(); 4235 if (thread != 0) { 4236 RecordThread *recordThread = (RecordThread *)thread.get(); 4237 return recordThread->start(this, tid); 4238 } else { 4239 return BAD_VALUE; 4240 } 4241} 4242 4243void AudioFlinger::RecordThread::RecordTrack::stop() 4244{ 4245 sp<ThreadBase> thread = mThread.promote(); 4246 if (thread != 0) { 4247 RecordThread *recordThread = (RecordThread *)thread.get(); 4248 recordThread->stop(this); 4249 TrackBase::reset(); 4250 // Force overerrun condition to avoid false overrun callback until first data is 4251 // read from buffer 4252 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4253 } 4254} 4255 4256void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4257{ 4258 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4259 (mClient == 0) ? getpid_cached : mClient->pid(), 4260 mFormat, 4261 mChannelMask, 4262 mSessionId, 4263 mFrameCount, 4264 mState, 4265 mCblk->sampleRate, 4266 mCblk->server, 4267 mCblk->user); 4268} 4269 4270 4271// ---------------------------------------------------------------------------- 4272 4273AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4274 PlaybackThread *playbackThread, 4275 DuplicatingThread *sourceThread, 4276 uint32_t sampleRate, 4277 audio_format_t format, 4278 uint32_t channelMask, 4279 int frameCount) 4280 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4281 mActive(false), mSourceThread(sourceThread) 4282{ 4283 4284 if (mCblk != NULL) { 4285 mCblk->flags |= CBLK_DIRECTION_OUT; 4286 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4287 mOutBuffer.frameCount = 0; 4288 playbackThread->mTracks.add(this); 4289 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4290 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4291 mCblk, mBuffer, mCblk->buffers, 4292 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4293 } else { 4294 ALOGW("Error creating output track on thread %p", playbackThread); 4295 } 4296} 4297 4298AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4299{ 4300 clearBufferQueue(); 4301} 4302 4303status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4304{ 4305 status_t status = Track::start(tid); 4306 if (status != NO_ERROR) { 4307 return status; 4308 } 4309 4310 mActive = true; 4311 mRetryCount = 127; 4312 return status; 4313} 4314 4315void AudioFlinger::PlaybackThread::OutputTrack::stop() 4316{ 4317 Track::stop(); 4318 clearBufferQueue(); 4319 mOutBuffer.frameCount = 0; 4320 mActive = false; 4321} 4322 4323bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4324{ 4325 Buffer *pInBuffer; 4326 Buffer inBuffer; 4327 uint32_t channelCount = mChannelCount; 4328 bool outputBufferFull = false; 4329 inBuffer.frameCount = frames; 4330 inBuffer.i16 = data; 4331 4332 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4333 4334 if (!mActive && frames != 0) { 4335 start(0); 4336 sp<ThreadBase> thread = mThread.promote(); 4337 if (thread != 0) { 4338 MixerThread *mixerThread = (MixerThread *)thread.get(); 4339 if (mCblk->frameCount > frames){ 4340 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4341 uint32_t startFrames = (mCblk->frameCount - frames); 4342 pInBuffer = new Buffer; 4343 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4344 pInBuffer->frameCount = startFrames; 4345 pInBuffer->i16 = pInBuffer->mBuffer; 4346 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4347 mBufferQueue.add(pInBuffer); 4348 } else { 4349 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4350 } 4351 } 4352 } 4353 } 4354 4355 while (waitTimeLeftMs) { 4356 // First write pending buffers, then new data 4357 if (mBufferQueue.size()) { 4358 pInBuffer = mBufferQueue.itemAt(0); 4359 } else { 4360 pInBuffer = &inBuffer; 4361 } 4362 4363 if (pInBuffer->frameCount == 0) { 4364 break; 4365 } 4366 4367 if (mOutBuffer.frameCount == 0) { 4368 mOutBuffer.frameCount = pInBuffer->frameCount; 4369 nsecs_t startTime = systemTime(); 4370 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4371 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4372 outputBufferFull = true; 4373 break; 4374 } 4375 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4376 if (waitTimeLeftMs >= waitTimeMs) { 4377 waitTimeLeftMs -= waitTimeMs; 4378 } else { 4379 waitTimeLeftMs = 0; 4380 } 4381 } 4382 4383 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4384 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4385 mCblk->stepUser(outFrames); 4386 pInBuffer->frameCount -= outFrames; 4387 pInBuffer->i16 += outFrames * channelCount; 4388 mOutBuffer.frameCount -= outFrames; 4389 mOutBuffer.i16 += outFrames * channelCount; 4390 4391 if (pInBuffer->frameCount == 0) { 4392 if (mBufferQueue.size()) { 4393 mBufferQueue.removeAt(0); 4394 delete [] pInBuffer->mBuffer; 4395 delete pInBuffer; 4396 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4397 } else { 4398 break; 4399 } 4400 } 4401 } 4402 4403 // If we could not write all frames, allocate a buffer and queue it for next time. 4404 if (inBuffer.frameCount) { 4405 sp<ThreadBase> thread = mThread.promote(); 4406 if (thread != 0 && !thread->standby()) { 4407 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4408 pInBuffer = new Buffer; 4409 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4410 pInBuffer->frameCount = inBuffer.frameCount; 4411 pInBuffer->i16 = pInBuffer->mBuffer; 4412 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4413 mBufferQueue.add(pInBuffer); 4414 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4415 } else { 4416 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4417 } 4418 } 4419 } 4420 4421 // Calling write() with a 0 length buffer, means that no more data will be written: 4422 // If no more buffers are pending, fill output track buffer to make sure it is started 4423 // by output mixer. 4424 if (frames == 0 && mBufferQueue.size() == 0) { 4425 if (mCblk->user < mCblk->frameCount) { 4426 frames = mCblk->frameCount - mCblk->user; 4427 pInBuffer = new Buffer; 4428 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4429 pInBuffer->frameCount = frames; 4430 pInBuffer->i16 = pInBuffer->mBuffer; 4431 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4432 mBufferQueue.add(pInBuffer); 4433 } else if (mActive) { 4434 stop(); 4435 } 4436 } 4437 4438 return outputBufferFull; 4439} 4440 4441status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4442{ 4443 int active; 4444 status_t result; 4445 audio_track_cblk_t* cblk = mCblk; 4446 uint32_t framesReq = buffer->frameCount; 4447 4448// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4449 buffer->frameCount = 0; 4450 4451 uint32_t framesAvail = cblk->framesAvailable(); 4452 4453 4454 if (framesAvail == 0) { 4455 Mutex::Autolock _l(cblk->lock); 4456 goto start_loop_here; 4457 while (framesAvail == 0) { 4458 active = mActive; 4459 if (CC_UNLIKELY(!active)) { 4460 ALOGV("Not active and NO_MORE_BUFFERS"); 4461 return NO_MORE_BUFFERS; 4462 } 4463 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4464 if (result != NO_ERROR) { 4465 return NO_MORE_BUFFERS; 4466 } 4467 // read the server count again 4468 start_loop_here: 4469 framesAvail = cblk->framesAvailable_l(); 4470 } 4471 } 4472 4473// if (framesAvail < framesReq) { 4474// return NO_MORE_BUFFERS; 4475// } 4476 4477 if (framesReq > framesAvail) { 4478 framesReq = framesAvail; 4479 } 4480 4481 uint32_t u = cblk->user; 4482 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4483 4484 if (u + framesReq > bufferEnd) { 4485 framesReq = bufferEnd - u; 4486 } 4487 4488 buffer->frameCount = framesReq; 4489 buffer->raw = (void *)cblk->buffer(u); 4490 return NO_ERROR; 4491} 4492 4493 4494void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4495{ 4496 size_t size = mBufferQueue.size(); 4497 4498 for (size_t i = 0; i < size; i++) { 4499 Buffer *pBuffer = mBufferQueue.itemAt(i); 4500 delete [] pBuffer->mBuffer; 4501 delete pBuffer; 4502 } 4503 mBufferQueue.clear(); 4504} 4505 4506// ---------------------------------------------------------------------------- 4507 4508AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4509 : RefBase(), 4510 mAudioFlinger(audioFlinger), 4511 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4512 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4513 mPid(pid), 4514 mTimedTrackCount(0) 4515{ 4516 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4517} 4518 4519// Client destructor must be called with AudioFlinger::mLock held 4520AudioFlinger::Client::~Client() 4521{ 4522 mAudioFlinger->removeClient_l(mPid); 4523} 4524 4525sp<MemoryDealer> AudioFlinger::Client::heap() const 4526{ 4527 return mMemoryDealer; 4528} 4529 4530// Reserve one of the limited slots for a timed audio track associated 4531// with this client 4532bool AudioFlinger::Client::reserveTimedTrack() 4533{ 4534 const int kMaxTimedTracksPerClient = 4; 4535 4536 Mutex::Autolock _l(mTimedTrackLock); 4537 4538 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4539 ALOGW("can not create timed track - pid %d has exceeded the limit", 4540 mPid); 4541 return false; 4542 } 4543 4544 mTimedTrackCount++; 4545 return true; 4546} 4547 4548// Release a slot for a timed audio track 4549void AudioFlinger::Client::releaseTimedTrack() 4550{ 4551 Mutex::Autolock _l(mTimedTrackLock); 4552 mTimedTrackCount--; 4553} 4554 4555// ---------------------------------------------------------------------------- 4556 4557AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4558 const sp<IAudioFlingerClient>& client, 4559 pid_t pid) 4560 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4561{ 4562} 4563 4564AudioFlinger::NotificationClient::~NotificationClient() 4565{ 4566} 4567 4568void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4569{ 4570 sp<NotificationClient> keep(this); 4571 mAudioFlinger->removeNotificationClient(mPid); 4572} 4573 4574// ---------------------------------------------------------------------------- 4575 4576AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4577 : BnAudioTrack(), 4578 mTrack(track) 4579{ 4580} 4581 4582AudioFlinger::TrackHandle::~TrackHandle() { 4583 // just stop the track on deletion, associated resources 4584 // will be freed from the main thread once all pending buffers have 4585 // been played. Unless it's not in the active track list, in which 4586 // case we free everything now... 4587 mTrack->destroy(); 4588} 4589 4590sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4591 return mTrack->getCblk(); 4592} 4593 4594status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4595 return mTrack->start(tid); 4596} 4597 4598void AudioFlinger::TrackHandle::stop() { 4599 mTrack->stop(); 4600} 4601 4602void AudioFlinger::TrackHandle::flush() { 4603 mTrack->flush(); 4604} 4605 4606void AudioFlinger::TrackHandle::mute(bool e) { 4607 mTrack->mute(e); 4608} 4609 4610void AudioFlinger::TrackHandle::pause() { 4611 mTrack->pause(); 4612} 4613 4614status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4615{ 4616 return mTrack->attachAuxEffect(EffectId); 4617} 4618 4619status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4620 sp<IMemory>* buffer) { 4621 if (!mTrack->isTimedTrack()) 4622 return INVALID_OPERATION; 4623 4624 PlaybackThread::TimedTrack* tt = 4625 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4626 return tt->allocateTimedBuffer(size, buffer); 4627} 4628 4629status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4630 int64_t pts) { 4631 if (!mTrack->isTimedTrack()) 4632 return INVALID_OPERATION; 4633 4634 PlaybackThread::TimedTrack* tt = 4635 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4636 return tt->queueTimedBuffer(buffer, pts); 4637} 4638 4639status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4640 const LinearTransform& xform, int target) { 4641 4642 if (!mTrack->isTimedTrack()) 4643 return INVALID_OPERATION; 4644 4645 PlaybackThread::TimedTrack* tt = 4646 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4647 return tt->setMediaTimeTransform( 4648 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4649} 4650 4651status_t AudioFlinger::TrackHandle::onTransact( 4652 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4653{ 4654 return BnAudioTrack::onTransact(code, data, reply, flags); 4655} 4656 4657// ---------------------------------------------------------------------------- 4658 4659sp<IAudioRecord> AudioFlinger::openRecord( 4660 pid_t pid, 4661 audio_io_handle_t input, 4662 uint32_t sampleRate, 4663 audio_format_t format, 4664 uint32_t channelMask, 4665 int frameCount, 4666 // FIXME dead, remove from IAudioFlinger 4667 uint32_t flags, 4668 int *sessionId, 4669 status_t *status) 4670{ 4671 sp<RecordThread::RecordTrack> recordTrack; 4672 sp<RecordHandle> recordHandle; 4673 sp<Client> client; 4674 status_t lStatus; 4675 RecordThread *thread; 4676 size_t inFrameCount; 4677 int lSessionId; 4678 4679 // check calling permissions 4680 if (!recordingAllowed()) { 4681 lStatus = PERMISSION_DENIED; 4682 goto Exit; 4683 } 4684 4685 // add client to list 4686 { // scope for mLock 4687 Mutex::Autolock _l(mLock); 4688 thread = checkRecordThread_l(input); 4689 if (thread == NULL) { 4690 lStatus = BAD_VALUE; 4691 goto Exit; 4692 } 4693 4694 client = registerPid_l(pid); 4695 4696 // If no audio session id is provided, create one here 4697 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4698 lSessionId = *sessionId; 4699 } else { 4700 lSessionId = nextUniqueId(); 4701 if (sessionId != NULL) { 4702 *sessionId = lSessionId; 4703 } 4704 } 4705 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4706 recordTrack = thread->createRecordTrack_l(client, 4707 sampleRate, 4708 format, 4709 channelMask, 4710 frameCount, 4711 lSessionId, 4712 &lStatus); 4713 } 4714 if (lStatus != NO_ERROR) { 4715 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4716 // destructor is called by the TrackBase destructor with mLock held 4717 client.clear(); 4718 recordTrack.clear(); 4719 goto Exit; 4720 } 4721 4722 // return to handle to client 4723 recordHandle = new RecordHandle(recordTrack); 4724 lStatus = NO_ERROR; 4725 4726Exit: 4727 if (status) { 4728 *status = lStatus; 4729 } 4730 return recordHandle; 4731} 4732 4733// ---------------------------------------------------------------------------- 4734 4735AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4736 : BnAudioRecord(), 4737 mRecordTrack(recordTrack) 4738{ 4739} 4740 4741AudioFlinger::RecordHandle::~RecordHandle() { 4742 stop(); 4743} 4744 4745sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4746 return mRecordTrack->getCblk(); 4747} 4748 4749status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4750 ALOGV("RecordHandle::start()"); 4751 return mRecordTrack->start(tid); 4752} 4753 4754void AudioFlinger::RecordHandle::stop() { 4755 ALOGV("RecordHandle::stop()"); 4756 mRecordTrack->stop(); 4757} 4758 4759status_t AudioFlinger::RecordHandle::onTransact( 4760 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4761{ 4762 return BnAudioRecord::onTransact(code, data, reply, flags); 4763} 4764 4765// ---------------------------------------------------------------------------- 4766 4767AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4768 AudioStreamIn *input, 4769 uint32_t sampleRate, 4770 uint32_t channels, 4771 audio_io_handle_t id, 4772 uint32_t device) : 4773 ThreadBase(audioFlinger, id, device, RECORD), 4774 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4775 // mRsmpInIndex and mInputBytes set by readInputParameters() 4776 mReqChannelCount(popcount(channels)), 4777 mReqSampleRate(sampleRate) 4778 // mBytesRead is only meaningful while active, and so is cleared in start() 4779 // (but might be better to also clear here for dump?) 4780{ 4781 snprintf(mName, kNameLength, "AudioIn_%d", id); 4782 4783 readInputParameters(); 4784} 4785 4786 4787AudioFlinger::RecordThread::~RecordThread() 4788{ 4789 delete[] mRsmpInBuffer; 4790 delete mResampler; 4791 delete[] mRsmpOutBuffer; 4792} 4793 4794void AudioFlinger::RecordThread::onFirstRef() 4795{ 4796 run(mName, PRIORITY_URGENT_AUDIO); 4797} 4798 4799status_t AudioFlinger::RecordThread::readyToRun() 4800{ 4801 status_t status = initCheck(); 4802 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4803 return status; 4804} 4805 4806bool AudioFlinger::RecordThread::threadLoop() 4807{ 4808 AudioBufferProvider::Buffer buffer; 4809 sp<RecordTrack> activeTrack; 4810 Vector< sp<EffectChain> > effectChains; 4811 4812 nsecs_t lastWarning = 0; 4813 4814 acquireWakeLock(); 4815 4816 // start recording 4817 while (!exitPending()) { 4818 4819 processConfigEvents(); 4820 4821 { // scope for mLock 4822 Mutex::Autolock _l(mLock); 4823 checkForNewParameters_l(); 4824 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4825 if (!mStandby) { 4826 mInput->stream->common.standby(&mInput->stream->common); 4827 mStandby = true; 4828 } 4829 4830 if (exitPending()) break; 4831 4832 releaseWakeLock_l(); 4833 ALOGV("RecordThread: loop stopping"); 4834 // go to sleep 4835 mWaitWorkCV.wait(mLock); 4836 ALOGV("RecordThread: loop starting"); 4837 acquireWakeLock_l(); 4838 continue; 4839 } 4840 if (mActiveTrack != 0) { 4841 if (mActiveTrack->mState == TrackBase::PAUSING) { 4842 if (!mStandby) { 4843 mInput->stream->common.standby(&mInput->stream->common); 4844 mStandby = true; 4845 } 4846 mActiveTrack.clear(); 4847 mStartStopCond.broadcast(); 4848 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4849 if (mReqChannelCount != mActiveTrack->channelCount()) { 4850 mActiveTrack.clear(); 4851 mStartStopCond.broadcast(); 4852 } else if (mBytesRead != 0) { 4853 // record start succeeds only if first read from audio input 4854 // succeeds 4855 if (mBytesRead > 0) { 4856 mActiveTrack->mState = TrackBase::ACTIVE; 4857 } else { 4858 mActiveTrack.clear(); 4859 } 4860 mStartStopCond.broadcast(); 4861 } 4862 mStandby = false; 4863 } 4864 } 4865 lockEffectChains_l(effectChains); 4866 } 4867 4868 if (mActiveTrack != 0) { 4869 if (mActiveTrack->mState != TrackBase::ACTIVE && 4870 mActiveTrack->mState != TrackBase::RESUMING) { 4871 unlockEffectChains(effectChains); 4872 usleep(kRecordThreadSleepUs); 4873 continue; 4874 } 4875 for (size_t i = 0; i < effectChains.size(); i ++) { 4876 effectChains[i]->process_l(); 4877 } 4878 4879 buffer.frameCount = mFrameCount; 4880 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4881 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4882 size_t framesOut = buffer.frameCount; 4883 if (mResampler == NULL) { 4884 // no resampling 4885 while (framesOut) { 4886 size_t framesIn = mFrameCount - mRsmpInIndex; 4887 if (framesIn) { 4888 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4889 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4890 if (framesIn > framesOut) 4891 framesIn = framesOut; 4892 mRsmpInIndex += framesIn; 4893 framesOut -= framesIn; 4894 if ((int)mChannelCount == mReqChannelCount || 4895 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4896 memcpy(dst, src, framesIn * mFrameSize); 4897 } else { 4898 int16_t *src16 = (int16_t *)src; 4899 int16_t *dst16 = (int16_t *)dst; 4900 if (mChannelCount == 1) { 4901 while (framesIn--) { 4902 *dst16++ = *src16; 4903 *dst16++ = *src16++; 4904 } 4905 } else { 4906 while (framesIn--) { 4907 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4908 src16 += 2; 4909 } 4910 } 4911 } 4912 } 4913 if (framesOut && mFrameCount == mRsmpInIndex) { 4914 if (framesOut == mFrameCount && 4915 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4916 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4917 framesOut = 0; 4918 } else { 4919 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4920 mRsmpInIndex = 0; 4921 } 4922 if (mBytesRead < 0) { 4923 ALOGE("Error reading audio input"); 4924 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4925 // Force input into standby so that it tries to 4926 // recover at next read attempt 4927 mInput->stream->common.standby(&mInput->stream->common); 4928 usleep(kRecordThreadSleepUs); 4929 } 4930 mRsmpInIndex = mFrameCount; 4931 framesOut = 0; 4932 buffer.frameCount = 0; 4933 } 4934 } 4935 } 4936 } else { 4937 // resampling 4938 4939 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4940 // alter output frame count as if we were expecting stereo samples 4941 if (mChannelCount == 1 && mReqChannelCount == 1) { 4942 framesOut >>= 1; 4943 } 4944 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4945 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4946 // are 32 bit aligned which should be always true. 4947 if (mChannelCount == 2 && mReqChannelCount == 1) { 4948 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4949 // the resampler always outputs stereo samples: do post stereo to mono conversion 4950 int16_t *src = (int16_t *)mRsmpOutBuffer; 4951 int16_t *dst = buffer.i16; 4952 while (framesOut--) { 4953 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4954 src += 2; 4955 } 4956 } else { 4957 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4958 } 4959 4960 } 4961 mActiveTrack->releaseBuffer(&buffer); 4962 mActiveTrack->overflow(); 4963 } 4964 // client isn't retrieving buffers fast enough 4965 else { 4966 if (!mActiveTrack->setOverflow()) { 4967 nsecs_t now = systemTime(); 4968 if ((now - lastWarning) > kWarningThrottleNs) { 4969 ALOGW("RecordThread: buffer overflow"); 4970 lastWarning = now; 4971 } 4972 } 4973 // Release the processor for a while before asking for a new buffer. 4974 // This will give the application more chance to read from the buffer and 4975 // clear the overflow. 4976 usleep(kRecordThreadSleepUs); 4977 } 4978 } 4979 // enable changes in effect chain 4980 unlockEffectChains(effectChains); 4981 effectChains.clear(); 4982 } 4983 4984 if (!mStandby) { 4985 mInput->stream->common.standby(&mInput->stream->common); 4986 } 4987 mActiveTrack.clear(); 4988 4989 mStartStopCond.broadcast(); 4990 4991 releaseWakeLock(); 4992 4993 ALOGV("RecordThread %p exiting", this); 4994 return false; 4995} 4996 4997 4998sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4999 const sp<AudioFlinger::Client>& client, 5000 uint32_t sampleRate, 5001 audio_format_t format, 5002 int channelMask, 5003 int frameCount, 5004 int sessionId, 5005 status_t *status) 5006{ 5007 sp<RecordTrack> track; 5008 status_t lStatus; 5009 5010 lStatus = initCheck(); 5011 if (lStatus != NO_ERROR) { 5012 ALOGE("Audio driver not initialized."); 5013 goto Exit; 5014 } 5015 5016 { // scope for mLock 5017 Mutex::Autolock _l(mLock); 5018 5019 track = new RecordTrack(this, client, sampleRate, 5020 format, channelMask, frameCount, sessionId); 5021 5022 if (track->getCblk() == 0) { 5023 lStatus = NO_MEMORY; 5024 goto Exit; 5025 } 5026 5027 mTrack = track.get(); 5028 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5029 bool suspend = audio_is_bluetooth_sco_device( 5030 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5031 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5032 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5033 } 5034 lStatus = NO_ERROR; 5035 5036Exit: 5037 if (status) { 5038 *status = lStatus; 5039 } 5040 return track; 5041} 5042 5043status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5044{ 5045 ALOGV("RecordThread::start tid=%d", tid); 5046 sp <ThreadBase> strongMe = this; 5047 status_t status = NO_ERROR; 5048 { 5049 AutoMutex lock(mLock); 5050 if (mActiveTrack != 0) { 5051 if (recordTrack != mActiveTrack.get()) { 5052 status = -EBUSY; 5053 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5054 mActiveTrack->mState = TrackBase::ACTIVE; 5055 } 5056 return status; 5057 } 5058 5059 recordTrack->mState = TrackBase::IDLE; 5060 mActiveTrack = recordTrack; 5061 mLock.unlock(); 5062 status_t status = AudioSystem::startInput(mId); 5063 mLock.lock(); 5064 if (status != NO_ERROR) { 5065 mActiveTrack.clear(); 5066 return status; 5067 } 5068 mRsmpInIndex = mFrameCount; 5069 mBytesRead = 0; 5070 if (mResampler != NULL) { 5071 mResampler->reset(); 5072 } 5073 mActiveTrack->mState = TrackBase::RESUMING; 5074 // signal thread to start 5075 ALOGV("Signal record thread"); 5076 mWaitWorkCV.signal(); 5077 // do not wait for mStartStopCond if exiting 5078 if (exitPending()) { 5079 mActiveTrack.clear(); 5080 status = INVALID_OPERATION; 5081 goto startError; 5082 } 5083 mStartStopCond.wait(mLock); 5084 if (mActiveTrack == 0) { 5085 ALOGV("Record failed to start"); 5086 status = BAD_VALUE; 5087 goto startError; 5088 } 5089 ALOGV("Record started OK"); 5090 return status; 5091 } 5092startError: 5093 AudioSystem::stopInput(mId); 5094 return status; 5095} 5096 5097void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5098 ALOGV("RecordThread::stop"); 5099 sp <ThreadBase> strongMe = this; 5100 { 5101 AutoMutex lock(mLock); 5102 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5103 mActiveTrack->mState = TrackBase::PAUSING; 5104 // do not wait for mStartStopCond if exiting 5105 if (exitPending()) { 5106 return; 5107 } 5108 mStartStopCond.wait(mLock); 5109 // if we have been restarted, recordTrack == mActiveTrack.get() here 5110 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5111 mLock.unlock(); 5112 AudioSystem::stopInput(mId); 5113 mLock.lock(); 5114 ALOGV("Record stopped OK"); 5115 } 5116 } 5117 } 5118} 5119 5120status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5121{ 5122 const size_t SIZE = 256; 5123 char buffer[SIZE]; 5124 String8 result; 5125 5126 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5127 result.append(buffer); 5128 5129 if (mActiveTrack != 0) { 5130 result.append("Active Track:\n"); 5131 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5132 mActiveTrack->dump(buffer, SIZE); 5133 result.append(buffer); 5134 5135 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5136 result.append(buffer); 5137 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5138 result.append(buffer); 5139 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5140 result.append(buffer); 5141 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5142 result.append(buffer); 5143 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5144 result.append(buffer); 5145 5146 5147 } else { 5148 result.append("No record client\n"); 5149 } 5150 write(fd, result.string(), result.size()); 5151 5152 dumpBase(fd, args); 5153 dumpEffectChains(fd, args); 5154 5155 return NO_ERROR; 5156} 5157 5158status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5159{ 5160 size_t framesReq = buffer->frameCount; 5161 size_t framesReady = mFrameCount - mRsmpInIndex; 5162 int channelCount; 5163 5164 if (framesReady == 0) { 5165 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5166 if (mBytesRead < 0) { 5167 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5168 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5169 // Force input into standby so that it tries to 5170 // recover at next read attempt 5171 mInput->stream->common.standby(&mInput->stream->common); 5172 usleep(kRecordThreadSleepUs); 5173 } 5174 buffer->raw = NULL; 5175 buffer->frameCount = 0; 5176 return NOT_ENOUGH_DATA; 5177 } 5178 mRsmpInIndex = 0; 5179 framesReady = mFrameCount; 5180 } 5181 5182 if (framesReq > framesReady) { 5183 framesReq = framesReady; 5184 } 5185 5186 if (mChannelCount == 1 && mReqChannelCount == 2) { 5187 channelCount = 1; 5188 } else { 5189 channelCount = 2; 5190 } 5191 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5192 buffer->frameCount = framesReq; 5193 return NO_ERROR; 5194} 5195 5196void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5197{ 5198 mRsmpInIndex += buffer->frameCount; 5199 buffer->frameCount = 0; 5200} 5201 5202bool AudioFlinger::RecordThread::checkForNewParameters_l() 5203{ 5204 bool reconfig = false; 5205 5206 while (!mNewParameters.isEmpty()) { 5207 status_t status = NO_ERROR; 5208 String8 keyValuePair = mNewParameters[0]; 5209 AudioParameter param = AudioParameter(keyValuePair); 5210 int value; 5211 audio_format_t reqFormat = mFormat; 5212 int reqSamplingRate = mReqSampleRate; 5213 int reqChannelCount = mReqChannelCount; 5214 5215 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5216 reqSamplingRate = value; 5217 reconfig = true; 5218 } 5219 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5220 reqFormat = (audio_format_t) value; 5221 reconfig = true; 5222 } 5223 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5224 reqChannelCount = popcount(value); 5225 reconfig = true; 5226 } 5227 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5228 // do not accept frame count changes if tracks are open as the track buffer 5229 // size depends on frame count and correct behavior would not be guaranteed 5230 // if frame count is changed after track creation 5231 if (mActiveTrack != 0) { 5232 status = INVALID_OPERATION; 5233 } else { 5234 reconfig = true; 5235 } 5236 } 5237 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5238 // forward device change to effects that have requested to be 5239 // aware of attached audio device. 5240 for (size_t i = 0; i < mEffectChains.size(); i++) { 5241 mEffectChains[i]->setDevice_l(value); 5242 } 5243 // store input device and output device but do not forward output device to audio HAL. 5244 // Note that status is ignored by the caller for output device 5245 // (see AudioFlinger::setParameters() 5246 if (value & AUDIO_DEVICE_OUT_ALL) { 5247 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5248 status = BAD_VALUE; 5249 } else { 5250 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5251 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5252 if (mTrack != NULL) { 5253 bool suspend = audio_is_bluetooth_sco_device( 5254 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5255 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5256 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5257 } 5258 } 5259 mDevice |= (uint32_t)value; 5260 } 5261 if (status == NO_ERROR) { 5262 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5263 if (status == INVALID_OPERATION) { 5264 mInput->stream->common.standby(&mInput->stream->common); 5265 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5266 } 5267 if (reconfig) { 5268 if (status == BAD_VALUE && 5269 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5270 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5271 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5272 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5273 (reqChannelCount < 3)) { 5274 status = NO_ERROR; 5275 } 5276 if (status == NO_ERROR) { 5277 readInputParameters(); 5278 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5279 } 5280 } 5281 } 5282 5283 mNewParameters.removeAt(0); 5284 5285 mParamStatus = status; 5286 mParamCond.signal(); 5287 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5288 // already timed out waiting for the status and will never signal the condition. 5289 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5290 } 5291 return reconfig; 5292} 5293 5294String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5295{ 5296 char *s; 5297 String8 out_s8 = String8(); 5298 5299 Mutex::Autolock _l(mLock); 5300 if (initCheck() != NO_ERROR) { 5301 return out_s8; 5302 } 5303 5304 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5305 out_s8 = String8(s); 5306 free(s); 5307 return out_s8; 5308} 5309 5310void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5311 AudioSystem::OutputDescriptor desc; 5312 void *param2 = NULL; 5313 5314 switch (event) { 5315 case AudioSystem::INPUT_OPENED: 5316 case AudioSystem::INPUT_CONFIG_CHANGED: 5317 desc.channels = mChannelMask; 5318 desc.samplingRate = mSampleRate; 5319 desc.format = mFormat; 5320 desc.frameCount = mFrameCount; 5321 desc.latency = 0; 5322 param2 = &desc; 5323 break; 5324 5325 case AudioSystem::INPUT_CLOSED: 5326 default: 5327 break; 5328 } 5329 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5330} 5331 5332void AudioFlinger::RecordThread::readInputParameters() 5333{ 5334 delete mRsmpInBuffer; 5335 // mRsmpInBuffer is always assigned a new[] below 5336 delete mRsmpOutBuffer; 5337 mRsmpOutBuffer = NULL; 5338 delete mResampler; 5339 mResampler = NULL; 5340 5341 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5342 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5343 mChannelCount = (uint16_t)popcount(mChannelMask); 5344 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5345 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5346 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5347 mFrameCount = mInputBytes / mFrameSize; 5348 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5349 5350 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5351 { 5352 int channelCount; 5353 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5354 // stereo to mono post process as the resampler always outputs stereo. 5355 if (mChannelCount == 1 && mReqChannelCount == 2) { 5356 channelCount = 1; 5357 } else { 5358 channelCount = 2; 5359 } 5360 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5361 mResampler->setSampleRate(mSampleRate); 5362 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5363 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5364 5365 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5366 if (mChannelCount == 1 && mReqChannelCount == 1) { 5367 mFrameCount >>= 1; 5368 } 5369 5370 } 5371 mRsmpInIndex = mFrameCount; 5372} 5373 5374unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5375{ 5376 Mutex::Autolock _l(mLock); 5377 if (initCheck() != NO_ERROR) { 5378 return 0; 5379 } 5380 5381 return mInput->stream->get_input_frames_lost(mInput->stream); 5382} 5383 5384uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5385{ 5386 Mutex::Autolock _l(mLock); 5387 uint32_t result = 0; 5388 if (getEffectChain_l(sessionId) != 0) { 5389 result = EFFECT_SESSION; 5390 } 5391 5392 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5393 result |= TRACK_SESSION; 5394 } 5395 5396 return result; 5397} 5398 5399AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5400{ 5401 Mutex::Autolock _l(mLock); 5402 return mTrack; 5403} 5404 5405AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5406{ 5407 Mutex::Autolock _l(mLock); 5408 return mInput; 5409} 5410 5411AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5412{ 5413 Mutex::Autolock _l(mLock); 5414 AudioStreamIn *input = mInput; 5415 mInput = NULL; 5416 return input; 5417} 5418 5419// this method must always be called either with ThreadBase mLock held or inside the thread loop 5420audio_stream_t* AudioFlinger::RecordThread::stream() 5421{ 5422 if (mInput == NULL) { 5423 return NULL; 5424 } 5425 return &mInput->stream->common; 5426} 5427 5428 5429// ---------------------------------------------------------------------------- 5430 5431audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5432 uint32_t *pSamplingRate, 5433 audio_format_t *pFormat, 5434 uint32_t *pChannels, 5435 uint32_t *pLatencyMs, 5436 uint32_t flags) 5437{ 5438 status_t status; 5439 PlaybackThread *thread = NULL; 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 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5465 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5466 &channels, &samplingRate, &outStream); 5467 mHardwareStatus = AUDIO_HW_IDLE; 5468 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5469 outStream, 5470 samplingRate, 5471 format, 5472 channels, 5473 status); 5474 5475 if (outStream != NULL) { 5476 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5477 audio_io_handle_t id = nextUniqueId(); 5478 5479 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5480 (format != AUDIO_FORMAT_PCM_16_BIT) || 5481 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5482 thread = new DirectOutputThread(this, output, id, *pDevices); 5483 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5484 } else { 5485 thread = new MixerThread(this, output, id, *pDevices); 5486 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5487 } 5488 mPlaybackThreads.add(id, thread); 5489 5490 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5491 if (pFormat != NULL) *pFormat = format; 5492 if (pChannels != NULL) *pChannels = channels; 5493 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5494 5495 // notify client processes of the new output creation 5496 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5497 return id; 5498 } 5499 5500 return 0; 5501} 5502 5503audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5504 audio_io_handle_t output2) 5505{ 5506 Mutex::Autolock _l(mLock); 5507 MixerThread *thread1 = checkMixerThread_l(output1); 5508 MixerThread *thread2 = checkMixerThread_l(output2); 5509 5510 if (thread1 == NULL || thread2 == NULL) { 5511 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5512 return 0; 5513 } 5514 5515 audio_io_handle_t id = nextUniqueId(); 5516 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5517 thread->addOutputTrack(thread2); 5518 mPlaybackThreads.add(id, thread); 5519 // notify client processes of the new output creation 5520 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5521 return id; 5522} 5523 5524status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5525{ 5526 // keep strong reference on the playback thread so that 5527 // it is not destroyed while exit() is executed 5528 sp <PlaybackThread> thread; 5529 { 5530 Mutex::Autolock _l(mLock); 5531 thread = checkPlaybackThread_l(output); 5532 if (thread == NULL) { 5533 return BAD_VALUE; 5534 } 5535 5536 ALOGV("closeOutput() %d", output); 5537 5538 if (thread->type() == ThreadBase::MIXER) { 5539 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5540 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5541 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5542 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5543 } 5544 } 5545 } 5546 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 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 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5691 mRecordThreads.removeItem(input); 5692 } 5693 thread->exit(); 5694 // The thread entity (active unit of execution) is no longer running here, 5695 // but the ThreadBase container still exists. 5696 5697 AudioStreamIn *in = thread->clearInput(); 5698 assert(in != NULL); 5699 // from now on thread->mInput is NULL 5700 in->hwDev->close_input_stream(in->hwDev, in->stream); 5701 delete in; 5702 5703 return NO_ERROR; 5704} 5705 5706status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5707{ 5708 Mutex::Autolock _l(mLock); 5709 MixerThread *dstThread = checkMixerThread_l(output); 5710 if (dstThread == NULL) { 5711 ALOGW("setStreamOutput() bad output id %d", output); 5712 return BAD_VALUE; 5713 } 5714 5715 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5716 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5717 5718 dstThread->setStreamValid(stream, true); 5719 5720 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5721 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5722 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5723 MixerThread *srcThread = (MixerThread *)thread; 5724 srcThread->setStreamValid(stream, false); 5725 srcThread->invalidateTracks(stream); 5726 } 5727 } 5728 5729 return NO_ERROR; 5730} 5731 5732 5733int AudioFlinger::newAudioSessionId() 5734{ 5735 return nextUniqueId(); 5736} 5737 5738void AudioFlinger::acquireAudioSessionId(int audioSession) 5739{ 5740 Mutex::Autolock _l(mLock); 5741 pid_t caller = IPCThreadState::self()->getCallingPid(); 5742 ALOGV("acquiring %d from %d", audioSession, caller); 5743 size_t num = mAudioSessionRefs.size(); 5744 for (size_t i = 0; i< num; i++) { 5745 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5746 if (ref->sessionid == audioSession && ref->pid == caller) { 5747 ref->cnt++; 5748 ALOGV(" incremented refcount to %d", ref->cnt); 5749 return; 5750 } 5751 } 5752 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5753 ALOGV(" added new entry for %d", audioSession); 5754} 5755 5756void AudioFlinger::releaseAudioSessionId(int audioSession) 5757{ 5758 Mutex::Autolock _l(mLock); 5759 pid_t caller = IPCThreadState::self()->getCallingPid(); 5760 ALOGV("releasing %d from %d", audioSession, caller); 5761 size_t num = mAudioSessionRefs.size(); 5762 for (size_t i = 0; i< num; i++) { 5763 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5764 if (ref->sessionid == audioSession && ref->pid == caller) { 5765 ref->cnt--; 5766 ALOGV(" decremented refcount to %d", ref->cnt); 5767 if (ref->cnt == 0) { 5768 mAudioSessionRefs.removeAt(i); 5769 delete ref; 5770 purgeStaleEffects_l(); 5771 } 5772 return; 5773 } 5774 } 5775 ALOGW("session id %d not found for pid %d", audioSession, caller); 5776} 5777 5778void AudioFlinger::purgeStaleEffects_l() { 5779 5780 ALOGV("purging stale effects"); 5781 5782 Vector< sp<EffectChain> > chains; 5783 5784 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5785 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5786 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5787 sp<EffectChain> ec = t->mEffectChains[j]; 5788 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5789 chains.push(ec); 5790 } 5791 } 5792 } 5793 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5794 sp<RecordThread> t = mRecordThreads.valueAt(i); 5795 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5796 sp<EffectChain> ec = t->mEffectChains[j]; 5797 chains.push(ec); 5798 } 5799 } 5800 5801 for (size_t i = 0; i < chains.size(); i++) { 5802 sp<EffectChain> ec = chains[i]; 5803 int sessionid = ec->sessionId(); 5804 sp<ThreadBase> t = ec->mThread.promote(); 5805 if (t == 0) { 5806 continue; 5807 } 5808 size_t numsessionrefs = mAudioSessionRefs.size(); 5809 bool found = false; 5810 for (size_t k = 0; k < numsessionrefs; k++) { 5811 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5812 if (ref->sessionid == sessionid) { 5813 ALOGV(" session %d still exists for %d with %d refs", 5814 sessionid, ref->pid, ref->cnt); 5815 found = true; 5816 break; 5817 } 5818 } 5819 if (!found) { 5820 // remove all effects from the chain 5821 while (ec->mEffects.size()) { 5822 sp<EffectModule> effect = ec->mEffects[0]; 5823 effect->unPin(); 5824 Mutex::Autolock _l (t->mLock); 5825 t->removeEffect_l(effect); 5826 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5827 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5828 if (handle != 0) { 5829 handle->mEffect.clear(); 5830 if (handle->mHasControl && handle->mEnabled) { 5831 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5832 } 5833 } 5834 } 5835 AudioSystem::unregisterEffect(effect->id()); 5836 } 5837 } 5838 } 5839 return; 5840} 5841 5842// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5843AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5844{ 5845 return mPlaybackThreads.valueFor(output).get(); 5846} 5847 5848// checkMixerThread_l() must be called with AudioFlinger::mLock held 5849AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5850{ 5851 PlaybackThread *thread = checkPlaybackThread_l(output); 5852 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5853} 5854 5855// checkRecordThread_l() must be called with AudioFlinger::mLock held 5856AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5857{ 5858 return mRecordThreads.valueFor(input).get(); 5859} 5860 5861uint32_t AudioFlinger::nextUniqueId() 5862{ 5863 return android_atomic_inc(&mNextUniqueId); 5864} 5865 5866AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5867{ 5868 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5869 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5870 AudioStreamOut *output = thread->getOutput(); 5871 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5872 return thread; 5873 } 5874 } 5875 return NULL; 5876} 5877 5878uint32_t AudioFlinger::primaryOutputDevice_l() 5879{ 5880 PlaybackThread *thread = primaryPlaybackThread_l(); 5881 5882 if (thread == NULL) { 5883 return 0; 5884 } 5885 5886 return thread->device(); 5887} 5888 5889 5890// ---------------------------------------------------------------------------- 5891// Effect management 5892// ---------------------------------------------------------------------------- 5893 5894 5895status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5896{ 5897 Mutex::Autolock _l(mLock); 5898 return EffectQueryNumberEffects(numEffects); 5899} 5900 5901status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5902{ 5903 Mutex::Autolock _l(mLock); 5904 return EffectQueryEffect(index, descriptor); 5905} 5906 5907status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5908 effect_descriptor_t *descriptor) const 5909{ 5910 Mutex::Autolock _l(mLock); 5911 return EffectGetDescriptor(pUuid, descriptor); 5912} 5913 5914 5915sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5916 effect_descriptor_t *pDesc, 5917 const sp<IEffectClient>& effectClient, 5918 int32_t priority, 5919 audio_io_handle_t io, 5920 int sessionId, 5921 status_t *status, 5922 int *id, 5923 int *enabled) 5924{ 5925 status_t lStatus = NO_ERROR; 5926 sp<EffectHandle> handle; 5927 effect_descriptor_t desc; 5928 5929 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5930 pid, effectClient.get(), priority, sessionId, io); 5931 5932 if (pDesc == NULL) { 5933 lStatus = BAD_VALUE; 5934 goto Exit; 5935 } 5936 5937 // check audio settings permission for global effects 5938 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5939 lStatus = PERMISSION_DENIED; 5940 goto Exit; 5941 } 5942 5943 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5944 // that can only be created by audio policy manager (running in same process) 5945 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5946 lStatus = PERMISSION_DENIED; 5947 goto Exit; 5948 } 5949 5950 if (io == 0) { 5951 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5952 // output must be specified by AudioPolicyManager when using session 5953 // AUDIO_SESSION_OUTPUT_STAGE 5954 lStatus = BAD_VALUE; 5955 goto Exit; 5956 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5957 // if the output returned by getOutputForEffect() is removed before we lock the 5958 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5959 // and we will exit safely 5960 io = AudioSystem::getOutputForEffect(&desc); 5961 } 5962 } 5963 5964 { 5965 Mutex::Autolock _l(mLock); 5966 5967 5968 if (!EffectIsNullUuid(&pDesc->uuid)) { 5969 // if uuid is specified, request effect descriptor 5970 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5971 if (lStatus < 0) { 5972 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5973 goto Exit; 5974 } 5975 } else { 5976 // if uuid is not specified, look for an available implementation 5977 // of the required type in effect factory 5978 if (EffectIsNullUuid(&pDesc->type)) { 5979 ALOGW("createEffect() no effect type"); 5980 lStatus = BAD_VALUE; 5981 goto Exit; 5982 } 5983 uint32_t numEffects = 0; 5984 effect_descriptor_t d; 5985 d.flags = 0; // prevent compiler warning 5986 bool found = false; 5987 5988 lStatus = EffectQueryNumberEffects(&numEffects); 5989 if (lStatus < 0) { 5990 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5991 goto Exit; 5992 } 5993 for (uint32_t i = 0; i < numEffects; i++) { 5994 lStatus = EffectQueryEffect(i, &desc); 5995 if (lStatus < 0) { 5996 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5997 continue; 5998 } 5999 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6000 // If matching type found save effect descriptor. If the session is 6001 // 0 and the effect is not auxiliary, continue enumeration in case 6002 // an auxiliary version of this effect type is available 6003 found = true; 6004 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6005 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6006 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6007 break; 6008 } 6009 } 6010 } 6011 if (!found) { 6012 lStatus = BAD_VALUE; 6013 ALOGW("createEffect() effect not found"); 6014 goto Exit; 6015 } 6016 // For same effect type, chose auxiliary version over insert version if 6017 // connect to output mix (Compliance to OpenSL ES) 6018 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6019 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6020 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6021 } 6022 } 6023 6024 // Do not allow auxiliary effects on a session different from 0 (output mix) 6025 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6026 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6027 lStatus = INVALID_OPERATION; 6028 goto Exit; 6029 } 6030 6031 // check recording permission for visualizer 6032 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6033 !recordingAllowed()) { 6034 lStatus = PERMISSION_DENIED; 6035 goto Exit; 6036 } 6037 6038 // return effect descriptor 6039 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6040 6041 // If output is not specified try to find a matching audio session ID in one of the 6042 // output threads. 6043 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6044 // because of code checking output when entering the function. 6045 // Note: io is never 0 when creating an effect on an input 6046 if (io == 0) { 6047 // look for the thread where the specified audio session is present 6048 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6049 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6050 io = mPlaybackThreads.keyAt(i); 6051 break; 6052 } 6053 } 6054 if (io == 0) { 6055 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6056 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6057 io = mRecordThreads.keyAt(i); 6058 break; 6059 } 6060 } 6061 } 6062 // If no output thread contains the requested session ID, default to 6063 // first output. The effect chain will be moved to the correct output 6064 // thread when a track with the same session ID is created 6065 if (io == 0 && mPlaybackThreads.size()) { 6066 io = mPlaybackThreads.keyAt(0); 6067 } 6068 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6069 } 6070 ThreadBase *thread = checkRecordThread_l(io); 6071 if (thread == NULL) { 6072 thread = checkPlaybackThread_l(io); 6073 if (thread == NULL) { 6074 ALOGE("createEffect() unknown output thread"); 6075 lStatus = BAD_VALUE; 6076 goto Exit; 6077 } 6078 } 6079 6080 sp<Client> client = registerPid_l(pid); 6081 6082 // create effect on selected output thread 6083 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6084 &desc, enabled, &lStatus); 6085 if (handle != 0 && id != NULL) { 6086 *id = handle->id(); 6087 } 6088 } 6089 6090Exit: 6091 if(status) { 6092 *status = lStatus; 6093 } 6094 return handle; 6095} 6096 6097status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6098 audio_io_handle_t dstOutput) 6099{ 6100 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6101 sessionId, srcOutput, dstOutput); 6102 Mutex::Autolock _l(mLock); 6103 if (srcOutput == dstOutput) { 6104 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6105 return NO_ERROR; 6106 } 6107 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6108 if (srcThread == NULL) { 6109 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6110 return BAD_VALUE; 6111 } 6112 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6113 if (dstThread == NULL) { 6114 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6115 return BAD_VALUE; 6116 } 6117 6118 Mutex::Autolock _dl(dstThread->mLock); 6119 Mutex::Autolock _sl(srcThread->mLock); 6120 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6121 6122 return NO_ERROR; 6123} 6124 6125// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6126status_t AudioFlinger::moveEffectChain_l(int sessionId, 6127 AudioFlinger::PlaybackThread *srcThread, 6128 AudioFlinger::PlaybackThread *dstThread, 6129 bool reRegister) 6130{ 6131 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6132 sessionId, srcThread, dstThread); 6133 6134 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6135 if (chain == 0) { 6136 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6137 sessionId, srcThread); 6138 return INVALID_OPERATION; 6139 } 6140 6141 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6142 // so that a new chain is created with correct parameters when first effect is added. This is 6143 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6144 // removed. 6145 srcThread->removeEffectChain_l(chain); 6146 6147 // transfer all effects one by one so that new effect chain is created on new thread with 6148 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6149 audio_io_handle_t dstOutput = dstThread->id(); 6150 sp<EffectChain> dstChain; 6151 uint32_t strategy = 0; // prevent compiler warning 6152 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6153 while (effect != 0) { 6154 srcThread->removeEffect_l(effect); 6155 dstThread->addEffect_l(effect); 6156 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6157 if (effect->state() == EffectModule::ACTIVE || 6158 effect->state() == EffectModule::STOPPING) { 6159 effect->start(); 6160 } 6161 // if the move request is not received from audio policy manager, the effect must be 6162 // re-registered with the new strategy and output 6163 if (dstChain == 0) { 6164 dstChain = effect->chain().promote(); 6165 if (dstChain == 0) { 6166 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6167 srcThread->addEffect_l(effect); 6168 return NO_INIT; 6169 } 6170 strategy = dstChain->strategy(); 6171 } 6172 if (reRegister) { 6173 AudioSystem::unregisterEffect(effect->id()); 6174 AudioSystem::registerEffect(&effect->desc(), 6175 dstOutput, 6176 strategy, 6177 sessionId, 6178 effect->id()); 6179 } 6180 effect = chain->getEffectFromId_l(0); 6181 } 6182 6183 return NO_ERROR; 6184} 6185 6186 6187// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6188sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6189 const sp<AudioFlinger::Client>& client, 6190 const sp<IEffectClient>& effectClient, 6191 int32_t priority, 6192 int sessionId, 6193 effect_descriptor_t *desc, 6194 int *enabled, 6195 status_t *status 6196 ) 6197{ 6198 sp<EffectModule> effect; 6199 sp<EffectHandle> handle; 6200 status_t lStatus; 6201 sp<EffectChain> chain; 6202 bool chainCreated = false; 6203 bool effectCreated = false; 6204 bool effectRegistered = false; 6205 6206 lStatus = initCheck(); 6207 if (lStatus != NO_ERROR) { 6208 ALOGW("createEffect_l() Audio driver not initialized."); 6209 goto Exit; 6210 } 6211 6212 // Do not allow effects with session ID 0 on direct output or duplicating threads 6213 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6214 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6215 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6216 desc->name, sessionId); 6217 lStatus = BAD_VALUE; 6218 goto Exit; 6219 } 6220 // Only Pre processor effects are allowed on input threads and only on input threads 6221 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6222 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6223 desc->name, desc->flags, mType); 6224 lStatus = BAD_VALUE; 6225 goto Exit; 6226 } 6227 6228 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6229 6230 { // scope for mLock 6231 Mutex::Autolock _l(mLock); 6232 6233 // check for existing effect chain with the requested audio session 6234 chain = getEffectChain_l(sessionId); 6235 if (chain == 0) { 6236 // create a new chain for this session 6237 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6238 chain = new EffectChain(this, sessionId); 6239 addEffectChain_l(chain); 6240 chain->setStrategy(getStrategyForSession_l(sessionId)); 6241 chainCreated = true; 6242 } else { 6243 effect = chain->getEffectFromDesc_l(desc); 6244 } 6245 6246 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6247 6248 if (effect == 0) { 6249 int id = mAudioFlinger->nextUniqueId(); 6250 // Check CPU and memory usage 6251 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6252 if (lStatus != NO_ERROR) { 6253 goto Exit; 6254 } 6255 effectRegistered = true; 6256 // create a new effect module if none present in the chain 6257 effect = new EffectModule(this, chain, desc, id, sessionId); 6258 lStatus = effect->status(); 6259 if (lStatus != NO_ERROR) { 6260 goto Exit; 6261 } 6262 lStatus = chain->addEffect_l(effect); 6263 if (lStatus != NO_ERROR) { 6264 goto Exit; 6265 } 6266 effectCreated = true; 6267 6268 effect->setDevice(mDevice); 6269 effect->setMode(mAudioFlinger->getMode()); 6270 } 6271 // create effect handle and connect it to effect module 6272 handle = new EffectHandle(effect, client, effectClient, priority); 6273 lStatus = effect->addHandle(handle); 6274 if (enabled != NULL) { 6275 *enabled = (int)effect->isEnabled(); 6276 } 6277 } 6278 6279Exit: 6280 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6281 Mutex::Autolock _l(mLock); 6282 if (effectCreated) { 6283 chain->removeEffect_l(effect); 6284 } 6285 if (effectRegistered) { 6286 AudioSystem::unregisterEffect(effect->id()); 6287 } 6288 if (chainCreated) { 6289 removeEffectChain_l(chain); 6290 } 6291 handle.clear(); 6292 } 6293 6294 if(status) { 6295 *status = lStatus; 6296 } 6297 return handle; 6298} 6299 6300sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6301{ 6302 sp<EffectChain> chain = getEffectChain_l(sessionId); 6303 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6304} 6305 6306// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6307// PlaybackThread::mLock held 6308status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6309{ 6310 // check for existing effect chain with the requested audio session 6311 int sessionId = effect->sessionId(); 6312 sp<EffectChain> chain = getEffectChain_l(sessionId); 6313 bool chainCreated = false; 6314 6315 if (chain == 0) { 6316 // create a new chain for this session 6317 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6318 chain = new EffectChain(this, sessionId); 6319 addEffectChain_l(chain); 6320 chain->setStrategy(getStrategyForSession_l(sessionId)); 6321 chainCreated = true; 6322 } 6323 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6324 6325 if (chain->getEffectFromId_l(effect->id()) != 0) { 6326 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6327 this, effect->desc().name, chain.get()); 6328 return BAD_VALUE; 6329 } 6330 6331 status_t status = chain->addEffect_l(effect); 6332 if (status != NO_ERROR) { 6333 if (chainCreated) { 6334 removeEffectChain_l(chain); 6335 } 6336 return status; 6337 } 6338 6339 effect->setDevice(mDevice); 6340 effect->setMode(mAudioFlinger->getMode()); 6341 return NO_ERROR; 6342} 6343 6344void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6345 6346 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6347 effect_descriptor_t desc = effect->desc(); 6348 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6349 detachAuxEffect_l(effect->id()); 6350 } 6351 6352 sp<EffectChain> chain = effect->chain().promote(); 6353 if (chain != 0) { 6354 // remove effect chain if removing last effect 6355 if (chain->removeEffect_l(effect) == 0) { 6356 removeEffectChain_l(chain); 6357 } 6358 } else { 6359 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6360 } 6361} 6362 6363void AudioFlinger::ThreadBase::lockEffectChains_l( 6364 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6365{ 6366 effectChains = mEffectChains; 6367 for (size_t i = 0; i < mEffectChains.size(); i++) { 6368 mEffectChains[i]->lock(); 6369 } 6370} 6371 6372void AudioFlinger::ThreadBase::unlockEffectChains( 6373 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6374{ 6375 for (size_t i = 0; i < effectChains.size(); i++) { 6376 effectChains[i]->unlock(); 6377 } 6378} 6379 6380sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6381{ 6382 Mutex::Autolock _l(mLock); 6383 return getEffectChain_l(sessionId); 6384} 6385 6386sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6387{ 6388 size_t size = mEffectChains.size(); 6389 for (size_t i = 0; i < size; i++) { 6390 if (mEffectChains[i]->sessionId() == sessionId) { 6391 return mEffectChains[i]; 6392 } 6393 } 6394 return 0; 6395} 6396 6397void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6398{ 6399 Mutex::Autolock _l(mLock); 6400 size_t size = mEffectChains.size(); 6401 for (size_t i = 0; i < size; i++) { 6402 mEffectChains[i]->setMode_l(mode); 6403 } 6404} 6405 6406void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6407 const wp<EffectHandle>& handle, 6408 bool unpinIfLast) { 6409 6410 Mutex::Autolock _l(mLock); 6411 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6412 // delete the effect module if removing last handle on it 6413 if (effect->removeHandle(handle) == 0) { 6414 if (!effect->isPinned() || unpinIfLast) { 6415 removeEffect_l(effect); 6416 AudioSystem::unregisterEffect(effect->id()); 6417 } 6418 } 6419} 6420 6421status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6422{ 6423 int session = chain->sessionId(); 6424 int16_t *buffer = mMixBuffer; 6425 bool ownsBuffer = false; 6426 6427 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6428 if (session > 0) { 6429 // Only one effect chain can be present in direct output thread and it uses 6430 // the mix buffer as input 6431 if (mType != DIRECT) { 6432 size_t numSamples = mFrameCount * mChannelCount; 6433 buffer = new int16_t[numSamples]; 6434 memset(buffer, 0, numSamples * sizeof(int16_t)); 6435 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6436 ownsBuffer = true; 6437 } 6438 6439 // Attach all tracks with same session ID to this chain. 6440 for (size_t i = 0; i < mTracks.size(); ++i) { 6441 sp<Track> track = mTracks[i]; 6442 if (session == track->sessionId()) { 6443 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6444 track->setMainBuffer(buffer); 6445 chain->incTrackCnt(); 6446 } 6447 } 6448 6449 // indicate all active tracks in the chain 6450 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6451 sp<Track> track = mActiveTracks[i].promote(); 6452 if (track == 0) continue; 6453 if (session == track->sessionId()) { 6454 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6455 chain->incActiveTrackCnt(); 6456 } 6457 } 6458 } 6459 6460 chain->setInBuffer(buffer, ownsBuffer); 6461 chain->setOutBuffer(mMixBuffer); 6462 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6463 // chains list in order to be processed last as it contains output stage effects 6464 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6465 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6466 // after track specific effects and before output stage 6467 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6468 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6469 // Effect chain for other sessions are inserted at beginning of effect 6470 // chains list to be processed before output mix effects. Relative order between other 6471 // sessions is not important 6472 size_t size = mEffectChains.size(); 6473 size_t i = 0; 6474 for (i = 0; i < size; i++) { 6475 if (mEffectChains[i]->sessionId() < session) break; 6476 } 6477 mEffectChains.insertAt(chain, i); 6478 checkSuspendOnAddEffectChain_l(chain); 6479 6480 return NO_ERROR; 6481} 6482 6483size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6484{ 6485 int session = chain->sessionId(); 6486 6487 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6488 6489 for (size_t i = 0; i < mEffectChains.size(); i++) { 6490 if (chain == mEffectChains[i]) { 6491 mEffectChains.removeAt(i); 6492 // detach all active tracks from the chain 6493 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6494 sp<Track> track = mActiveTracks[i].promote(); 6495 if (track == 0) continue; 6496 if (session == track->sessionId()) { 6497 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6498 chain.get(), session); 6499 chain->decActiveTrackCnt(); 6500 } 6501 } 6502 6503 // detach all tracks with same session ID from this chain 6504 for (size_t i = 0; i < mTracks.size(); ++i) { 6505 sp<Track> track = mTracks[i]; 6506 if (session == track->sessionId()) { 6507 track->setMainBuffer(mMixBuffer); 6508 chain->decTrackCnt(); 6509 } 6510 } 6511 break; 6512 } 6513 } 6514 return mEffectChains.size(); 6515} 6516 6517status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6518 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6519{ 6520 Mutex::Autolock _l(mLock); 6521 return attachAuxEffect_l(track, EffectId); 6522} 6523 6524status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6525 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6526{ 6527 status_t status = NO_ERROR; 6528 6529 if (EffectId == 0) { 6530 track->setAuxBuffer(0, NULL); 6531 } else { 6532 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6533 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6534 if (effect != 0) { 6535 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6536 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6537 } else { 6538 status = INVALID_OPERATION; 6539 } 6540 } else { 6541 status = BAD_VALUE; 6542 } 6543 } 6544 return status; 6545} 6546 6547void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6548{ 6549 for (size_t i = 0; i < mTracks.size(); ++i) { 6550 sp<Track> track = mTracks[i]; 6551 if (track->auxEffectId() == effectId) { 6552 attachAuxEffect_l(track, 0); 6553 } 6554 } 6555} 6556 6557status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6558{ 6559 // only one chain per input thread 6560 if (mEffectChains.size() != 0) { 6561 return INVALID_OPERATION; 6562 } 6563 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6564 6565 chain->setInBuffer(NULL); 6566 chain->setOutBuffer(NULL); 6567 6568 checkSuspendOnAddEffectChain_l(chain); 6569 6570 mEffectChains.add(chain); 6571 6572 return NO_ERROR; 6573} 6574 6575size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6576{ 6577 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6578 ALOGW_IF(mEffectChains.size() != 1, 6579 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6580 chain.get(), mEffectChains.size(), this); 6581 if (mEffectChains.size() == 1) { 6582 mEffectChains.removeAt(0); 6583 } 6584 return 0; 6585} 6586 6587// ---------------------------------------------------------------------------- 6588// EffectModule implementation 6589// ---------------------------------------------------------------------------- 6590 6591#undef LOG_TAG 6592#define LOG_TAG "AudioFlinger::EffectModule" 6593 6594AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6595 const wp<AudioFlinger::EffectChain>& chain, 6596 effect_descriptor_t *desc, 6597 int id, 6598 int sessionId) 6599 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6600 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6601{ 6602 ALOGV("Constructor %p", this); 6603 int lStatus; 6604 if (thread == NULL) { 6605 return; 6606 } 6607 6608 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6609 6610 // create effect engine from effect factory 6611 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6612 6613 if (mStatus != NO_ERROR) { 6614 return; 6615 } 6616 lStatus = init(); 6617 if (lStatus < 0) { 6618 mStatus = lStatus; 6619 goto Error; 6620 } 6621 6622 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6623 mPinned = true; 6624 } 6625 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6626 return; 6627Error: 6628 EffectRelease(mEffectInterface); 6629 mEffectInterface = NULL; 6630 ALOGV("Constructor Error %d", mStatus); 6631} 6632 6633AudioFlinger::EffectModule::~EffectModule() 6634{ 6635 ALOGV("Destructor %p", this); 6636 if (mEffectInterface != NULL) { 6637 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6638 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6639 sp<ThreadBase> thread = mThread.promote(); 6640 if (thread != 0) { 6641 audio_stream_t *stream = thread->stream(); 6642 if (stream != NULL) { 6643 stream->remove_audio_effect(stream, mEffectInterface); 6644 } 6645 } 6646 } 6647 // release effect engine 6648 EffectRelease(mEffectInterface); 6649 } 6650} 6651 6652status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6653{ 6654 status_t status; 6655 6656 Mutex::Autolock _l(mLock); 6657 int priority = handle->priority(); 6658 size_t size = mHandles.size(); 6659 sp<EffectHandle> h; 6660 size_t i; 6661 for (i = 0; i < size; i++) { 6662 h = mHandles[i].promote(); 6663 if (h == 0) continue; 6664 if (h->priority() <= priority) break; 6665 } 6666 // if inserted in first place, move effect control from previous owner to this handle 6667 if (i == 0) { 6668 bool enabled = false; 6669 if (h != 0) { 6670 enabled = h->enabled(); 6671 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6672 } 6673 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6674 status = NO_ERROR; 6675 } else { 6676 status = ALREADY_EXISTS; 6677 } 6678 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6679 mHandles.insertAt(handle, i); 6680 return status; 6681} 6682 6683size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6684{ 6685 Mutex::Autolock _l(mLock); 6686 size_t size = mHandles.size(); 6687 size_t i; 6688 for (i = 0; i < size; i++) { 6689 if (mHandles[i] == handle) break; 6690 } 6691 if (i == size) { 6692 return size; 6693 } 6694 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6695 6696 bool enabled = false; 6697 EffectHandle *hdl = handle.unsafe_get(); 6698 if (hdl != NULL) { 6699 ALOGV("removeHandle() unsafe_get OK"); 6700 enabled = hdl->enabled(); 6701 } 6702 mHandles.removeAt(i); 6703 size = mHandles.size(); 6704 // if removed from first place, move effect control from this handle to next in line 6705 if (i == 0 && size != 0) { 6706 sp<EffectHandle> h = mHandles[0].promote(); 6707 if (h != 0) { 6708 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6709 } 6710 } 6711 6712 // Prevent calls to process() and other functions on effect interface from now on. 6713 // The effect engine will be released by the destructor when the last strong reference on 6714 // this object is released which can happen after next process is called. 6715 if (size == 0 && !mPinned) { 6716 mState = DESTROYED; 6717 } 6718 6719 return size; 6720} 6721 6722sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6723{ 6724 Mutex::Autolock _l(mLock); 6725 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6726} 6727 6728void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6729{ 6730 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6731 // keep a strong reference on this EffectModule to avoid calling the 6732 // destructor before we exit 6733 sp<EffectModule> keep(this); 6734 { 6735 sp<ThreadBase> thread = mThread.promote(); 6736 if (thread != 0) { 6737 thread->disconnectEffect(keep, handle, unpinIfLast); 6738 } 6739 } 6740} 6741 6742void AudioFlinger::EffectModule::updateState() { 6743 Mutex::Autolock _l(mLock); 6744 6745 switch (mState) { 6746 case RESTART: 6747 reset_l(); 6748 // FALL THROUGH 6749 6750 case STARTING: 6751 // clear auxiliary effect input buffer for next accumulation 6752 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6753 memset(mConfig.inputCfg.buffer.raw, 6754 0, 6755 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6756 } 6757 start_l(); 6758 mState = ACTIVE; 6759 break; 6760 case STOPPING: 6761 stop_l(); 6762 mDisableWaitCnt = mMaxDisableWaitCnt; 6763 mState = STOPPED; 6764 break; 6765 case STOPPED: 6766 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6767 // turn off sequence. 6768 if (--mDisableWaitCnt == 0) { 6769 reset_l(); 6770 mState = IDLE; 6771 } 6772 break; 6773 default: //IDLE , ACTIVE, DESTROYED 6774 break; 6775 } 6776} 6777 6778void AudioFlinger::EffectModule::process() 6779{ 6780 Mutex::Autolock _l(mLock); 6781 6782 if (mState == DESTROYED || mEffectInterface == NULL || 6783 mConfig.inputCfg.buffer.raw == NULL || 6784 mConfig.outputCfg.buffer.raw == NULL) { 6785 return; 6786 } 6787 6788 if (isProcessEnabled()) { 6789 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6790 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6791 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6792 mConfig.inputCfg.buffer.s32, 6793 mConfig.inputCfg.buffer.frameCount/2); 6794 } 6795 6796 // do the actual processing in the effect engine 6797 int ret = (*mEffectInterface)->process(mEffectInterface, 6798 &mConfig.inputCfg.buffer, 6799 &mConfig.outputCfg.buffer); 6800 6801 // force transition to IDLE state when engine is ready 6802 if (mState == STOPPED && ret == -ENODATA) { 6803 mDisableWaitCnt = 1; 6804 } 6805 6806 // clear auxiliary effect input buffer for next accumulation 6807 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6808 memset(mConfig.inputCfg.buffer.raw, 0, 6809 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6810 } 6811 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6812 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6813 // If an insert effect is idle and input buffer is different from output buffer, 6814 // accumulate input onto output 6815 sp<EffectChain> chain = mChain.promote(); 6816 if (chain != 0 && chain->activeTrackCnt() != 0) { 6817 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6818 int16_t *in = mConfig.inputCfg.buffer.s16; 6819 int16_t *out = mConfig.outputCfg.buffer.s16; 6820 for (size_t i = 0; i < frameCnt; i++) { 6821 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6822 } 6823 } 6824 } 6825} 6826 6827void AudioFlinger::EffectModule::reset_l() 6828{ 6829 if (mEffectInterface == NULL) { 6830 return; 6831 } 6832 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6833} 6834 6835status_t AudioFlinger::EffectModule::configure() 6836{ 6837 uint32_t channels; 6838 if (mEffectInterface == NULL) { 6839 return NO_INIT; 6840 } 6841 6842 sp<ThreadBase> thread = mThread.promote(); 6843 if (thread == 0) { 6844 return DEAD_OBJECT; 6845 } 6846 6847 // TODO: handle configuration of effects replacing track process 6848 if (thread->channelCount() == 1) { 6849 channels = AUDIO_CHANNEL_OUT_MONO; 6850 } else { 6851 channels = AUDIO_CHANNEL_OUT_STEREO; 6852 } 6853 6854 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6855 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6856 } else { 6857 mConfig.inputCfg.channels = channels; 6858 } 6859 mConfig.outputCfg.channels = channels; 6860 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6861 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6862 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6863 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6864 mConfig.inputCfg.bufferProvider.cookie = NULL; 6865 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6866 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6867 mConfig.outputCfg.bufferProvider.cookie = NULL; 6868 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6869 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6870 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6871 // Insert effect: 6872 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6873 // always overwrites output buffer: input buffer == output buffer 6874 // - in other sessions: 6875 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6876 // other effect: overwrites output buffer: input buffer == output buffer 6877 // Auxiliary effect: 6878 // accumulates in output buffer: input buffer != output buffer 6879 // Therefore: accumulate <=> input buffer != output buffer 6880 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6881 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6882 } else { 6883 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6884 } 6885 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6886 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6887 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6888 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6889 6890 ALOGV("configure() %p thread %p buffer %p framecount %d", 6891 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6892 6893 status_t cmdStatus; 6894 uint32_t size = sizeof(int); 6895 status_t status = (*mEffectInterface)->command(mEffectInterface, 6896 EFFECT_CMD_SET_CONFIG, 6897 sizeof(effect_config_t), 6898 &mConfig, 6899 &size, 6900 &cmdStatus); 6901 if (status == 0) { 6902 status = cmdStatus; 6903 } 6904 6905 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6906 (1000 * mConfig.outputCfg.buffer.frameCount); 6907 6908 return status; 6909} 6910 6911status_t AudioFlinger::EffectModule::init() 6912{ 6913 Mutex::Autolock _l(mLock); 6914 if (mEffectInterface == NULL) { 6915 return NO_INIT; 6916 } 6917 status_t cmdStatus; 6918 uint32_t size = sizeof(status_t); 6919 status_t status = (*mEffectInterface)->command(mEffectInterface, 6920 EFFECT_CMD_INIT, 6921 0, 6922 NULL, 6923 &size, 6924 &cmdStatus); 6925 if (status == 0) { 6926 status = cmdStatus; 6927 } 6928 return status; 6929} 6930 6931status_t AudioFlinger::EffectModule::start() 6932{ 6933 Mutex::Autolock _l(mLock); 6934 return start_l(); 6935} 6936 6937status_t AudioFlinger::EffectModule::start_l() 6938{ 6939 if (mEffectInterface == NULL) { 6940 return NO_INIT; 6941 } 6942 status_t cmdStatus; 6943 uint32_t size = sizeof(status_t); 6944 status_t status = (*mEffectInterface)->command(mEffectInterface, 6945 EFFECT_CMD_ENABLE, 6946 0, 6947 NULL, 6948 &size, 6949 &cmdStatus); 6950 if (status == 0) { 6951 status = cmdStatus; 6952 } 6953 if (status == 0 && 6954 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6955 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6956 sp<ThreadBase> thread = mThread.promote(); 6957 if (thread != 0) { 6958 audio_stream_t *stream = thread->stream(); 6959 if (stream != NULL) { 6960 stream->add_audio_effect(stream, mEffectInterface); 6961 } 6962 } 6963 } 6964 return status; 6965} 6966 6967status_t AudioFlinger::EffectModule::stop() 6968{ 6969 Mutex::Autolock _l(mLock); 6970 return stop_l(); 6971} 6972 6973status_t AudioFlinger::EffectModule::stop_l() 6974{ 6975 if (mEffectInterface == NULL) { 6976 return NO_INIT; 6977 } 6978 status_t cmdStatus; 6979 uint32_t size = sizeof(status_t); 6980 status_t status = (*mEffectInterface)->command(mEffectInterface, 6981 EFFECT_CMD_DISABLE, 6982 0, 6983 NULL, 6984 &size, 6985 &cmdStatus); 6986 if (status == 0) { 6987 status = cmdStatus; 6988 } 6989 if (status == 0 && 6990 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6991 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6992 sp<ThreadBase> thread = mThread.promote(); 6993 if (thread != 0) { 6994 audio_stream_t *stream = thread->stream(); 6995 if (stream != NULL) { 6996 stream->remove_audio_effect(stream, mEffectInterface); 6997 } 6998 } 6999 } 7000 return status; 7001} 7002 7003status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7004 uint32_t cmdSize, 7005 void *pCmdData, 7006 uint32_t *replySize, 7007 void *pReplyData) 7008{ 7009 Mutex::Autolock _l(mLock); 7010// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7011 7012 if (mState == DESTROYED || mEffectInterface == NULL) { 7013 return NO_INIT; 7014 } 7015 status_t status = (*mEffectInterface)->command(mEffectInterface, 7016 cmdCode, 7017 cmdSize, 7018 pCmdData, 7019 replySize, 7020 pReplyData); 7021 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7022 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7023 for (size_t i = 1; i < mHandles.size(); i++) { 7024 sp<EffectHandle> h = mHandles[i].promote(); 7025 if (h != 0) { 7026 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7027 } 7028 } 7029 } 7030 return status; 7031} 7032 7033status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7034{ 7035 7036 Mutex::Autolock _l(mLock); 7037 ALOGV("setEnabled %p enabled %d", this, enabled); 7038 7039 if (enabled != isEnabled()) { 7040 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7041 if (enabled && status != NO_ERROR) { 7042 return status; 7043 } 7044 7045 switch (mState) { 7046 // going from disabled to enabled 7047 case IDLE: 7048 mState = STARTING; 7049 break; 7050 case STOPPED: 7051 mState = RESTART; 7052 break; 7053 case STOPPING: 7054 mState = ACTIVE; 7055 break; 7056 7057 // going from enabled to disabled 7058 case RESTART: 7059 mState = STOPPED; 7060 break; 7061 case STARTING: 7062 mState = IDLE; 7063 break; 7064 case ACTIVE: 7065 mState = STOPPING; 7066 break; 7067 case DESTROYED: 7068 return NO_ERROR; // simply ignore as we are being destroyed 7069 } 7070 for (size_t i = 1; i < mHandles.size(); i++) { 7071 sp<EffectHandle> h = mHandles[i].promote(); 7072 if (h != 0) { 7073 h->setEnabled(enabled); 7074 } 7075 } 7076 } 7077 return NO_ERROR; 7078} 7079 7080bool AudioFlinger::EffectModule::isEnabled() const 7081{ 7082 switch (mState) { 7083 case RESTART: 7084 case STARTING: 7085 case ACTIVE: 7086 return true; 7087 case IDLE: 7088 case STOPPING: 7089 case STOPPED: 7090 case DESTROYED: 7091 default: 7092 return false; 7093 } 7094} 7095 7096bool AudioFlinger::EffectModule::isProcessEnabled() const 7097{ 7098 switch (mState) { 7099 case RESTART: 7100 case ACTIVE: 7101 case STOPPING: 7102 case STOPPED: 7103 return true; 7104 case IDLE: 7105 case STARTING: 7106 case DESTROYED: 7107 default: 7108 return false; 7109 } 7110} 7111 7112status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7113{ 7114 Mutex::Autolock _l(mLock); 7115 status_t status = NO_ERROR; 7116 7117 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7118 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7119 if (isProcessEnabled() && 7120 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7121 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7122 status_t cmdStatus; 7123 uint32_t volume[2]; 7124 uint32_t *pVolume = NULL; 7125 uint32_t size = sizeof(volume); 7126 volume[0] = *left; 7127 volume[1] = *right; 7128 if (controller) { 7129 pVolume = volume; 7130 } 7131 status = (*mEffectInterface)->command(mEffectInterface, 7132 EFFECT_CMD_SET_VOLUME, 7133 size, 7134 volume, 7135 &size, 7136 pVolume); 7137 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7138 *left = volume[0]; 7139 *right = volume[1]; 7140 } 7141 } 7142 return status; 7143} 7144 7145status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7146{ 7147 Mutex::Autolock _l(mLock); 7148 status_t status = NO_ERROR; 7149 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7150 // audio pre processing modules on RecordThread can receive both output and 7151 // input device indication in the same call 7152 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7153 if (dev) { 7154 status_t cmdStatus; 7155 uint32_t size = sizeof(status_t); 7156 7157 status = (*mEffectInterface)->command(mEffectInterface, 7158 EFFECT_CMD_SET_DEVICE, 7159 sizeof(uint32_t), 7160 &dev, 7161 &size, 7162 &cmdStatus); 7163 if (status == NO_ERROR) { 7164 status = cmdStatus; 7165 } 7166 } 7167 dev = device & AUDIO_DEVICE_IN_ALL; 7168 if (dev) { 7169 status_t cmdStatus; 7170 uint32_t size = sizeof(status_t); 7171 7172 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7173 EFFECT_CMD_SET_INPUT_DEVICE, 7174 sizeof(uint32_t), 7175 &dev, 7176 &size, 7177 &cmdStatus); 7178 if (status2 == NO_ERROR) { 7179 status2 = cmdStatus; 7180 } 7181 if (status == NO_ERROR) { 7182 status = status2; 7183 } 7184 } 7185 } 7186 return status; 7187} 7188 7189status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7190{ 7191 Mutex::Autolock _l(mLock); 7192 status_t status = NO_ERROR; 7193 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7194 status_t cmdStatus; 7195 uint32_t size = sizeof(status_t); 7196 status = (*mEffectInterface)->command(mEffectInterface, 7197 EFFECT_CMD_SET_AUDIO_MODE, 7198 sizeof(audio_mode_t), 7199 &mode, 7200 &size, 7201 &cmdStatus); 7202 if (status == NO_ERROR) { 7203 status = cmdStatus; 7204 } 7205 } 7206 return status; 7207} 7208 7209void AudioFlinger::EffectModule::setSuspended(bool suspended) 7210{ 7211 Mutex::Autolock _l(mLock); 7212 mSuspended = suspended; 7213} 7214 7215bool AudioFlinger::EffectModule::suspended() const 7216{ 7217 Mutex::Autolock _l(mLock); 7218 return mSuspended; 7219} 7220 7221status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7222{ 7223 const size_t SIZE = 256; 7224 char buffer[SIZE]; 7225 String8 result; 7226 7227 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7228 result.append(buffer); 7229 7230 bool locked = tryLock(mLock); 7231 // failed to lock - AudioFlinger is probably deadlocked 7232 if (!locked) { 7233 result.append("\t\tCould not lock Fx mutex:\n"); 7234 } 7235 7236 result.append("\t\tSession Status State Engine:\n"); 7237 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7238 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7239 result.append(buffer); 7240 7241 result.append("\t\tDescriptor:\n"); 7242 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7243 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7244 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7245 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7246 result.append(buffer); 7247 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7248 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7249 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7250 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7251 result.append(buffer); 7252 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7253 mDescriptor.apiVersion, 7254 mDescriptor.flags); 7255 result.append(buffer); 7256 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7257 mDescriptor.name); 7258 result.append(buffer); 7259 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7260 mDescriptor.implementor); 7261 result.append(buffer); 7262 7263 result.append("\t\t- Input configuration:\n"); 7264 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7265 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7266 (uint32_t)mConfig.inputCfg.buffer.raw, 7267 mConfig.inputCfg.buffer.frameCount, 7268 mConfig.inputCfg.samplingRate, 7269 mConfig.inputCfg.channels, 7270 mConfig.inputCfg.format); 7271 result.append(buffer); 7272 7273 result.append("\t\t- Output configuration:\n"); 7274 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7275 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7276 (uint32_t)mConfig.outputCfg.buffer.raw, 7277 mConfig.outputCfg.buffer.frameCount, 7278 mConfig.outputCfg.samplingRate, 7279 mConfig.outputCfg.channels, 7280 mConfig.outputCfg.format); 7281 result.append(buffer); 7282 7283 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7284 result.append(buffer); 7285 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7286 for (size_t i = 0; i < mHandles.size(); ++i) { 7287 sp<EffectHandle> handle = mHandles[i].promote(); 7288 if (handle != 0) { 7289 handle->dump(buffer, SIZE); 7290 result.append(buffer); 7291 } 7292 } 7293 7294 result.append("\n"); 7295 7296 write(fd, result.string(), result.length()); 7297 7298 if (locked) { 7299 mLock.unlock(); 7300 } 7301 7302 return NO_ERROR; 7303} 7304 7305// ---------------------------------------------------------------------------- 7306// EffectHandle implementation 7307// ---------------------------------------------------------------------------- 7308 7309#undef LOG_TAG 7310#define LOG_TAG "AudioFlinger::EffectHandle" 7311 7312AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7313 const sp<AudioFlinger::Client>& client, 7314 const sp<IEffectClient>& effectClient, 7315 int32_t priority) 7316 : BnEffect(), 7317 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7318 mPriority(priority), mHasControl(false), mEnabled(false) 7319{ 7320 ALOGV("constructor %p", this); 7321 7322 if (client == 0) { 7323 return; 7324 } 7325 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7326 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7327 if (mCblkMemory != 0) { 7328 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7329 7330 if (mCblk != NULL) { 7331 new(mCblk) effect_param_cblk_t(); 7332 mBuffer = (uint8_t *)mCblk + bufOffset; 7333 } 7334 } else { 7335 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7336 return; 7337 } 7338} 7339 7340AudioFlinger::EffectHandle::~EffectHandle() 7341{ 7342 ALOGV("Destructor %p", this); 7343 disconnect(false); 7344 ALOGV("Destructor DONE %p", this); 7345} 7346 7347status_t AudioFlinger::EffectHandle::enable() 7348{ 7349 ALOGV("enable %p", this); 7350 if (!mHasControl) return INVALID_OPERATION; 7351 if (mEffect == 0) return DEAD_OBJECT; 7352 7353 if (mEnabled) { 7354 return NO_ERROR; 7355 } 7356 7357 mEnabled = true; 7358 7359 sp<ThreadBase> thread = mEffect->thread().promote(); 7360 if (thread != 0) { 7361 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7362 } 7363 7364 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7365 if (mEffect->suspended()) { 7366 return NO_ERROR; 7367 } 7368 7369 status_t status = mEffect->setEnabled(true); 7370 if (status != NO_ERROR) { 7371 if (thread != 0) { 7372 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7373 } 7374 mEnabled = false; 7375 } 7376 return status; 7377} 7378 7379status_t AudioFlinger::EffectHandle::disable() 7380{ 7381 ALOGV("disable %p", this); 7382 if (!mHasControl) return INVALID_OPERATION; 7383 if (mEffect == 0) return DEAD_OBJECT; 7384 7385 if (!mEnabled) { 7386 return NO_ERROR; 7387 } 7388 mEnabled = false; 7389 7390 if (mEffect->suspended()) { 7391 return NO_ERROR; 7392 } 7393 7394 status_t status = mEffect->setEnabled(false); 7395 7396 sp<ThreadBase> thread = mEffect->thread().promote(); 7397 if (thread != 0) { 7398 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7399 } 7400 7401 return status; 7402} 7403 7404void AudioFlinger::EffectHandle::disconnect() 7405{ 7406 disconnect(true); 7407} 7408 7409void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7410{ 7411 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7412 if (mEffect == 0) { 7413 return; 7414 } 7415 mEffect->disconnect(this, unpinIfLast); 7416 7417 if (mHasControl && mEnabled) { 7418 sp<ThreadBase> thread = mEffect->thread().promote(); 7419 if (thread != 0) { 7420 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7421 } 7422 } 7423 7424 // release sp on module => module destructor can be called now 7425 mEffect.clear(); 7426 if (mClient != 0) { 7427 if (mCblk != NULL) { 7428 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7429 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7430 } 7431 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7432 // Client destructor must run with AudioFlinger mutex locked 7433 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7434 mClient.clear(); 7435 } 7436} 7437 7438status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7439 uint32_t cmdSize, 7440 void *pCmdData, 7441 uint32_t *replySize, 7442 void *pReplyData) 7443{ 7444// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7445// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7446 7447 // only get parameter command is permitted for applications not controlling the effect 7448 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7449 return INVALID_OPERATION; 7450 } 7451 if (mEffect == 0) return DEAD_OBJECT; 7452 if (mClient == 0) return INVALID_OPERATION; 7453 7454 // handle commands that are not forwarded transparently to effect engine 7455 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7456 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7457 // no risk to block the whole media server process or mixer threads is we are stuck here 7458 Mutex::Autolock _l(mCblk->lock); 7459 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7460 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7461 mCblk->serverIndex = 0; 7462 mCblk->clientIndex = 0; 7463 return BAD_VALUE; 7464 } 7465 status_t status = NO_ERROR; 7466 while (mCblk->serverIndex < mCblk->clientIndex) { 7467 int reply; 7468 uint32_t rsize = sizeof(int); 7469 int *p = (int *)(mBuffer + mCblk->serverIndex); 7470 int size = *p++; 7471 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7472 ALOGW("command(): invalid parameter block size"); 7473 break; 7474 } 7475 effect_param_t *param = (effect_param_t *)p; 7476 if (param->psize == 0 || param->vsize == 0) { 7477 ALOGW("command(): null parameter or value size"); 7478 mCblk->serverIndex += size; 7479 continue; 7480 } 7481 uint32_t psize = sizeof(effect_param_t) + 7482 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7483 param->vsize; 7484 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7485 psize, 7486 p, 7487 &rsize, 7488 &reply); 7489 // stop at first error encountered 7490 if (ret != NO_ERROR) { 7491 status = ret; 7492 *(int *)pReplyData = reply; 7493 break; 7494 } else if (reply != NO_ERROR) { 7495 *(int *)pReplyData = reply; 7496 break; 7497 } 7498 mCblk->serverIndex += size; 7499 } 7500 mCblk->serverIndex = 0; 7501 mCblk->clientIndex = 0; 7502 return status; 7503 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7504 *(int *)pReplyData = NO_ERROR; 7505 return enable(); 7506 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7507 *(int *)pReplyData = NO_ERROR; 7508 return disable(); 7509 } 7510 7511 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7512} 7513 7514void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7515{ 7516 ALOGV("setControl %p control %d", this, hasControl); 7517 7518 mHasControl = hasControl; 7519 mEnabled = enabled; 7520 7521 if (signal && mEffectClient != 0) { 7522 mEffectClient->controlStatusChanged(hasControl); 7523 } 7524} 7525 7526void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7527 uint32_t cmdSize, 7528 void *pCmdData, 7529 uint32_t replySize, 7530 void *pReplyData) 7531{ 7532 if (mEffectClient != 0) { 7533 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7534 } 7535} 7536 7537 7538 7539void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7540{ 7541 if (mEffectClient != 0) { 7542 mEffectClient->enableStatusChanged(enabled); 7543 } 7544} 7545 7546status_t AudioFlinger::EffectHandle::onTransact( 7547 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7548{ 7549 return BnEffect::onTransact(code, data, reply, flags); 7550} 7551 7552 7553void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7554{ 7555 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7556 7557 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7558 (mClient == 0) ? getpid_cached : mClient->pid(), 7559 mPriority, 7560 mHasControl, 7561 !locked, 7562 mCblk ? mCblk->clientIndex : 0, 7563 mCblk ? mCblk->serverIndex : 0 7564 ); 7565 7566 if (locked) { 7567 mCblk->lock.unlock(); 7568 } 7569} 7570 7571#undef LOG_TAG 7572#define LOG_TAG "AudioFlinger::EffectChain" 7573 7574AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7575 int sessionId) 7576 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7577 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7578 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7579{ 7580 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7581 if (thread == NULL) { 7582 return; 7583 } 7584 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7585 thread->frameCount(); 7586} 7587 7588AudioFlinger::EffectChain::~EffectChain() 7589{ 7590 if (mOwnInBuffer) { 7591 delete mInBuffer; 7592 } 7593 7594} 7595 7596// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7597sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7598{ 7599 size_t size = mEffects.size(); 7600 7601 for (size_t i = 0; i < size; i++) { 7602 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7603 return mEffects[i]; 7604 } 7605 } 7606 return 0; 7607} 7608 7609// getEffectFromId_l() must be called with ThreadBase::mLock held 7610sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7611{ 7612 size_t size = mEffects.size(); 7613 7614 for (size_t i = 0; i < size; i++) { 7615 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7616 if (id == 0 || mEffects[i]->id() == id) { 7617 return mEffects[i]; 7618 } 7619 } 7620 return 0; 7621} 7622 7623// getEffectFromType_l() must be called with ThreadBase::mLock held 7624sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7625 const effect_uuid_t *type) 7626{ 7627 size_t size = mEffects.size(); 7628 7629 for (size_t i = 0; i < size; i++) { 7630 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7631 return mEffects[i]; 7632 } 7633 } 7634 return 0; 7635} 7636 7637// Must be called with EffectChain::mLock locked 7638void AudioFlinger::EffectChain::process_l() 7639{ 7640 sp<ThreadBase> thread = mThread.promote(); 7641 if (thread == 0) { 7642 ALOGW("process_l(): cannot promote mixer thread"); 7643 return; 7644 } 7645 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7646 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7647 // always process effects unless no more tracks are on the session and the effect tail 7648 // has been rendered 7649 bool doProcess = true; 7650 if (!isGlobalSession) { 7651 bool tracksOnSession = (trackCnt() != 0); 7652 7653 if (!tracksOnSession && mTailBufferCount == 0) { 7654 doProcess = false; 7655 } 7656 7657 if (activeTrackCnt() == 0) { 7658 // if no track is active and the effect tail has not been rendered, 7659 // the input buffer must be cleared here as the mixer process will not do it 7660 if (tracksOnSession || mTailBufferCount > 0) { 7661 size_t numSamples = thread->frameCount() * thread->channelCount(); 7662 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7663 if (mTailBufferCount > 0) { 7664 mTailBufferCount--; 7665 } 7666 } 7667 } 7668 } 7669 7670 size_t size = mEffects.size(); 7671 if (doProcess) { 7672 for (size_t i = 0; i < size; i++) { 7673 mEffects[i]->process(); 7674 } 7675 } 7676 for (size_t i = 0; i < size; i++) { 7677 mEffects[i]->updateState(); 7678 } 7679} 7680 7681// addEffect_l() must be called with PlaybackThread::mLock held 7682status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7683{ 7684 effect_descriptor_t desc = effect->desc(); 7685 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7686 7687 Mutex::Autolock _l(mLock); 7688 effect->setChain(this); 7689 sp<ThreadBase> thread = mThread.promote(); 7690 if (thread == 0) { 7691 return NO_INIT; 7692 } 7693 effect->setThread(thread); 7694 7695 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7696 // Auxiliary effects are inserted at the beginning of mEffects vector as 7697 // they are processed first and accumulated in chain input buffer 7698 mEffects.insertAt(effect, 0); 7699 7700 // the input buffer for auxiliary effect contains mono samples in 7701 // 32 bit format. This is to avoid saturation in AudoMixer 7702 // accumulation stage. Saturation is done in EffectModule::process() before 7703 // calling the process in effect engine 7704 size_t numSamples = thread->frameCount(); 7705 int32_t *buffer = new int32_t[numSamples]; 7706 memset(buffer, 0, numSamples * sizeof(int32_t)); 7707 effect->setInBuffer((int16_t *)buffer); 7708 // auxiliary effects output samples to chain input buffer for further processing 7709 // by insert effects 7710 effect->setOutBuffer(mInBuffer); 7711 } else { 7712 // Insert effects are inserted at the end of mEffects vector as they are processed 7713 // after track and auxiliary effects. 7714 // Insert effect order as a function of indicated preference: 7715 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7716 // another effect is present 7717 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7718 // last effect claiming first position 7719 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7720 // first effect claiming last position 7721 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7722 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7723 // already present 7724 7725 size_t size = mEffects.size(); 7726 size_t idx_insert = size; 7727 ssize_t idx_insert_first = -1; 7728 ssize_t idx_insert_last = -1; 7729 7730 for (size_t i = 0; i < size; i++) { 7731 effect_descriptor_t d = mEffects[i]->desc(); 7732 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7733 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7734 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7735 // check invalid effect chaining combinations 7736 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7737 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7738 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7739 return INVALID_OPERATION; 7740 } 7741 // remember position of first insert effect and by default 7742 // select this as insert position for new effect 7743 if (idx_insert == size) { 7744 idx_insert = i; 7745 } 7746 // remember position of last insert effect claiming 7747 // first position 7748 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7749 idx_insert_first = i; 7750 } 7751 // remember position of first insert effect claiming 7752 // last position 7753 if (iPref == EFFECT_FLAG_INSERT_LAST && 7754 idx_insert_last == -1) { 7755 idx_insert_last = i; 7756 } 7757 } 7758 } 7759 7760 // modify idx_insert from first position if needed 7761 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7762 if (idx_insert_last != -1) { 7763 idx_insert = idx_insert_last; 7764 } else { 7765 idx_insert = size; 7766 } 7767 } else { 7768 if (idx_insert_first != -1) { 7769 idx_insert = idx_insert_first + 1; 7770 } 7771 } 7772 7773 // always read samples from chain input buffer 7774 effect->setInBuffer(mInBuffer); 7775 7776 // if last effect in the chain, output samples to chain 7777 // output buffer, otherwise to chain input buffer 7778 if (idx_insert == size) { 7779 if (idx_insert != 0) { 7780 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7781 mEffects[idx_insert-1]->configure(); 7782 } 7783 effect->setOutBuffer(mOutBuffer); 7784 } else { 7785 effect->setOutBuffer(mInBuffer); 7786 } 7787 mEffects.insertAt(effect, idx_insert); 7788 7789 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7790 } 7791 effect->configure(); 7792 return NO_ERROR; 7793} 7794 7795// removeEffect_l() must be called with PlaybackThread::mLock held 7796size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7797{ 7798 Mutex::Autolock _l(mLock); 7799 size_t size = mEffects.size(); 7800 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7801 7802 for (size_t i = 0; i < size; i++) { 7803 if (effect == mEffects[i]) { 7804 // calling stop here will remove pre-processing effect from the audio HAL. 7805 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7806 // the middle of a read from audio HAL 7807 if (mEffects[i]->state() == EffectModule::ACTIVE || 7808 mEffects[i]->state() == EffectModule::STOPPING) { 7809 mEffects[i]->stop(); 7810 } 7811 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7812 delete[] effect->inBuffer(); 7813 } else { 7814 if (i == size - 1 && i != 0) { 7815 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7816 mEffects[i - 1]->configure(); 7817 } 7818 } 7819 mEffects.removeAt(i); 7820 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7821 break; 7822 } 7823 } 7824 7825 return mEffects.size(); 7826} 7827 7828// setDevice_l() must be called with PlaybackThread::mLock held 7829void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7830{ 7831 size_t size = mEffects.size(); 7832 for (size_t i = 0; i < size; i++) { 7833 mEffects[i]->setDevice(device); 7834 } 7835} 7836 7837// setMode_l() must be called with PlaybackThread::mLock held 7838void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7839{ 7840 size_t size = mEffects.size(); 7841 for (size_t i = 0; i < size; i++) { 7842 mEffects[i]->setMode(mode); 7843 } 7844} 7845 7846// setVolume_l() must be called with PlaybackThread::mLock held 7847bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7848{ 7849 uint32_t newLeft = *left; 7850 uint32_t newRight = *right; 7851 bool hasControl = false; 7852 int ctrlIdx = -1; 7853 size_t size = mEffects.size(); 7854 7855 // first update volume controller 7856 for (size_t i = size; i > 0; i--) { 7857 if (mEffects[i - 1]->isProcessEnabled() && 7858 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7859 ctrlIdx = i - 1; 7860 hasControl = true; 7861 break; 7862 } 7863 } 7864 7865 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7866 if (hasControl) { 7867 *left = mNewLeftVolume; 7868 *right = mNewRightVolume; 7869 } 7870 return hasControl; 7871 } 7872 7873 mVolumeCtrlIdx = ctrlIdx; 7874 mLeftVolume = newLeft; 7875 mRightVolume = newRight; 7876 7877 // second get volume update from volume controller 7878 if (ctrlIdx >= 0) { 7879 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7880 mNewLeftVolume = newLeft; 7881 mNewRightVolume = newRight; 7882 } 7883 // then indicate volume to all other effects in chain. 7884 // Pass altered volume to effects before volume controller 7885 // and requested volume to effects after controller 7886 uint32_t lVol = newLeft; 7887 uint32_t rVol = newRight; 7888 7889 for (size_t i = 0; i < size; i++) { 7890 if ((int)i == ctrlIdx) continue; 7891 // this also works for ctrlIdx == -1 when there is no volume controller 7892 if ((int)i > ctrlIdx) { 7893 lVol = *left; 7894 rVol = *right; 7895 } 7896 mEffects[i]->setVolume(&lVol, &rVol, false); 7897 } 7898 *left = newLeft; 7899 *right = newRight; 7900 7901 return hasControl; 7902} 7903 7904status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7905{ 7906 const size_t SIZE = 256; 7907 char buffer[SIZE]; 7908 String8 result; 7909 7910 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7911 result.append(buffer); 7912 7913 bool locked = tryLock(mLock); 7914 // failed to lock - AudioFlinger is probably deadlocked 7915 if (!locked) { 7916 result.append("\tCould not lock mutex:\n"); 7917 } 7918 7919 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7920 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7921 mEffects.size(), 7922 (uint32_t)mInBuffer, 7923 (uint32_t)mOutBuffer, 7924 mActiveTrackCnt); 7925 result.append(buffer); 7926 write(fd, result.string(), result.size()); 7927 7928 for (size_t i = 0; i < mEffects.size(); ++i) { 7929 sp<EffectModule> effect = mEffects[i]; 7930 if (effect != 0) { 7931 effect->dump(fd, args); 7932 } 7933 } 7934 7935 if (locked) { 7936 mLock.unlock(); 7937 } 7938 7939 return NO_ERROR; 7940} 7941 7942// must be called with ThreadBase::mLock held 7943void AudioFlinger::EffectChain::setEffectSuspended_l( 7944 const effect_uuid_t *type, bool suspend) 7945{ 7946 sp<SuspendedEffectDesc> desc; 7947 // use effect type UUID timelow as key as there is no real risk of identical 7948 // timeLow fields among effect type UUIDs. 7949 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7950 if (suspend) { 7951 if (index >= 0) { 7952 desc = mSuspendedEffects.valueAt(index); 7953 } else { 7954 desc = new SuspendedEffectDesc(); 7955 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7956 mSuspendedEffects.add(type->timeLow, desc); 7957 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7958 } 7959 if (desc->mRefCount++ == 0) { 7960 sp<EffectModule> effect = getEffectIfEnabled(type); 7961 if (effect != 0) { 7962 desc->mEffect = effect; 7963 effect->setSuspended(true); 7964 effect->setEnabled(false); 7965 } 7966 } 7967 } else { 7968 if (index < 0) { 7969 return; 7970 } 7971 desc = mSuspendedEffects.valueAt(index); 7972 if (desc->mRefCount <= 0) { 7973 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7974 desc->mRefCount = 1; 7975 } 7976 if (--desc->mRefCount == 0) { 7977 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7978 if (desc->mEffect != 0) { 7979 sp<EffectModule> effect = desc->mEffect.promote(); 7980 if (effect != 0) { 7981 effect->setSuspended(false); 7982 sp<EffectHandle> handle = effect->controlHandle(); 7983 if (handle != 0) { 7984 effect->setEnabled(handle->enabled()); 7985 } 7986 } 7987 desc->mEffect.clear(); 7988 } 7989 mSuspendedEffects.removeItemsAt(index); 7990 } 7991 } 7992} 7993 7994// must be called with ThreadBase::mLock held 7995void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7996{ 7997 sp<SuspendedEffectDesc> desc; 7998 7999 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8000 if (suspend) { 8001 if (index >= 0) { 8002 desc = mSuspendedEffects.valueAt(index); 8003 } else { 8004 desc = new SuspendedEffectDesc(); 8005 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8006 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8007 } 8008 if (desc->mRefCount++ == 0) { 8009 Vector< sp<EffectModule> > effects; 8010 getSuspendEligibleEffects(effects); 8011 for (size_t i = 0; i < effects.size(); i++) { 8012 setEffectSuspended_l(&effects[i]->desc().type, true); 8013 } 8014 } 8015 } else { 8016 if (index < 0) { 8017 return; 8018 } 8019 desc = mSuspendedEffects.valueAt(index); 8020 if (desc->mRefCount <= 0) { 8021 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8022 desc->mRefCount = 1; 8023 } 8024 if (--desc->mRefCount == 0) { 8025 Vector<const effect_uuid_t *> types; 8026 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8027 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8028 continue; 8029 } 8030 types.add(&mSuspendedEffects.valueAt(i)->mType); 8031 } 8032 for (size_t i = 0; i < types.size(); i++) { 8033 setEffectSuspended_l(types[i], false); 8034 } 8035 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8036 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8037 } 8038 } 8039} 8040 8041 8042// The volume effect is used for automated tests only 8043#ifndef OPENSL_ES_H_ 8044static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8045 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8046const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8047#endif //OPENSL_ES_H_ 8048 8049bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8050{ 8051 // auxiliary effects and visualizer are never suspended on output mix 8052 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8053 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8054 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8055 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8056 return false; 8057 } 8058 return true; 8059} 8060 8061void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8062{ 8063 effects.clear(); 8064 for (size_t i = 0; i < mEffects.size(); i++) { 8065 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8066 effects.add(mEffects[i]); 8067 } 8068 } 8069} 8070 8071sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8072 const effect_uuid_t *type) 8073{ 8074 sp<EffectModule> effect = getEffectFromType_l(type); 8075 return effect != 0 && effect->isEnabled() ? effect : 0; 8076} 8077 8078void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8079 bool enabled) 8080{ 8081 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8082 if (enabled) { 8083 if (index < 0) { 8084 // if the effect is not suspend check if all effects are suspended 8085 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8086 if (index < 0) { 8087 return; 8088 } 8089 if (!isEffectEligibleForSuspend(effect->desc())) { 8090 return; 8091 } 8092 setEffectSuspended_l(&effect->desc().type, enabled); 8093 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8094 if (index < 0) { 8095 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8096 return; 8097 } 8098 } 8099 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8100 effect->desc().type.timeLow); 8101 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8102 // if effect is requested to suspended but was not yet enabled, supend it now. 8103 if (desc->mEffect == 0) { 8104 desc->mEffect = effect; 8105 effect->setEnabled(false); 8106 effect->setSuspended(true); 8107 } 8108 } else { 8109 if (index < 0) { 8110 return; 8111 } 8112 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8113 effect->desc().type.timeLow); 8114 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8115 desc->mEffect.clear(); 8116 effect->setSuspended(false); 8117 } 8118} 8119 8120#undef LOG_TAG 8121#define LOG_TAG "AudioFlinger" 8122 8123// ---------------------------------------------------------------------------- 8124 8125status_t AudioFlinger::onTransact( 8126 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8127{ 8128 return BnAudioFlinger::onTransact(code, data, reply, flags); 8129} 8130 8131}; // namespace android 8132